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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 1 Jul 2013 11:01:20 +0200
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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Hi Stefan,


2013/7/1 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> 4.5: This really underlines the need to specify in detail what the
> allowed SDP variations are (or, if we remove SDP, to specify any other
> mechanism to exchange the things needed to set up RTP and other things
> going on the wire).

The draft http://www.ietf.org/id/draft-raymond-rtcweb-webrtc-js-obj-api-rat=
ionale-00.txt
clearly proposes that NO media description format is required in the
wire. This is not about SDP vs SDP+XML vs SDP-NG.

The draft explains (in section 5) why it is not needed to mandate how
media info must be sent in the wire. Instead, let the developer to
retrieve the media/transport information from the API and send it in
the wire in the way it wants.


Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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On 2013-07-01 11:01, I=F1aki Baz Castillo wrote:=0A=
> Hi Stefan,=0A=
>=0A=
>=0A=
> 2013/7/1 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com>:=0A=
>> 4.5: This really underlines the need to specify in detail what the=0A=
>> allowed SDP variations are (or, if we remove SDP, to specify any other=
=0A=
>> mechanism to exchange the things needed to set up RTP and other things=
=0A=
>> going on the wire).=0A=
>=0A=
> The draft http://www.ietf.org/id/draft-raymond-rtcweb-webrtc-js-obj-api-r=
ationale-00.txt=0A=
> clearly proposes that NO media description format is required in the=0A=
> wire. This is not about SDP vs SDP+XML vs SDP-NG.=0A=
>=0A=
> The draft explains (in section 5) why it is not needed to mandate how=0A=
> media info must be sent in the wire. Instead, let the developer to=0A=
> retrieve the media/transport information from the API and send it in=0A=
> the wire in the way it wants.=0A=
=0A=
I still think my point is valid. If the app i browser A retrieves this =0A=
media/transport info using the API, and sends it to the peer app in =0A=
another browser B, it must still mean the same for both browsers.=0A=
=0A=
True, we're struggling with this part for SDP. But removing it and only =0A=
have API calls does not get us away from defining exactly what those =0A=
calls mean.=0A=
=0A=
>=0A=
>=0A=
> Best regards.=0A=
>=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2013/7/1 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
>> The draft explains (in section 5) why it is not needed to mandate how
>> media info must be sent in the wire. Instead, let the developer to
>> retrieve the media/transport information from the API and send it in
>> the wire in the way it wants.
>
> I still think my point is valid. If the app i browser A retrieves this
> media/transport info using the API, and sends it to the peer app in
> another browser B, it must still mean the same for both browsers.
>
> True, we're struggling with this part for SDP. But removing it and only
> have API calls does not get us away from defining exactly what those
> calls mean.


Stefan, the media/transport info will be sent from browser A to
browser B using the custom format the developer of the website chose,
period. I don't understand why you think that the format of the
media/transport info must be constant/fixed for any website using
WebRTC. It makes no sense IMHO. That is up to the website developer.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From ted.ietf@gmail.com  Mon Jul  1 08:59:16 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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--089e0122894664793804e0754fc6
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On Mon, Jul 1, 2013 at 7:28 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> Stefan, the media/transport info will be sent from browser A to
> browser B using the custom format the developer of the website chose,
> period. I don't understand why you think that the format of the
> media/transport info must be constant/fixed for any website using
> WebRTC.


Because ultimately the media is sent via transports set up by the browser.
If the browser must understand every custom format, the system cannot
scale.

regards,

Ted Hardie

--089e0122894664793804e0754fc6
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On Mon, Jul 1, 2013 at 7:28 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;=
<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</s=
pan> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Stefan, the media/transport info will be sent from browser A to<br>
browser B using the custom format the developer of the website chose,<br>
period. I don&#39;t understand why you think that the format of the<br>
media/transport info must be constant/fixed for any website using<br>
WebRTC. </blockquote><div><br>Because ultimately the media is sent via tran=
sports set up by the browser.=A0 If the browser must understand every custo=
m format, the system cannot scale.=A0 <br><br>regards,<br><br>Ted Hardie<br=
>
</div></div>

--089e0122894664793804e0754fc6--

From ibc@aliax.net  Mon Jul  1 09:06:47 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 1 Jul 2013 18:05:52 +0200
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2013/7/1 Ted Hardie <ted.ietf@gmail.com>:
> On Mon, Jul 1, 2013 at 7:28 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
>>
>> Stefan, the media/transport info will be sent from browser A to
>> browser B using the custom format the developer of the website chose,
>> period. I don't understand why you think that the format of the
>> media/transport info must be constant/fixed for any website using
>> WebRTC.
>
>
> Because ultimately the media is sent via transports set up by the browser=
.
> If the browser must understand every custom format, the system cannot sca=
le.


Ted, the media can be *perfectly* sent in-the-wire with any custom
format the web developer builts (JSON, XML, plain SDP or whatever).
The receiver JS app parses the information (i.e. converts from JSON to
an Object), extracts the media/transport fields and pass them to the
PeerConnection via API methods.

It is really so hard? this is how WWW works (in contrast to how SIP works).

If it is possible, please take a look to
http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationale=
-00.

Best regards.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On 2013-07-01 18:06, I=F1aki Baz Castillo wrote:=0A=
> 2013/7/1 Ted Hardie <ted.ietf@gmail.com>:=0A=
>> On Mon, Jul 1, 2013 at 7:28 AM, I=F1aki Baz Castillo <ibc@aliax.net> wro=
te:=0A=
>>>=0A=
>>> Stefan, the media/transport info will be sent from browser A to=0A=
>>> browser B using the custom format the developer of the website chose,=
=0A=
>>> period. I don't understand why you think that the format of the=0A=
>>> media/transport info must be constant/fixed for any website using=0A=
>>> WebRTC.=0A=
>>=0A=
>>=0A=
>> Because ultimately the media is sent via transports set up by the browse=
r.=0A=
>> If the browser must understand every custom format, the system cannot sc=
ale.=0A=
>=0A=
>=0A=
> Ted, the media can be *perfectly* sent in-the-wire with any custom=0A=
> format the web developer builts (JSON, XML, plain SDP or whatever).=0A=
=0A=
I had the impression that we had consensus for using (S)RTP for the media.=
=0A=
=0A=
> The receiver JS app parses the information (i.e. converts from JSON to=0A=
> an Object), extracts the media/transport fields and pass them to the=0A=
> PeerConnection via API methods.=0A=
=0A=
My point is that when the app passes the fields to the PeerConnection, =0A=
it must be specified exactly what those fields mean (so that the RTP =0A=
send by one browser can be used by another browser receiving it). And =0A=
that requirement is the same regardless of if we use SDP generated and =0A=
consumed by the browser to describe it or something else.=0A=
=0A=
>=0A=
> It is really so hard? this is how WWW works (in contrast to how SIP works=
).=0A=
>=0A=
> If it is possible, please take a look to=0A=
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00.=0A=
>=0A=
> Best regards.=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

From saverio.mascolo@gmail.com  Mon Jul  1 10:10:09 2013
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Subject: Re: [rtcweb] More H.264 vs VP8 tests
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an evaluation of how h264 and vp8 are able to match a changing quality
would be interesting


On Sat, Jun 29, 2013 at 3:17 PM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 6/25/13 12:17 PM, Harald Alvestrand wrote:
> > Again - thanks for releasing this openly!
> >
> > I ran the scripts (with a few tweaks; you run on a system where sh is
> > bash, not dash, for instance), and got the same numbers within +/- 0.5%
> > (probably some binary version skew);
>
> I re-run the scripts on another computer with another OS today, and I
> get exactly the same results as Bo sent out. I noted however that if the
> input clips are not cut at 10s (but used in their entire length) the
> results get slightly different, but within +/- 0.5%. Can this be the
> reason why you get slightly different numbers?
>
>
> > we may have disagreements on the
> > parameters to use, but we agree on the numbers those parameters produce=
.
> >
> > (I have since modified the Google framework to include a script that
> > pulls in the sources for the needed binaries and compiles them - if you
> > want to make 100% sure people are working from the same sources, you ma=
y
> > want to rebase to a newer version of the comparision toolkit.)
> >
> > On 06/22/2013 03:41 PM, Bo Burman wrote:
> >> Hi all,
> >>
> >> We have had a look at Google's comparison between VP8 and H.264
> constrained baseline that was posted on April 3rd (
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07028.html). This
> post contains, as the one mentioned above (and if the attachments make it
> to the list), information on the exact tools and options used for encodin=
g
> and should thus be repeatable by anyone interested.
> >>
> >> As was already stated by others on this list, one major problem is tha=
t
> Google's test involves the rate control mechanism. Typically codecs are
> measured with rate control turned off, since it acts as a huge noise on t=
he
> measurement. Instead we propose to compare the codecs using fixed
> qp-levels. The qp-level is the quantization parameter that affects the
> rate/distortion tradeoff. Comparing using fixed qp-levels is what has bee=
n
> used when benchmarking HEVC against H.264 in the JCT-VC standardization,
> for instance. We are going to select a codec (essentially bit stream
> format), not a rate control mechanism: Once the codec is selected you can
> choose whatever rate control mechanism you wish.
> >>
> >> We used Google's excellent framework as the baseline and changed the
> parameter settings in order to make it possible to measure using fixed qp=
.
> We used the same sequences, but limited them to the first 10 seconds sinc=
e
> they varied from 10 seconds to minutes; this also eased computation time.
> >>
> >> We used two H.264 encoder implementations: X264, which is an
> open-source codec that can operate in everything from real-time to slow,
> and JM which is the reference implementation that was used to develop
> H.264. JM is very slow but attempts to be very efficient in terms of bits
> per quality. The results were as follows:
> >>
> >> X264 baseline vs VP8: H.264 wins with 1%
> >> JM baseline vs VP8: H.264 wins with 4%
> >>
> >> Running times:
> >> X264: 1 hour 3 minutes
> >> VP8: 2 hours 0 minutes
> >> JM: order of magnitude slower
> >>
> >> It is interesting to note that the measurements are more stable in the
> new test; the variance of the percentages for the sequences is now around
> 70, down from around 700 in Google's test of April 3rd.  We believe this =
is
> due to the removal of the rate controller, which acts like noise on the
> measurements.
> >>
> >> We also tried setting H.264 to constrained high (no interlace and no
> B-pictures, compared to high). The results were then:
> >>
> >> X264 constrained high vs VP8: H.264 wins with 25%
> >> JM constrained high vs VP8: H.264 wins with 24%
> >>
> >> We also note that the script that Google provided to calculate the rat=
e
> differences ("BD-rate") does not give exactly the same numbers as the
> JCT-VC-way of calculating BD-rate. The main difference is that the JM sco=
re
> for constrained high is better (around 29%) if the JCT-VC way of
> calculating BD-rate is used.
> >>
> >> In summary we think that proper testing can conclude that there is no
> clear performance advantage to any codec between VP8 and H.264 baseline.
> When comparing VP8 against H.264 constrained high on the other hand, it
> seems like there is an advantage for H.264 constrained high.
> >>
> >> The attached file includes the files necessary to reproduce the test.
> >>
> >> Best Regards,
> >>
> >> Bo Burman
> >>
> >>
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c1f7a4e7aa6a04e0764cad
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<div dir=3D"ltr">an evaluation of how h264 and vp8 are able to match a chan=
ging quality would be interesting<div class=3D"gmail_extra"><br><br><div cl=
ass=3D"gmail_quote">On Sat, Jun 29, 2013 at 3:17 PM, Stefan H=E5kansson LK =
<span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" t=
arget=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 6/25/13 12:17 PM, Haral=
d Alvestrand wrote:<br>
&gt; Again - thanks for releasing this openly!<br>
&gt;<br>
&gt; I ran the scripts (with a few tweaks; you run on a system where sh is<=
br>
&gt; bash, not dash, for instance), and got the same numbers within +/- 0.5=
%<br>
&gt; (probably some binary version skew);<br>
<br>
</div>I re-run the scripts on another computer with another OS today, and I=
<br>
get exactly the same results as Bo sent out. I noted however that if the<br=
>
input clips are not cut at 10s (but used in their entire length) the<br>
results get slightly different, but within +/- 0.5%. Can this be the<br>
reason why you get slightly different numbers?<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; we may have disagreements on the<br>
&gt; parameters to use, but we agree on the numbers those parameters produc=
e.<br>
&gt;<br>
&gt; (I have since modified the Google framework to include a script that<b=
r>
&gt; pulls in the sources for the needed binaries and compiles them - if yo=
u<br>
&gt; want to make 100% sure people are working from the same sources, you m=
ay<br>
&gt; want to rebase to a newer version of the comparision toolkit.)<br>
&gt;<br>
&gt; On 06/22/2013 03:41 PM, Bo Burman wrote:<br>
&gt;&gt; Hi all,<br>
&gt;&gt;<br>
&gt;&gt; We have had a look at Google&#39;s comparison between VP8 and H.26=
4 constrained baseline that was posted on April 3rd (<a href=3D"http://www.=
ietf.org/mail-archive/web/rtcweb/current/msg07028.html" target=3D"_blank">h=
ttp://www.ietf.org/mail-archive/web/rtcweb/current/msg07028.html</a>). This=
 post contains, as the one mentioned above (and if the attachments make it =
to the list), information on the exact tools and options used for encoding =
and should thus be repeatable by anyone interested.<br>


&gt;&gt;<br>
&gt;&gt; As was already stated by others on this list, one major problem is=
 that Google&#39;s test involves the rate control mechanism. Typically code=
cs are measured with rate control turned off, since it acts as a huge noise=
 on the measurement. Instead we propose to compare the codecs using fixed q=
p-levels. The qp-level is the quantization parameter that affects the rate/=
distortion tradeoff. Comparing using fixed qp-levels is what has been used =
when benchmarking HEVC against H.264 in the JCT-VC standardization, for ins=
tance. We are going to select a codec (essentially bit stream format), not =
a rate control mechanism: Once the codec is selected you can choose whateve=
r rate control mechanism you wish.<br>


&gt;&gt;<br>
&gt;&gt; We used Google&#39;s excellent framework as the baseline and chang=
ed the parameter settings in order to make it possible to measure using fix=
ed qp. We used the same sequences, but limited them to the first 10 seconds=
 since they varied from 10 seconds to minutes; this also eased computation =
time.<br>


&gt;&gt;<br>
&gt;&gt; We used two H.264 encoder implementations: X264, which is an open-=
source codec that can operate in everything from real-time to slow, and JM =
which is the reference implementation that was used to develop H.264. JM is=
 very slow but attempts to be very efficient in terms of bits per quality. =
The results were as follows:<br>


&gt;&gt;<br>
&gt;&gt; X264 baseline vs VP8: H.264 wins with 1%<br>
&gt;&gt; JM baseline vs VP8: H.264 wins with 4%<br>
&gt;&gt;<br>
&gt;&gt; Running times:<br>
&gt;&gt; X264: 1 hour 3 minutes<br>
&gt;&gt; VP8: 2 hours 0 minutes<br>
&gt;&gt; JM: order of magnitude slower<br>
&gt;&gt;<br>
&gt;&gt; It is interesting to note that the measurements are more stable in=
 the new test; the variance of the percentages for the sequences is now aro=
und 70, down from around 700 in Google&#39;s test of April 3rd. =A0We belie=
ve this is due to the removal of the rate controller, which acts like noise=
 on the measurements.<br>


&gt;&gt;<br>
&gt;&gt; We also tried setting H.264 to constrained high (no interlace and =
no B-pictures, compared to high). The results were then:<br>
&gt;&gt;<br>
&gt;&gt; X264 constrained high vs VP8: H.264 wins with 25%<br>
&gt;&gt; JM constrained high vs VP8: H.264 wins with 24%<br>
&gt;&gt;<br>
&gt;&gt; We also note that the script that Google provided to calculate the=
 rate differences (&quot;BD-rate&quot;) does not give exactly the same numb=
ers as the JCT-VC-way of calculating BD-rate. The main difference is that t=
he JM score for constrained high is better (around 29%) if the JCT-VC way o=
f calculating BD-rate is used.<br>


&gt;&gt;<br>
&gt;&gt; In summary we think that proper testing can conclude that there is=
 no clear performance advantage to any codec between VP8 and H.264 baseline=
. When comparing VP8 against H.264 constrained high on the other hand, it s=
eems like there is an advantage for H.264 constrained high.<br>


&gt;&gt;<br>
&gt;&gt; The attached file includes the files necessary to reproduce the te=
st.<br>
&gt;&gt;<br>
&gt;&gt; Best Regards,<br>
&gt;&gt;<br>
&gt;&gt; Bo Burman<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div>=A0</div></div></=
div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:


>> Ted, the media can be *perfectly* sent in-the-wire with any custom
>> format the web developer builts (JSON, XML, plain SDP or whatever).
>
> I had the impression that we had consensus for using (S)RTP for the media=
.

I meant "SDP". Of course the browser sends RTP. What I mean is that we
DO NOT need to play with a SDP blob to transmit media/transport info
between browsers and between JS and WebRTC API.


>> The receiver JS app parses the information (i.e. converts from JSON to
>> an Object), extracts the media/transport fields and pass them to the
>> PeerConnection via API methods.
>
> My point is that when the app passes the fields to the PeerConnection,
> it must be specified exactly what those fields mean (so that the RTP
> send by one browser can be used by another browser receiving it). And
> that requirement is the same regardless of if we use SDP generated and
> consumed by the browser to describe it or something else.

Sure. The point is that we don't need (and don't want) SDP since it
becomes a barrier for innovation. All the details in the draft:

  http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00.


Best regards.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On Mon, Jul 1, 2013 at 9:05 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/1 Ted Hardie <ted.ietf@gmail.com>:
> > On Mon, Jul 1, 2013 at 7:28 AM, I=F1aki Baz Castillo <ibc@aliax.net>
> wrote:
> >>
> >> Stefan, the media/transport info will be sent from browser A to
> >> browser B using the custom format the developer of the website chose,
> >> period. I don't understand why you think that the format of the
> >> media/transport info must be constant/fixed for any website using
> >> WebRTC.
> >
> >
> > Because ultimately the media is sent via transports set up by the
> browser.
> > If the browser must understand every custom format, the system cannot
> scale.
>
>
> Ted, the media can be *perfectly* sent in-the-wire with any custom
> format the web developer builts (JSON, XML, plain SDP or whatever).
>

I'm sorry, I think I must misunderstand you.  You are planning to take
media from a source  and encode it using json, xml, or SDP and send that
over the wire?  Or you mean the media *signalling* can use json, xml, or
SDP and go over the wire?


> The receiver JS app parses the information (i.e. converts from JSON to
> an Object), extracts the media/transport fields and pass them to the
> PeerConnection via API methods.
>

It is really so hard? this is how WWW works (in contrast to how SIP works).
>
>
As has been repeatedly mentioned, you have a peer communication here that
has to be implemented by the browser.  That means the downloaded javascript
application must tell a browser what transports it needs so that it can
create the appropriate connections.  If it is using RTP/UDP, it ends up
needing to tell the browser what enough detail to construct the RTP/UPD
streams.  What payload format, for example, is needed.  As I said before,
that means that having no standard format across the API between the JS
application and the browser has very poor scaling characteristics.  You
might replace SDP as the interface between them, but not specifying
anything there doesn't seem to work, at least in my personal opinion.

regards,

Ted Hardie


If it is possible, please take a look to
>
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00
> .
>
> Best regards.
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>

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On Mon, Jul 1, 2013 at 9:05 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;=
<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</s=
pan> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
2013/7/1 Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmai=
l.com</a>&gt;:<br>
<div><div class=3D"h5">&gt; On Mon, Jul 1, 2013 at 7:28 AM, I=F1aki Baz Cas=
tillo &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Stefan, the media/transport info will be sent from browser A to<br=
>
&gt;&gt; browser B using the custom format the developer of the website cho=
se,<br>
&gt;&gt; period. I don&#39;t understand why you think that the format of th=
e<br>
&gt;&gt; media/transport info must be constant/fixed for any website using<=
br>
&gt;&gt; WebRTC.<br>
&gt;<br>
&gt;<br>
&gt; Because ultimately the media is sent via transports set up by the brow=
ser.<br>
&gt; If the browser must understand every custom format, the system cannot =
scale.<br>
<br>
<br>
</div></div>Ted, the media can be *perfectly* sent in-the-wire with any cus=
tom<br>
format the web developer builts (JSON, XML, plain SDP or whatever).<br></bl=
ockquote><div><br>I&#39;m sorry, I think I must misunderstand you.=A0 You a=
re planning to take media from a source=A0 and encode it using json, xml, o=
r SDP and send that over the wire?=A0 Or you mean the media *signalling* ca=
n use json, xml, or SDP and go over the wire?<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex">
The receiver JS app parses the information (i.e. converts from JSON to<br>
an Object), extracts the media/transport fields and pass them to the<br>
PeerConnection via API methods.<br>
=A0</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
It is really so hard? this is how WWW works (in contrast to how SIP works).=
<br>
<br></blockquote><div><br>As has been repeatedly mentioned, you have a peer=
 communication here that has to be implemented by the browser.=A0 That mean=
s the downloaded javascript application must tell a browser what transports=
 it needs so that it can create the appropriate connections.=A0 If it is us=
ing RTP/UDP, it ends up needing to tell the browser what enough detail to c=
onstruct the RTP/UPD streams.=A0 What payload format, for example, is neede=
d.=A0 As I said before, that means that having no standard format across th=
e API between the JS application and the browser has very poor scaling char=
acteristics.=A0 You might replace SDP as the interface between them, but no=
t specifying anything there doesn&#39;t seem to work, at least in my person=
al opinion.<br>
<br>regards,<br><br>Ted Hardie<br><br><br></div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
If it is possible, please take a look to<br>
<a href=3D"http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-ap=
i-rationale-00" target=3D"_blank">http://tools.ietf.org/html/draft-raymond-=
rtcweb-webrtc-js-obj-api-rationale-00</a>.<br>
<br>
Best regards.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div></div></blockquote></div><br>

--e89a8f3bb04f47672d04e0765aa8--

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Ted Hardie <ted.ietf@gmail.com>:
>> Ted, the media can be *perfectly* sent in-the-wire with any custom
>> format the web developer builts (JSON, XML, plain SDP or whatever).
>
>
> I'm sorry, I think I must misunderstand you.  You are planning to take me=
dia
> from a source  and encode it using json, xml, or SDP and send that over t=
he
> wire?  Or you mean the media *signalling* can use json, xml, or SDP and g=
o
> over the wire?

Of course, I mean "the media signaling". But that is not just about
"in-the-wire". Current WebRTC specs mandate that the information unit
exchanged between the JS app and the WebRTC stack must be a SDP blob.

What I mean is:

- The JS app uses the WebRTC API to retrieve a JS Object (or Objects)
with transport and media info.

- The JS app serializes such a information in any custom way defined
by the website developer (this can be a JSON body, XML, SDP! or
whatever).

- The JS app sends the information to the remote peer.

- The JS app of the remote peer decodes the information (since it is a
JS app designed by the same domain/website) and extracts all the
transport/media fields.

- The JS app of the remote peer creates a local PC and makes calls to
API methods of PC for passing those transport/media fields.

- ...and remove O/A mandate.

But honestly, all of this is really well described and exposed in the draft=
:

http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationale=
-00





>> The receiver JS app parses the information (i.e. converts from JSON to
>> an Object), extracts the media/transport fields and pass them to the
>> PeerConnection via API methods.
>>
>>
>> It is really so hard? this is how WWW works (in contrast to how SIP
>> works).
>>
>
> As has been repeatedly mentioned, you have a peer communication here that
> has to be implemented by the browser.  That means the downloaded javascri=
pt
> application must tell a browser what transports it needs so that it can
> create the appropriate connections.  If it is using RTP/UDP, it ends up
> needing to tell the browser what enough detail to construct the RTP/UPD
> streams.  What payload format, for example, is needed.  As I said before,
> that means that having no standard format across the API between the JS
> application and the browser has very poor scaling characteristics.  You
> might replace SDP as the interface between them, but not specifying anyth=
ing
> there doesn't seem to work, at least in my personal opinion.

I invite you to read the draft which describes how your concerns can
be perfectly accomplished without mandating SDP or any other format:

http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationale=
-00



Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 2013-07-01 19:22, I=F1aki Baz Castillo wrote:=0A=
> 2013/7/1 Ted Hardie <ted.ietf@gmail.com>:=0A=
>>> Ted, the media can be *perfectly* sent in-the-wire with any custom=0A=
>>> format the web developer builts (JSON, XML, plain SDP or whatever).=0A=
>>=0A=
>>=0A=
>> I'm sorry, I think I must misunderstand you.  You are planning to take m=
edia=0A=
>> from a source  and encode it using json, xml, or SDP and send that over =
the=0A=
>> wire?  Or you mean the media *signalling* can use json, xml, or SDP and =
go=0A=
>> over the wire?=0A=
>=0A=
> Of course, I mean "the media signaling". But that is not just about=0A=
> "in-the-wire". Current WebRTC specs mandate that the information unit=0A=
> exchanged between the JS app and the WebRTC stack must be a SDP blob.=0A=
>=0A=
> What I mean is:=0A=
>=0A=
> - The JS app uses the WebRTC API to retrieve a JS Object (or Objects)=0A=
> with transport and media info.=0A=
>=0A=
> - The JS app serializes such a information in any custom way defined=0A=
> by the website developer (this can be a JSON body, XML, SDP! or=0A=
> whatever).=0A=
>=0A=
> - The JS app sends the information to the remote peer.=0A=
>=0A=
> - The JS app of the remote peer decodes the information (since it is a=0A=
> JS app designed by the same domain/website) and extracts all the=0A=
> transport/media fields.=0A=
>=0A=
> - The JS app of the remote peer creates a local PC and makes calls to=0A=
> API methods of PC for passing those transport/media fields.=0A=
>=0A=
> - ...and remove O/A mandate.=0A=
=0A=
What if first browser generated a JS Object that said (for this example =0A=
we assume one single audio track) it preferred to send audio using =0A=
Opus2.0 but could also do Opus and G.711. If the second browser (which =0A=
has not yet been upgraded) could only decode Opus and G.711, would it =0A=
not have to tell the first browser about this?=0A=
=0A=
Are you perhaps assuming a capability exchange of some sort before any =0A=
media can be set up?=0A=
=0A=
>=0A=
> But honestly, all of this is really well described and exposed in the dra=
ft:=0A=
>=0A=
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>>> The receiver JS app parses the information (i.e. converts from JSON to=
=0A=
>>> an Object), extracts the media/transport fields and pass them to the=0A=
>>> PeerConnection via API methods.=0A=
>>>=0A=
>>>=0A=
>>> It is really so hard? this is how WWW works (in contrast to how SIP=0A=
>>> works).=0A=
>>>=0A=
>>=0A=
>> As has been repeatedly mentioned, you have a peer communication here tha=
t=0A=
>> has to be implemented by the browser.  That means the downloaded javascr=
ipt=0A=
>> application must tell a browser what transports it needs so that it can=
=0A=
>> create the appropriate connections.  If it is using RTP/UDP, it ends up=
=0A=
>> needing to tell the browser what enough detail to construct the RTP/UPD=
=0A=
>> streams.  What payload format, for example, is needed.  As I said before=
,=0A=
>> that means that having no standard format across the API between the JS=
=0A=
>> application and the browser has very poor scaling characteristics.  You=
=0A=
>> might replace SDP as the interface between them, but not specifying anyt=
hing=0A=
>> there doesn't seem to work, at least in my personal opinion.=0A=
>=0A=
> I invite you to read the draft which describes how your concerns can=0A=
> be perfectly accomplished without mandating SDP or any other format:=0A=
>=0A=
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00=0A=
>=0A=
>=0A=
>=0A=
> Best regards.=0A=
>=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

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Date: Mon, 1 Jul 2013 11:17:49 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On Mon, Jul 1, 2013 at 10:22 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

Preface:  I have read the document you refer to, and I find it personally a
lot less clear than you do.  While I would welcome specific pointers when I
ask questions, more elucidation is needed for me.  Thanks for understanding=
.

More questions in-line.


> Of course, I mean "the media signaling". But that is not just about
> "in-the-wire". Current WebRTC specs mandate that the information unit
> exchanged between the JS app and the WebRTC stack must be a SDP blob.
>
> What I mean is:
>
> - The JS app uses the WebRTC API to retrieve a JS Object (or Objects)
> with transport and media info.
>
>
Retrieve from where?  Does the browser generate these at the direction of
the JS application,
or do you believe it generates them when media sources are connected.  How
are data sources
created?


> - The JS app serializes such a information in any custom way defined
> by the website developer (this can be a JSON body, XML, SDP! or
> whatever).
>
> - The JS app sends the information to the remote peer.
>
> - The JS app of the remote peer decodes the information (since it is a
> JS app designed by the same domain/website) and extracts all the
> transport/media fields.
>
> - The JS app of the remote peer creates a local PC and makes calls to
> API methods of PC for passing those transport/media fields.
>
> - ...and remove O/A mandate.
>
> But honestly, all of this is really well described and exposed in the
> draft:
>
>
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00
>
>
>
>
None of the above appears to touch on the API which uses SDP in the current
design; that's between the JS and the browser.

>
>
> >> The receiver JS app parses the information (i.e. converts from JSON to
> >> an Object), extracts the media/transport fields and pass them to the
> >> PeerConnection via API methods.
> >>
> >>
> >> It is really so hard? this is how WWW works (in contrast to how SIP
> >> works).
> >>
> >
> > As has been repeatedly mentioned, you have a peer communication here th=
at
> > has to be implemented by the browser.  That means the downloaded
> javascript
> > application must tell a browser what transports it needs so that it can
> > create the appropriate connections.  If it is using RTP/UDP, it ends up
> > needing to tell the browser what enough detail to construct the RTP/UPD
> > streams.  What payload format, for example, is needed.  As I said befor=
e,
> > that means that having no standard format across the API between the JS
> > application and the browser has very poor scaling characteristics.  You
> > might replace SDP as the interface between them, but not specifying
> anything
> > there doesn't seem to work, at least in my personal opinion.
>
> I invite you to read the draft which describes how your concerns can
> be perfectly accomplished without mandating SDP or any other format:
>
>
> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationa=
le-00
>
>
Honestly, I have read it and I don't see how you avoid mandating some
standard
interface between the JS and browser.  That needn't be SDP, but if it is
causing the
two sides to negotiate candidate transports and emit RTP, it's going to be
hard going
to avoid it being SDPNG.

Ted


>
>
> Best regards.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>

--e89a8f3bb04ffdbdaf04e0773ee9
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Content-Transfer-Encoding: quoted-printable

On Mon, Jul 1, 2013 at 10:22 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</=
span> wrote:<br><div class=3D"gmail_quote"><div><br>Preface:=A0 I have read=
 the document you refer to, and I find it personally a lot less clear than =
you do.=A0 While I would welcome specific pointers when I ask questions, mo=
re elucidation is needed for me.=A0 Thanks for understanding.<br>
<br>More questions in-line.<br>=A0<br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
Of course, I mean &quot;the media signaling&quot;. But that is not just abo=
ut<br>

&quot;in-the-wire&quot;. Current WebRTC specs mandate that the information =
unit<br>
exchanged between the JS app and the WebRTC stack must be a SDP blob.<br>
<br>
What I mean is:<br>
<br>
- The JS app uses the WebRTC API to retrieve a JS Object (or Objects)<br>
with transport and media info.<br>
<br></blockquote><div><br>Retrieve from where?=A0 Does the browser generate=
 these at the direction of the JS application,<br>or do you believe it gene=
rates them when media sources are connected.=A0 How are data sources<br>cre=
ated?<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex">
- The JS app serializes such a information in any custom way defined<br>
by the website developer (this can be a JSON body, XML, SDP! or<br>
whatever).<br>
<br>
- The JS app sends the information to the remote peer.<br>
<br>
- The JS app of the remote peer decodes the information (since it is a<br>
JS app designed by the same domain/website) and extracts all the<br>
transport/media fields.<br>
<br>
- The JS app of the remote peer creates a local PC and makes calls to<br>
API methods of PC for passing those transport/media fields.<br>
<br>
- ...and remove O/A mandate.<br>
<br>
But honestly, all of this is really well described and exposed in the draft=
:<br>
<br>
<a href=3D"http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-ap=
i-rationale-00" target=3D"_blank">http://tools.ietf.org/html/draft-raymond-=
rtcweb-webrtc-js-obj-api-rationale-00</a><br>
<div class=3D"im"><br>
<br>
<br></div></blockquote><div><br>None of the above appears to touch on the A=
PI which uses SDP in the current design; that&#39;s between the JS and the =
browser. <br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im">
<br>
<br>
&gt;&gt; The receiver JS app parses the information (i.e. converts from JSO=
N to<br>
&gt;&gt; an Object), extracts the media/transport fields and pass them to t=
he<br>
&gt;&gt; PeerConnection via API methods.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; It is really so hard? this is how WWW works (in contrast to how SI=
P<br>
&gt;&gt; works).<br>
&gt;&gt;<br>
&gt;<br>
&gt; As has been repeatedly mentioned, you have a peer communication here t=
hat<br>
&gt; has to be implemented by the browser. =A0That means the downloaded jav=
ascript<br>
&gt; application must tell a browser what transports it needs so that it ca=
n<br>
&gt; create the appropriate connections. =A0If it is using RTP/UDP, it ends=
 up<br>
&gt; needing to tell the browser what enough detail to construct the RTP/UP=
D<br>
&gt; streams. =A0What payload format, for example, is needed. =A0As I said =
before,<br>
&gt; that means that having no standard format across the API between the J=
S<br>
&gt; application and the browser has very poor scaling characteristics. =A0=
You<br>
&gt; might replace SDP as the interface between them, but not specifying an=
ything<br>
&gt; there doesn&#39;t seem to work, at least in my personal opinion.<br>
<br>
</div>I invite you to read the draft which describes how your concerns can<=
br>
be perfectly accomplished without mandating SDP or any other format:<br>
<br>
<a href=3D"http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-ap=
i-rationale-00" target=3D"_blank">http://tools.ietf.org/html/draft-raymond-=
rtcweb-webrtc-js-obj-api-rationale-00</a><br>
<div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></blockquote><div><=
br>Honestly, I have read it and I don&#39;t see how you avoid mandating som=
e standard<br>interface between the JS and browser.=A0 That needn&#39;t be =
SDP, but if it is causing the<br>
two sides to negotiate candidate transports and emit RTP, it&#39;s going to=
 be hard going<br>to avoid it being SDPNG.<br><br>Ted<br>=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">
<br>
<br>
Best regards.<br>
<br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div></div></blockquote></div><br>

--e89a8f3bb04ffdbdaf04e0773ee9--

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References: <51CA6FEE.6030702@hookflash.com> <1447FA0C20ED5147A1AA0EF02890A64B1C3093E0@ESESSMB209.ericsson.se> <CALiegfmsahUM6w00thQSCu3CpKse2C3LKSb1LzkwodNgKTOK0g@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C309655@ESESSMB209.ericsson.se> <CALiegfnAN9SJx0nLyegFJoQscG-18Gs4umd-pe7S3y6xREpByQ@mail.gmail.com> <CA+9kkMC5FoxKSz7DuHxN8cEO=0PDpoAGrLshpFmnDe3gco06cw@mail.gmail.com> <CALiegfnpj+nBdhn0g8A7iTdXKZdqvwyffjdLAOuM_qQkhTuKew@mail.gmail.com> <CA+9kkMA1PhoJB9qGUnXDUHRJNac5OszM4o5O7-6aCU-ahszMNw@mail.gmail.com> <CALiegfkei-dS-EHt2vqofft_uhKNz+UuK7R5LX3bgDkh=0ZwqQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30979C@ESESSMB209.ericsson.se>
Date: Mon, 1 Jul 2013 14:24:24 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On Mon, Jul 1, 2013 at 1:33 PM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> What if first browser generated a JS Object that said (for this example
> we assume one single audio track) it preferred to send audio using
> Opus2.0 but could also do Opus and G.711. If the second browser (which
> has not yet been upgraded) could only decode Opus and G.711, would it
> not have to tell the first browser about this?
>
> Are you perhaps assuming a capability exchange of some sort before any
> media can be set up?
>
>
Any API that we propose would include methods to expose browser
capabilities (ie list of available codecs, with default payload type,
payload name, clock rate, number of channels, and preferred codec
parameters). Generation of SDP c line, rtpmap, and fmtp lines from this
would be trivial, as well as generation of other media descriptions for
XMPP or other negotiation protocols.

In other words, we are not removing the need for negotiation. All we want
is to unbundle it from single offer/answer exchange with single string blob
into separate methods where transport negotiation (ie DTLS, SRTP, and ICE
candidates), mapping transport to streams (ie any sort of bundle related
mapping remote address, SSRC, payload or others), and media
encoding/decoding (ie send and receive codecs, codec parameters, payload
times, etc) are all negotiated separately and independently. This is what
is happening with the API anyway with trickle ICE and no-plan proposal
adding ability to update transport and media parameters once the initial
connection is established. All we want is to extend the same logic to the
initial connection establishment,
_____________
Roman Shpount

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<div class=3D"gmail_quote">On Mon, Jul 1, 2013 at 1:33 PM, Stefan H=E5kanss=
on LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.=
com" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">
<div class=3D"im">What if first browser generated a JS Object that said (fo=
r this example</div>
we assume one single audio track) it preferred to send audio using<br>
Opus2.0 but could also do Opus and G.711. If the second browser (which<br>
has not yet been upgraded) could only decode Opus and G.711, would it<br>
not have to tell the first browser about this?<br>
<br>
Are you perhaps assuming a capability exchange of some sort before any<br>
media can be set up?<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></blockquote><div><=
br></div><div>Any API that we propose would include methods to expose brows=
er capabilities (ie list of available codecs, with default payload type, pa=
yload name, clock rate, number of channels, and preferred codec parameters)=
. Generation of SDP c line, rtpmap, and fmtp lines from this would be trivi=
al, as well as generation of other media descriptions for XMPP or other neg=
otiation protocols.</div>
<div><br></div><div>In other words, we are not removing the need for negoti=
ation. All we want is to unbundle it from single offer/answer exchange with=
 single string blob into separate methods where transport negotiation (ie D=
TLS, SRTP, and ICE candidates), mapping transport to streams (ie any sort o=
f bundle related mapping remote address, SSRC, payload or others), and medi=
a encoding/decoding (ie send and receive codecs, codec parameters, payload =
times, etc) are all negotiated separately and independently. This is what i=
s happening with the API anyway with trickle ICE and no-plan proposal addin=
g ability to update transport and media parameters once the initial connect=
ion is established. All we want is to extend the same logic to the initial =
connection establishment,=A0</div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--089e012292e292ca2204e0775632--

From martin.thomson@gmail.com  Mon Jul  1 11:57:54 2013
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Subject: [rtcweb] New language (was: ... WebRTC JS Object API Model)
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On 1 July 2013 11:17, Ted Hardie <ted.ietf@gmail.com> wrote:
> Honestly, I have read it and I don't see how you avoid mandating some
> standard
> interface between the JS and browser.  That needn't be SDP, but if it is
> causing the
> two sides to negotiate candidate transports and emit RTP, it's going to be
> hard going
> to avoid it being SDPNG.

This is an important argument, but it's only correct up to a point.
And that is misleading.

SDP does a lot of things that an API can do in far simpler ways.  Take
BUNDLE, or the plan A/B/No question.  We all know what these need to
do, but we are encumbered by existing SDP - which is no single thing -
in trying to reach a conclusion.

Interconnectedness makes SDP hard.  Harder than it needs to be.  The
main advantages of a new form of expression come from its ability to
isolate problems. For instance, separating transport establishment was
clearly a big win for Jingle in terms of flexibility, simplicity,
capabilities.

But those examples speak directly to your point.  A new "language"
might be unencumbered in a way that might make addressing these
problems easier, it might even be more capable, but it is still just a
new way of saying the same old thing.

Offer/answer is the problem.  It's a contrivance that can be chosen or
rejected depending on how you plan to interact with the API.  That
embeds a negotiation engine into the browser, along with all the
machinery for how to negotiate everything that SDP expresses.

Negotiation is an extra, a feature that the IETF decided to add to the
API to make it "easier to use".  That wouldn't be a problem if it
weren't mandatory, but it's the only way to do anything.  (In
contrast, CU-RTC-Web provides a negotiation feature, but it's an
optional extra.)

So SDP truly becomes a problem when coupled to offer/answer.  The way
that features interact become a problem for standardization, not
because they interact with each other - interaction problems are what
engineers are for - but because they are negotiated.

And it's not the SDP negotiation that is the worst: it's the
negotiation of what it means.  We all know what it means, but I'm
certain no two people "know" SDP the same way.  That means another
sort of negotiation.  And those negotiations require that those
engineers be sent to places like the IETF to argue endlessly, or play
at politics.

--Martin

From matthew@matthew.at  Mon Jul  1 12:22:21 2013
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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> Honestly, I have read it and I don't see how you avoid mandating some stan=
dard
> interface between the JS and browser.  That needn't be SDP, but if it is c=
ausing the
> two sides to negotiate candidate transports and emit RTP, it's going to be=
 hard going
> to avoid it being SDPNG.

This lack of imagination among people designing the APIs we are supposed to u=
se is disturbing.

Have you read, just as an example, the CU-RTCWEB API? It achieves a browser A=
PI without resorting to SDP or anything like it.

If you insist on Offer/Answer *and* "opaque" blobs (that we all know people w=
ill modify anyway), then sure, SDP looks attractive... But that's just a fai=
lure to imagine a world where VoIP isn't based on SIP and SDP O/A.

Matthew Kaufman

(Sent from my iPhone)=

From radhika.r.roy.civ@mail.mil  Mon Jul  1 12:28:33 2013
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From: "Roy, Radhika R CIV USARMY (US)" <radhika.r.roy.civ@mail.mil>
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Classification: UNCLASSIFIED
Caveats: NONE

Yes, the problems with capability negotiations with SDP had been well-known
for the folks when H.323 (H.245) and SIP (SDP) were debated and when SIP RFC
3261 was created. Later on, SDP O/A RFC was created to help SDP to have some
negotiation capabilities. 

Again, SDP could not provide any mechanisms for setting up of the multipoint
conferencing call dynamically where media bridging is needed evolving from
the two party point-to-point call that does not need any media bridging.

Later on, it was felt impossible to modify SDP for multipoint conference
call dynamically. So, it was decided that let us define a conference focus
that is the central point is well known a priori to all and people will dial
to that conference focus to set up the centralized multipoint conference
call statically only in a star-like configuration. That is, we gave up the
setup of the distributed conference call with sa mesh like configuration
dynamically because of limitations of SDP signaling architecture.

Now, if we keep the option that makes the WebRTC API is not 'hardwired' to
SDP as the ONLY one as mandatory for media signaling protocol for media
negotiations, then it will open up a huge innovative opportunities for all
of us in the future to have much more richer multimedia conference
capabilities.

So, let us keep SDP as one of the many alternatives for negotiations for
signaling of media capabilities (as opposed to ONLY one as mandatory that
stops all future innovations knowing SDP's all those limitations - we cannot
block our future innovativeness for real-time multipoint multimedia
communications - future of real-time multimedia communications are yet to be
imagined through creativity of the future generations).

Best regards,
Radhika


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Martin Thomson
Sent: Monday, July 01, 2013 2:58 PM
To: Ted Hardie
Cc: rtcweb@ietf.org
Subject: [rtcweb] New language (was: ... WebRTC JS Object API Model)

On 1 July 2013 11:17, Ted Hardie <ted.ietf@gmail.com> wrote:
> Honestly, I have read it and I don't see how you avoid mandating some 
> standard interface between the JS and browser.  That needn't be SDP, 
> but if it is causing the two sides to negotiate candidate transports 
> and emit RTP, it's going to be hard going to avoid it being SDPNG.

This is an important argument, but it's only correct up to a point.
And that is misleading.

SDP does a lot of things that an API can do in far simpler ways.  Take
BUNDLE, or the plan A/B/No question.  We all know what these need to do, but
we are encumbered by existing SDP - which is no single thing - in trying to
reach a conclusion.

Interconnectedness makes SDP hard.  Harder than it needs to be.  The main
advantages of a new form of expression come from its ability to isolate
problems. For instance, separating transport establishment was clearly a big
win for Jingle in terms of flexibility, simplicity, capabilities.

But those examples speak directly to your point.  A new "language"
might be unencumbered in a way that might make addressing these problems
easier, it might even be more capable, but it is still just a new way of
saying the same old thing.

Offer/answer is the problem.  It's a contrivance that can be chosen or
rejected depending on how you plan to interact with the API.  That embeds a
negotiation engine into the browser, along with all the machinery for how to
negotiate everything that SDP expresses.

Negotiation is an extra, a feature that the IETF decided to add to the API
to make it "easier to use".  That wouldn't be a problem if it weren't
mandatory, but it's the only way to do anything.  (In contrast, CU-RTC-Web
provides a negotiation feature, but it's an optional extra.)

So SDP truly becomes a problem when coupled to offer/answer.  The way that
features interact become a problem for standardization, not because they
interact with each other - interaction problems are what engineers are for -
but because they are negotiated.

And it's not the SDP negotiation that is the worst: it's the negotiation of
what it means.  We all know what it means, but I'm certain no two people
"know" SDP the same way.  That means another sort of negotiation.  And those
negotiations require that those engineers be sent to places like the IETF to
argue endlessly, or play at politics.

--Martin
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb



Classification: UNCLASSIFIED
Caveats: NONE


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Date: Mon, 1 Jul 2013 13:11:23 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: Matthew Kaufman <matthew@matthew.at>
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On Mon, Jul 1, 2013 at 12:22 PM, Matthew Kaufman <matthew@matthew.at> wrote=
:

>
> > Honestly, I have read it and I don't see how you avoid mandating some
> standard
> > interface between the JS and browser.  That needn't be SDP, but if it i=
s
> causing the
> > two sides to negotiate candidate transports and emit RTP, it's going to
> be hard going
> > to avoid it being SDPNG.
>
> This lack of imagination among people designing the APIs we are supposed
> to use is disturbing.
>
> Have you read, just as an example, the CU-RTCWEB API? It achieves a
> browser API without resorting to SDP or anything like it.
>
>
I have read it, though I see from the date on it that it has been updated
since the last time I did so.  You are right that this does not look
anything like SDP, and it is therefore not clear to apply the moniker SDPNG
to it.  But CU-RTCWEB also contains elements that are standardized to meet
similar requirements, which I believe I=F1aki is arguing is not needed.  To
put it in CU-RTCWEB terms, I don't see how not having a standardized
version of something like RtpStreamDescription,  RtpExtension, or
CodecDescription scales, since the result is that browser must understand
whatever approach the JS application chooses to use for those.

To restate this:  having the JS application to JS application API surface
not be standardized works because the private agreements can be expressed
in the application; having the JS application to browser interface be not
be standardized doesn't seem to me to work at scale. It could be done by
having an independent browser extension for every site that wants to use a
specialized API, but that hardly seems to meet the goals of the group.

Once again, no hats or company swag on,

Ted


If you insist on Offer/Answer *and* "opaque" blobs (that we all know people
> will modify anyway), then sure, SDP looks attractive... But that's just a
> failure to imagine a world where VoIP isn't based on SIP and SDP O/A.
>
> Matthew Kaufman
>
> (Sent from my iPhone)

--089e013c6c122b6d7f04e078d5e4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

On Mon, Jul 1, 2013 at 12:22 PM, Matthew Kaufman <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&=
gt;</span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
<div class=3D"im"><br>
&gt; Honestly, I have read it and I don&#39;t see how you avoid mandating s=
ome standard<br>
&gt; interface between the JS and browser. =A0That needn&#39;t be SDP, but =
if it is causing the<br>
&gt; two sides to negotiate candidate transports and emit RTP, it&#39;s goi=
ng to be hard going<br>
&gt; to avoid it being SDPNG.<br>
<br>
</div>This lack of imagination among people designing the APIs we are suppo=
sed to use is disturbing.<br>
<br>
Have you read, just as an example, the CU-RTCWEB API? It achieves a browser=
 API without resorting to SDP or anything like it.<br>
<br></blockquote><div><br>I have read it, though I see from the date on it =
that it has been updated since the last time I did so. =A0You are right tha=
t this does not look anything like SDP, and it is therefore not clear to ap=
ply the moniker SDPNG to it. =A0But CU-RTCWEB also contains elements that a=
re standardized to meet similar requirements, which I believe I=F1aki is ar=
guing is not needed. =A0To put it in CU-RTCWEB terms, I don&#39;t see how n=
ot having a standardized version of something like RtpStreamDescription, =
=A0RtpExtension, or CodecDescription scales, since the result is that brows=
er must understand whatever approach the JS application chooses to use for =
those.<br>
<br>To restate this: =A0having the JS application to JS application API sur=
face not be standardized works because the private agreements can be expres=
sed in the application; having the JS application to browser interface be n=
ot be standardized doesn&#39;t seem to me to work at scale. It could be don=
e by having an independent browser extension for every site that wants to u=
se a specialized API, but that hardly seems to meet the goals of the group.=
<br>
<br>Once again, no hats or company swag on,<code><br><br>Ted<br></code><cod=
e></code>=A0<br><br></div><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
If you insist on Offer/Answer *and* &quot;opaque&quot; blobs (that we all k=
now people will modify anyway), then sure, SDP looks attractive... But that=
&#39;s just a failure to imagine a world where VoIP isn&#39;t based on SIP =
and SDP O/A.<br>

<br>
Matthew Kaufman<br>
<br>
(Sent from my iPhone)</blockquote></div><br>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 1 Jul 2013 22:37:54 +0200
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> What if first browser generated a JS Object that said (for this example
> we assume one single audio track) it preferred to send audio using
> Opus2.0 but could also do Opus and G.711. If the second browser (which
> has not yet been upgraded) could only decode Opus and G.711, would it
> not have to tell the first browser about this?
>
> Are you perhaps assuming a capability exchange of some sort before any
> media can be set up?

The fact that a capability exchange or media signaling negotiation
exists does not mean that it must be based on a opaque blob which is
unmanageable at JS level. The JS app should be able to understand
*what* exactly it is sending in-the-wire, and for that purpose a
opaque blob is not a ellegant solution. Instead a JS Object based API
(like *any* WWW API) makes sense. And then the JS app can decide how
to encode/serialize/whatever such a media information in-the-wire (and
yes, it can be into a SDP, but a SDP created by the JS library because
the author chose it, while others can use whatever format or send the
information "splitted" into different requests or whatever.

Of couse the SDP O/A model is a valid model, and of course SIP is a
valid model (which carries the SDP exchange), but that is not the only
one. IMHO the IETF/W3C should not mandate like-SIP-models for every
future RTC applications running in the web.

Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Ted Hardie <ted.ietf@gmail.com>:
> On Mon, Jul 1, 2013 at 10:22 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>
> Preface:  I have read the document you refer to, and I find it personally=
 a
> lot less clear than you do.  While I would welcome specific pointers when=
 I
> ask questions, more elucidation is needed for me.  Thanks for understandi=
ng.

Sure Ted, and thanks a lot for reading the draft.



>> Of course, I mean "the media signaling". But that is not just about
>> "in-the-wire". Current WebRTC specs mandate that the information unit
>> exchanged between the JS app and the WebRTC stack must be a SDP blob.
>>
>> What I mean is:
>>
>> - The JS app uses the WebRTC API to retrieve a JS Object (or Objects)
>> with transport and media info.
>>
>
> Retrieve from where?

I don't want to design an API in a few lines because I may be 100%
wrong, but (very) basically:

The JS app:
- Calls getUserMedia().
- Creates a PC instance.
- Provides the PC with the MediaTracks obtained from getUserMedia().
- Calls to some API method*s* in PC to retrieve ICE transport
information (which is obtained as JS Object, with attributes
representing the corresponding fields of a ICE "line").
- Calls to some other API method*s* in PC to get the list of supported
codecs and tells the PC, via another PC API method, the preference
order.
- Calls to some other API method*s* in PC to get local MediaStream
and/or MediaTracks instances (i.e. to render local video).
- Calls to some other API method*s* in PC to get crypto/DTLS
information (an Object with the appropriate attributes fully defining
such an information).
- ...and so on.

And the the JS app serializes all the retrieved information and
encodes/serializes/whatever it into a JSON body, XML, etc etc, and
sends it in-the-wire (or splittes it into various chunks of
information to be send separately).



> Does the browser generate these at the direction of
> the JS application,
> or do you believe it generates them when media sources are connected.  Ho=
w
> are data sources
> created?

I don't understand the question. In the current SDP-based model you
get all the media/transport information when you get the SDP
(SessionDescription instance) from the PC, right? and then you send it
in-the-wire and ICE stuff starts.

Some here, but instead of passing a horrile opaque blob, let's use JS Objec=
ts.

NOTE: This is just one of the various changes/improvements the draft
describes. It is not all about the interface between the JS layer and
the WebRTC stack in the browser.




>> - The JS app serializes such a information in any custom way defined
>> by the website developer (this can be a JSON body, XML, SDP! or
>> whatever).
>>
>> - The JS app sends the information to the remote peer.
>>
>> - The JS app of the remote peer decodes the information (since it is a
>> JS app designed by the same domain/website) and extracts all the
>> transport/media fields.
>>
>> - The JS app of the remote peer creates a local PC and makes calls to
>> API methods of PC for passing those transport/media fields.
>>
>> - ...and remove O/A mandate.
>>
>> But honestly, all of this is really well described and exposed in the
>> draft:
>>
>>
>> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-ration=
ale-00
>>
>>
>>
>
> None of the above appears to touch on the API which uses SDP in the curre=
nt
> design; that's between the JS and the browser.

In fact the proposed API between the JS and the browser is what
encourages that the JS app can send whatever it wants in-the-wire.

Anyhow I don't fully understand what you mean with "the API which uses
SDP in the current design". Do you mean setLocalDescription (which is
provided with an opaque blob) and getLocalDescription (which provides
an opaque blob)?






>> http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-ration=
ale-00
>>
>
> Honestly, I have read it and I don't see how you avoid mandating some
> standard
> interface between the JS and browser.  That needn't be SDP,

Of course we need a standard interface (this is: API) between the JS
and browser. That's why we are proposing a "JS Object API rationale".
But we need a *real* JS API (which means functions that returns JS
Objects and are provided with JS Objects) rather than the current API
that just works by passing an opaque/unmanageable blob between JS and
browser (in both directions).


Best regards.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Ted Hardie <ted.ietf@gmail.com>:
> You are right that this does not look anything like SDP, and it is theref=
ore
> not clear to apply the moniker SDPNG to it.  But CU-RTCWEB also contains
> elements that are standardized to meet similar requirements, which I beli=
eve
> I=C3=B1aki is arguing is not needed.

May be I explained it wrong, but as said in my previous mail, of
course some kind of *REAL* JS Object-based API is required to
interface between JS and browser.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] New Draft - WebRTC JS Object API Model
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On 2013-07-01 20:24, Roman Shpount wrote:=0A=
> On Mon, Jul 1, 2013 at 1:33 PM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     What if first browser generated a JS Object that said (for this examp=
le=0A=
>     we assume one single audio track) it preferred to send audio using=0A=
>     Opus2.0 but could also do Opus and G.711. If the second browser (whic=
h=0A=
>     has not yet been upgraded) could only decode Opus and G.711, would it=
=0A=
>     not have to tell the first browser about this?=0A=
>=0A=
>     Are you perhaps assuming a capability exchange of some sort before an=
y=0A=
>     media can be set up?=0A=
>=0A=
>=0A=
> Any API that we propose would include methods to expose browser=0A=
> capabilities (ie list of available codecs, with default payload type,=0A=
> payload name, clock rate, number of channels, and preferred codec=0A=
> parameters). Generation of SDP c line, rtpmap, and fmtp lines from this=
=0A=
> would be trivial, as well as generation of other media descriptions for=
=0A=
> XMPP or other negotiation protocols.=0A=
>=0A=
> In other words, we are not removing the need for negotiation. All we=0A=
> want is to unbundle it from single offer/answer exchange with single=0A=
> string blob into separate methods where transport negotiation (ie DTLS,=
=0A=
> SRTP, and ICE candidates), mapping transport to streams (ie any sort of=
=0A=
> bundle related mapping remote address, SSRC, payload or others), and=0A=
> media encoding/decoding (ie send and receive codecs, codec parameters,=0A=
> payload times, etc) are all negotiated separately and independently.=0A=
=0A=
Roman, thanks for clarifying. Personally (wearing no hat) I think =0A=
splitting things up makes a lot of sense. In fact, our first prototype =0A=
browser (back in early 2011) was sort of implemented this way: each =0A=
MediaStream was described and negotiated separately. We used SDPs for =0A=
the purpose though (each addStream lead to a separate SDP O/A setting up =
=0A=
that unidirectional MediaStream with its tracks).=0A=
=0A=
But I think we should not underestimate the work needed to describe all =0A=
the nitty gritty details using another "language" than SDP.=0A=
=0A=
> This is what is happening with the API anyway with trickle ICE and=0A=
> no-plan proposal adding ability to update transport and media parameters=
=0A=
> once the initial connection is established. All we want is to extend the=
=0A=
> same logic to the initial connection establishment,=0A=
> _____________=0A=
> Roman Shpount=0A=
=0A=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2013/7/2 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> But I think we should not underestimate the work needed to describe all
> the nitty gritty details using another "language" than SDP.

Hi Stefan,

As said multiple times we are not proposing a replacement for SDP.
Passing an opaque blob string between browser and WebRTC is not the
only solution, even more, I don't consider that an "API".

If after reading the draft you have understood that we are proposing a
replacement for SDP then we've done something wrong.

Regards.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 2 Jul 2013, at 08:21, Stefan H=E5kansson LK wrote:
>=20
> But I think we should not underestimate the work needed to describe =
all=20
> the nitty gritty details using another "language" than SDP.

Conversely we shouldn't underestimate (again) the difficulty of =
describing all the
nitty gritty details in SDP.=20

I have the sense that we may have pushed the flexibility of SDP beyond =
it's elastic limit.

I think that over the last year we have discovered that a dynamically =
programmable browser
is a fundamentally different beast from a deskphone and that what suits =
one may well not suit the other.

Tim.


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 2 Jul 2013, at 08:21, Stefan H=E5kansson LK =
wrote:</div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" color=3D"#000000"><br></font>But I think we =
should not underestimate the work needed to describe all <br>the nitty =
gritty details using another "language" than =
SDP.<br></div></blockquote></div><br><div>Conversely we shouldn't =
underestimate (again) the difficulty of describing all =
the</div><div>nitty gritty details in =
SDP.&nbsp;</div><div><br></div><div>I have the sense that we may have =
pushed the flexibility of SDP beyond it's elastic =
limit.</div><div><br></div><div>I think that over the last year we have =
discovered that a dynamically programmable browser</div><div>is a =
fundamentally different beast from a deskphone and that what suits one =
may well not suit the =
other.</div><div><br></div><div>Tim.</div><div><br></div></body></html>=

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Saverio Mascolo <saverio.mascolo@gmail.com>
Thread-Topic: [rtcweb] More H.264 vs VP8 tests
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Date: Tue, 2 Jul 2013 10:16:02 +0000
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Subject: Re: [rtcweb] More H.264 vs VP8 tests
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On 2013-07-01 19:10, Saverio Mascolo wrote:=0A=
> an evaluation of how h264 and vp8 are able to match a changing quality=0A=
> would be interesting=0A=
=0A=
Such an evaluation would be very hard to do. The H.264 standard does not =
=0A=
specify any encoder behavior, only the decoder is specified. This means =0A=
that the encoder can be implemented freely, as long as it produces a =0A=
bitstream compliant to the standard. Specifically, there is no =0A=
specification in H.264 on how to implement the rate controller. Thus the =
=0A=
rate control mechanisms of two different H.264 encoder implementations =0A=
could react differently to a change in scene complexity.=0A=
=0A=
There is a wealth of H.264 implementations on the market today. Some may =
=0A=
have implemented a better rate control mechanism than x264, some may =0A=
have implemented a worse variant. Testing x264 versus the WebM vp8 =0A=
implemenation with rate control turned on may give some information =0A=
about the relative merits of these two rate control implementations =0A=
(although the rate control gives rise to so much measurement noise so =0A=
even that is hard to do) but it says very little about how the H.264 =0A=
standard compares to the vp8 specification in situations where the scene =
=0A=
complexity varies.=0A=
=0A=
=0A=
>=0A=
>=0A=
> On Sat, Jun 29, 2013 at 3:17 PM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     On 6/25/13 12:17 PM, Harald Alvestrand wrote:=0A=
>      > Again - thanks for releasing this openly!=0A=
>      >=0A=
>      > I ran the scripts (with a few tweaks; you run on a system where sh=
 is=0A=
>      > bash, not dash, for instance), and got the same numbers within=0A=
>     +/- 0.5%=0A=
>      > (probably some binary version skew);=0A=
>=0A=
>     I re-run the scripts on another computer with another OS today, and I=
=0A=
>     get exactly the same results as Bo sent out. I noted however that if =
the=0A=
>     input clips are not cut at 10s (but used in their entire length) the=
=0A=
>     results get slightly different, but within +/- 0.5%. Can this be the=
=0A=
>     reason why you get slightly different numbers?=0A=
>=0A=
>=0A=
>      > we may have disagreements on the=0A=
>      > parameters to use, but we agree on the numbers those parameters=0A=
>     produce.=0A=
>      >=0A=
>      > (I have since modified the Google framework to include a script th=
at=0A=
>      > pulls in the sources for the needed binaries and compiles them -=
=0A=
>     if you=0A=
>      > want to make 100% sure people are working from the same sources,=
=0A=
>     you may=0A=
>      > want to rebase to a newer version of the comparision toolkit.)=0A=
>      >=0A=
>      > On 06/22/2013 03:41 PM, Bo Burman wrote:=0A=
>      >> Hi all,=0A=
>      >>=0A=
>      >> We have had a look at Google's comparison between VP8 and H.264=
=0A=
>     constrained baseline that was posted on April 3rd=0A=
>     (http://www.ietf.org/mail-archive/web/rtcweb/current/msg07028.html).=
=0A=
>     This post contains, as the one mentioned above (and if the=0A=
>     attachments make it to the list), information on the exact tools and=
=0A=
>     options used for encoding and should thus be repeatable by anyone=0A=
>     interested.=0A=
>      >>=0A=
>      >> As was already stated by others on this list, one major problem=
=0A=
>     is that Google's test involves the rate control mechanism. Typically=
=0A=
>     codecs are measured with rate control turned off, since it acts as a=
=0A=
>     huge noise on the measurement. Instead we propose to compare the=0A=
>     codecs using fixed qp-levels. The qp-level is the quantization=0A=
>     parameter that affects the rate/distortion tradeoff. Comparing using=
=0A=
>     fixed qp-levels is what has been used when benchmarking HEVC against=
=0A=
>     H.264 in the JCT-VC standardization, for instance. We are going to=0A=
>     select a codec (essentially bit stream format), not a rate control=0A=
>     mechanism: Once the codec is selected you can choose whatever rate=0A=
>     control mechanism you wish.=0A=
>      >>=0A=
>      >> We used Google's excellent framework as the baseline and changed=
=0A=
>     the parameter settings in order to make it possible to measure using=
=0A=
>     fixed qp. We used the same sequences, but limited them to the first=
=0A=
>     10 seconds since they varied from 10 seconds to minutes; this also=0A=
>     eased computation time.=0A=
>      >>=0A=
>      >> We used two H.264 encoder implementations: X264, which is an=0A=
>     open-source codec that can operate in everything from real-time to=0A=
>     slow, and JM which is the reference implementation that was used to=
=0A=
>     develop H.264. JM is very slow but attempts to be very efficient in=
=0A=
>     terms of bits per quality. The results were as follows:=0A=
>      >>=0A=
>      >> X264 baseline vs VP8: H.264 wins with 1%=0A=
>      >> JM baseline vs VP8: H.264 wins with 4%=0A=
>      >>=0A=
>      >> Running times:=0A=
>      >> X264: 1 hour 3 minutes=0A=
>      >> VP8: 2 hours 0 minutes=0A=
>      >> JM: order of magnitude slower=0A=
>      >>=0A=
>      >> It is interesting to note that the measurements are more stable=
=0A=
>     in the new test; the variance of the percentages for the sequences=0A=
>     is now around 70, down from around 700 in Google's test of April=0A=
>     3rd.  We believe this is due to the removal of the rate controller,=
=0A=
>     which acts like noise on the measurements.=0A=
>      >>=0A=
>      >> We also tried setting H.264 to constrained high (no interlace=0A=
>     and no B-pictures, compared to high). The results were then:=0A=
>      >>=0A=
>      >> X264 constrained high vs VP8: H.264 wins with 25%=0A=
>      >> JM constrained high vs VP8: H.264 wins with 24%=0A=
>      >>=0A=
>      >> We also note that the script that Google provided to calculate=0A=
>     the rate differences ("BD-rate") does not give exactly the same=0A=
>     numbers as the JCT-VC-way of calculating BD-rate. The main=0A=
>     difference is that the JM score for constrained high is better=0A=
>     (around 29%) if the JCT-VC way of calculating BD-rate is used.=0A=
>      >>=0A=
>      >> In summary we think that proper testing can conclude that there=
=0A=
>     is no clear performance advantage to any codec between VP8 and H.264=
=0A=
>     baseline. When comparing VP8 against H.264 constrained high on the=0A=
>     other hand, it seems like there is an advantage for H.264=0A=
>     constrained high.=0A=
>      >>=0A=
>      >> The attached file includes the files necessary to reproduce the=
=0A=
>     test.=0A=
>      >>=0A=
>      >> Best Regards,=0A=
>      >>=0A=
>      >> Bo Burman=0A=
>      >>=0A=
>      >>=0A=
>      >> _______________________________________________=0A=
>      >> rtcweb mailing list=0A=
>      >> rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>      >> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>      >=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
>=0A=
=0A=

From radhika.r.roy.civ@mail.mil  Tue Jul  2 03:44:40 2013
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From: "Roy, Radhika R CIV USARMY (US)" <radhika.r.roy.civ@mail.mil>
To: Tim Panton <tim@phonefromhere.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
Thread-Topic: [rtcweb] New Draft - WebRTC JS Object API Model (UNCLASSIFIED)
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Date: Tue, 2 Jul 2013 10:44:03 +0000
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model (UNCLASSIFIED)
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Classification: UNCLASSIFIED
Caveats: NONE

Inline [RRR]

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Tim Panton
Sent: Tuesday, July 02, 2013 4:31 AM
To: Stefan H=E5kansson LK
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model


On 2 Jul 2013, at 08:21, Stefan H=E5kansson LK wrote:

=09
	But I think we should not underestimate the work needed to describe
all=20
	the nitty gritty details using another "language" than SDP.
=09


Conversely we shouldn't underestimate (again) the difficulty of =
describing
all the nitty gritty details in SDP.=20

I have the sense that we may have pushed the flexibility of SDP beyond =
it's
elastic limit.

[<RRR] SDP has been and is basically a session "description" protocol, =
not a
session "negotiation" protocol unless and until a new method like UPDATE =
and
O/A have added to increase SDP's "elasticity" to perform a crude type of
negotiations with one dimensional intelligence. [</RRR>]

I think that over the last year we have discovered that a dynamically
programmable browser is a fundamentally different beast from a deskphone =
and
that what suits one may well not suit the other.

Tim.




Classification: UNCLASSIFIED
Caveats: NONE


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Date: Tue, 02 Jul 2013 08:22:58 -0400
From: Robin Raymond <robin@hookflash.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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You are absolutely, there are  a lot of details. But splitting the 
details into logical separate components reduces the overall complexity 
greatly. Plus many of those details are only needed because of a lack 
ability to properly describe the concept of "streams" at the lower level 
RTP layer (which really only supports tracks at best, and lacks 
sufficient details to coordinate any information between them to 
reassembly remotely as a "stream"). So we are left in a situation where 
the high layers must coordinate a lot of information to be capable of 
reassembling the streams when they really should be just negotiating the 
capabilities of the RTP layer itself.

For example, the only thing that should be needed is a flag that says 
FEC is available in the signaling for the the RTP layer. If it is set 
then that would imply the engine supports this feature and thus FEC 
streams may be created by the sending party. Alas, because RTP lacks 
sufficient depth for such concepts like FEC, you have to signal it. RTP 
doesn't even have enough information to reassemble tracks into stream, 
let allow do things like FEC between them.

Having said that, by making a scoped object that understands there is 
some definition of what a stream means needs to be describes, but 
perhaps that the "future' may not require such crazy definitions like 
all the stream/ssrc/track mappings, it allows for a short/long-term 
temporary situation where stream definitions are passed until we are 
able move to a model where that is no longer needed. When a theoretical 
next-gen RTP transport is capable of sending entire streams and 
reassembling them without explicit signaling from an upper layer, we can 
stop defining all that mapping stuff in signaling and remove that 
complexity.

Basically, it's about scoping the objects, and limiting the requirements 
needed to exchange per object, and defining exactly what is 
expected/required in those definitions and allowing for future 
alternative objects that could replace current ones (if they should 
become available). The other important aspect is removing the definition 
how this information must be coordinated and exchanged to allow for a 
variety of signaling scenarios and alternative state machines. And 
keeping it relatively simple is important for the simple use cases too 
so the burden is low. We are well on our way to producing such a draft 
in short order that does exactly that.

All we ask is that people keep an open mind. We have no desire to 
prevent SDP or offer/answer, just not mandate its usage as an API model.

Make sense?

-Robin

> Tim Panton <mailto:tim@phonefromhere.com>
> 2 July, 2013 4:31 AM
>
> On 2 Jul 2013, at 08:21, Stefan Håkansson LK wrote:
>
> Conversely we shouldn't underestimate (again) the difficulty of 
> describing all the
> nitty gritty details in SDP.
>
> I have the sense that we may have pushed the flexibility of SDP beyond 
> it's elastic limit.
>
> I think that over the last year we have discovered that a dynamically 
> programmable browser
> is a fundamentally different beast from a deskphone and that what 
> suits one may well not suit the other.
>
> Tim.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<html><head>
<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
</head><body bgcolor="#FFFFFF" text="#000000"><br>
You are absolutely, there are&nbsp; a lot of details. But splitting the 
details into logical separate components reduces the overall complexity 
greatly. Plus many of those details are only needed because of a lack 
ability to properly describe the concept of "streams" at the lower level
 RTP layer (which really only supports tracks at best, and lacks 
sufficient details to coordinate any information between them to 
reassembly remotely as a "stream"). So we are left in a situation where 
the high layers must coordinate a lot of information to be capable of 
reassembling the streams when they really should be just negotiating the
 capabilities of the RTP layer itself.<br>
<br>
For example, the only thing that should be needed is a flag that says 
FEC is available in the signaling for the the RTP layer. If it is set 
then that would imply the engine supports this feature and thus FEC 
streams may be created by the sending party. Alas, because RTP lacks 
sufficient depth for such concepts like FEC, you have to signal it. RTP 
doesn't even have enough information to reassemble tracks into stream, 
let allow do things like FEC between them.<br>
<br>
Having said that, by making a scoped object that understands there is 
some definition of what a stream means needs to be describes, but 
perhaps that the "future' may not require such crazy definitions like 
all the stream/ssrc/track mappings, it allows for a short/long-term 
temporary situation where stream definitions are passed until we are 
able move to a model where that is no longer needed. When a theoretical 
next-gen RTP transport is capable of sending entire streams and 
reassembling them without explicit signaling from an upper layer, we can
 stop defining all that mapping stuff in signaling and remove that 
complexity.<br>
<br>
Basically, it's about scoping the objects, and limiting the requirements
 needed to exchange per object, and defining exactly what is 
expected/required in those definitions and allowing for future 
alternative objects that could replace current ones (if they should 
become available). The other important aspect is removing the definition
 how this information must be coordinated and exchanged to allow for a 
variety of signaling scenarios and alternative state machines. And 
keeping it relatively simple is important for the simple use cases too 
so the burden is low. We are well on our way to producing such a draft 
in short order that does exactly that.<br>
<br>
All we ask is that people keep an open mind. We have no desire to 
prevent SDP or offer/answer, just not mandate its usage as an API model.<br>
<br>
Make sense?<br>
<br>
-Robin<br>
<br>
<blockquote style="border: 0px none;" 
cite="mid:E6230622-C447-44B4-8F7B-108760792A16@phonefromhere.com" 
type="cite">
  <div style="margin:30px 25px 10px 25px;" class="__pbConvHr"><div 
style="display:table;width:100%;border-top:1px solid 
#EDEEF0;padding-top:5px"> 	<div 
style="display:table-cell;vertical-align:middle;padding-right:6px;"><img
 photoaddress="tim@phonefromhere.com" photoname="Tim Panton" 
src="cid:part1.02020900.02090603@hookflash.com" 
name="postbox-contact.jpg" height="25px" width="25px"></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;width:100%">
   	<a moz-do-not-send="true" href="mailto:tim@phonefromhere.com" 
style="color:#737F92 
!important;padding-right:6px;font-weight:bold;text-decoration:none 
!important;">Tim Panton</a></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;">   
  <font color="#9FA2A5"><span style="padding-left:6px">2 July, 2013 4:31
 AM</span></font></div></div></div>
  <div style="color:#888888;margin-left:24px;margin-right:24px;" 
__pbrmquotes="true" class="__pbConvBody"><br><div><div>On 2 Jul 2013, at
 08:21, Stefan H&aring;kansson LK wrote:</div></div><br><div>Conversely we 
shouldn't underestimate (again) the difficulty of describing all the</div><div>nitty
 gritty details in SDP.&nbsp;</div><div><br></div><div>I have the sense that 
we may have pushed the flexibility of SDP beyond it's elastic limit.</div><div><br></div><div>I
 think that over the last year we have discovered that a dynamically 
programmable browser</div><div>is a fundamentally different beast from a
 deskphone and that what suits one may well not suit the other.</div><div><br></div><div>Tim.</div><div><br></div><div>_______________________________________________<br>rtcweb
 mailing list<br><a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></div></div>
  <br>
</blockquote>
</body></html>

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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: New Version Notification for draft-martinsen-mmusic-malice-00.txt
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Subject: [rtcweb] New Version Notification for draft-martinsen-mmusic-malice-00.txt
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Hi,

We submitted a new draft called MALICE today. It is all about smarter appli=
cation/network communication. Se abstract for some details. Read the whole =
draft to get a lot more details.

>=20
> A new version of I-D, draft-martinsen-mmusic-malice-00.txt
> has been successfully submitted by Reinaldo Penno and posted to the
> IETF repository.
>=20
> Filename:	 draft-martinsen-mmusic-malice
> Revision:	 00
> Title:		 Meta-data Attribute signaLling with ICE
> Creation date:	 2013-07-02
> Group:		 Individual Submission
> Number of pages: 33
> URL:             http://www.ietf.org/internet-drafts/draft-martinsen-mmus=
ic-malice-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-martinsen-mmusic-m=
alice
> Htmlized:        http://tools.ietf.org/html/draft-martinsen-mmusic-malice=
-00
>=20
>=20
> Abstract:
>  It can be useful for applications to provide flow metadata
>  information to on-path devices to influence flow treatment in the
>  network.  Provided that the network is able to provide useful
>  feedback, this can also influence path selection if an application
>  have multiple flow paths to choose from.
>=20
>  This draft describes how this can be achieved by adding metadata to
>  the STUN packets sent during the ICE connectivity checks or a
>  slightly modified version of the keep-alive mechanism.  Devices on
>  the media path can use the metadata information to prioritize the
>  flow, perform traffic engineering, or provide network analytics and
>  notifications as requested by the endpoints.  On-path devices can
>  append or modify the existing metadata information in the STUN/ICE
>  messages to enable feedback to other on-path devices or the
>  applications in both ends of the media session.
>=20
>  This document describes a framework mechanism for how such metadata
>  can be transported by STUN when ICE is in use and it covers the
>  endpoint and on path device processing.  The functionality described
>  here is referred to as MALICE.

.-.
P=E5l-Erik=

From jmillan@aliax.net  Tue Jul  2 09:04:07 2013
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Subject: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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Hi,

Please, let me know how this normal use case could be solved with the
current API.

Alice and Bob have accessed to a Web page offering a basic WebRTC
application   that they use to speak and see each other. They are in the
middle of a media session right now.

Alice presses the "Video" button in her web application. A SDP offer with
video description is sent to Bob:

Bob doesn't want to see her, so his web application rejects the media by
using any valid method for that:
-- a=3Dinactive
-- m=3Dvideo 0

Alice web application received a SDP answer which was consumed successfully
by the RTCPeerConnection. Alice thinks she is sharing her video, but she is
not.

The application "Video" button is disabled since the video is already being
shared (theoretically)

The application "self-Video" HTML5 video element displays Alice's face. She
thinks that Bob is seeing her face too, but he did indeed rejected so.

I hope this common scenario can be achieved somehow with the current API. I
can't find how to do it though.

Thanks in advance

--=20
Jos=C3=A9 Luis Mill=C3=A1n

--001a11c3e45241576e04e0897e0b
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<div dir=3D"ltr">Hi,<div><br></div><div>Please, let me know how this normal=
 use case could be solved with the current API.</div><div><br></div><div st=
yle>Alice and Bob have accessed to a Web page offering a basic WebRTC appli=
cation =C2=A0 that they use to speak and see each other. They are in the mi=
ddle of a media session right now.</div>
<div><br></div><div style>Alice presses the &quot;Video&quot; button in her=
 web application. A SDP offer with video description is sent to Bob:</div><=
div style><br></div><div style>Bob doesn&#39;t want to see her, so his web =
application rejects the media by using any valid method for that:</div>
<div style>-- a=3Dinactive</div><div style>-- m=3Dvideo 0</div><div style><=
br></div><div style>Alice web application received a SDP answer which was c=
onsumed successfully by the RTCPeerConnection. Alice thinks she is sharing =
her video, but she is not.</div>
<div style><br></div><div style>The application &quot;Video&quot; button is=
 disabled since the video is already being shared (theoretically)</div><div=
 style><br></div><div style>The application &quot;self-Video&quot; HTML5 vi=
deo element displays Alice&#39;s face. She thinks that Bob is seeing her fa=
ce too, but he did indeed rejected so.</div>
<div style><br></div><div>I hope this common scenario can be achieved someh=
ow with the current API. I can&#39;t find how to do it though.</div><div><b=
r></div><div>Thanks in advance<br clear=3D"all"><div><br></div>-- <br>Jos=
=C3=A9 Luis Mill=C3=A1n
</div></div>

--001a11c3e45241576e04e0897e0b--

From enrico.marocco@telecomitalia.it  Tue Jul  2 09:14:06 2013
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On 7/2/13 6:04 PM, Jos=C3=A9 Luis Mill=C3=A1n wrote:
> Hi,
>=20
> Please, let me know how this normal use case could be solved with the
> current API.

Just off the top of my head:

Audio-only call establishment:

A --{"action":"CALL","sdp":"v=3D0..."}-> B
B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A

Rejected audio+video upgrade proposal:

A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
B --{"action":"REJECT UPGRADE"}-> A

Alice rolls back and removes local video display.

I may be missing something, but doesn't look like rocket science to me...=


Enrico



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To: Enrico Marocco <enrico.marocco@telecomitalia.it>
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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Thanks Enrico,

Yes, you are right. I was missing a detail.

Now imagine you are not using your own wire protocol but a standard one,
where accepting the request doesn't imply accepting or rejecting the SDP
offer, since SDP has its own methods to accept or reject the call as
commented before.

A --{"action":"CALL","sdp":"v=3D0..."}-> B
B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A

A --{"action":"CALL","sdp":"v=3D0..."}-> B
B --{"action":"ACCEPT CALL","sdp":"v=3D0\r\nm=3Dvideo 0..."}-> A



2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it>

> On 7/2/13 6:04 PM, Jos=C3=A9 Luis Mill=C3=A1n wrote:
> > Hi,
> >
> > Please, let me know how this normal use case could be solved with the
> > current API.
>
> Just off the top of my head:
>
> Audio-only call establishment:
>
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>
> Rejected audio+video upgrade proposal:
>
> A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
> B --{"action":"REJECT UPGRADE"}-> A
>
> Alice rolls back and removes local video display.
>
> I may be missing something, but doesn't look like rocket science to me...
>
> Enrico
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


--=20
Jos=C3=A9 Luis Mill=C3=A1n

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<div dir=3D"ltr">Thanks Enrico,<br><div class=3D"gmail_extra"><br></div><di=
v class=3D"gmail_extra">Yes, you are right. I was missing a detail.</div><d=
iv class=3D"gmail_extra"><br>Now imagine you are not using your own wire pr=
otocol but a standard one, where accepting the request doesn&#39;t imply ac=
cepting or rejecting the SDP offer, since SDP has its own methods to accept=
 or reject the call as commented before.</div>
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">A --{&quot;=
action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quot;}-&gt; B<=
br>B --{&quot;action&quot;:&quot;ACCEPT CALL&quot;,&quot;sdp&quot;:&quot;v=
=3D0...&quot;}-&gt; A<br>
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">A --{=
&quot;action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quot;}-&=
gt; B<br>B --{&quot;action&quot;:&quot;ACCEPT CALL&quot;,&quot;sdp&quot;:&q=
uot;v=3D0\r\nm=3Dvideo 0...&quot;}-&gt; A<br>
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra" style=
><br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">2013/7=
/2 Enrico Marocco <span dir=3D"ltr">&lt;<a href=3D"mailto:enrico.marocco@te=
lecomitalia.it" target=3D"_blank">enrico.marocco@telecomitalia.it</a>&gt;</=
span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/2/13 6:04 PM, Jos=C3=A9 Luis Mill=
=C3=A1n wrote:<br>

&gt; Hi,<br>
&gt;<br>
&gt; Please, let me know how this normal use case could be solved with the<=
br>
&gt; current API.<br>
<br>
</div>Just off the top of my head:<br>
<br>
Audio-only call establishment:<br>
<br>
A --{&quot;action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quo=
t;}-&gt; B<br>
B --{&quot;action&quot;:&quot;ACCEPT CALL&quot;,&quot;sdp&quot;:&quot;v=3D0=
...&quot;}-&gt; A<br>
<br>
Rejected audio+video upgrade proposal:<br>
<br>
A --{&quot;action&quot;:&quot;UPGRADE&quot;,&quot;sdp&quot;:&quot;v=3D0...&=
quot;}-&gt; B<br>
B --{&quot;action&quot;:&quot;REJECT UPGRADE&quot;}-&gt; A<br>
<br>
Alice rolls back and removes local video display.<br>
<br>
I may be missing something, but doesn&#39;t look like rocket science to me.=
..<br>
<span class=3D""><font color=3D"#888888"><br>
Enrico<br>
<br>
<br>
</font></span><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>Jos=C3=
=A9 Luis Mill=C3=A1n
</div></div>

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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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2013/7/2 Jos=C3=A9 Luis Mill=C3=A1n <jmillan@aliax.net>

> Thanks Enrico,
>
> Yes, you are right. I was missing a detail.
>
> Now imagine you are not using your own wire protocol but a standard one,
> where accepting the request doesn't imply accepting or rejecting the SDP
> offer, since SDP has its own methods to accept or reject the call as
> commented before.
>

I meant 'accept or reject the SDP offer or part of it'  where is said
'accept or reject the call'

>
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0\r\nm=3Dvideo 0..."}-> A
>
>
>
> 2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it>
>
>> On 7/2/13 6:04 PM, Jos=C3=A9 Luis Mill=C3=A1n wrote:
>> > Hi,
>> >
>> > Please, let me know how this normal use case could be solved with the
>> > current API.
>>
>> Just off the top of my head:
>>
>> Audio-only call establishment:
>>
>> A --{"action":"CALL","sdp":"v=3D0..."}-> B
>> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>>
>> Rejected audio+video upgrade proposal:
>>
>> A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
>> B --{"action":"REJECT UPGRADE"}-> A
>>
>> Alice rolls back and removes local video display.
>>
>> I may be missing something, but doesn't look like rocket science to me..=
.
>>
>> Enrico
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> --
> Jos=C3=A9 Luis Mill=C3=A1n
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

--089e0129420c31d49c04e089ed13
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">2013/7/2 Jos=C3=A9 Luis Mill=C3=A1n <span dir=3D"ltr">&lt;<a href=
=3D"mailto:jmillan@aliax.net" target=3D"_blank">jmillan@aliax.net</a>&gt;</=
span><br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr">Thanks Enrico,<br><div class=3D"gmail_extra"><br></div><di=
v class=3D"gmail_extra">Yes, you are right. I was missing a detail.</div><d=
iv class=3D"gmail_extra"><br>Now imagine you are not using your own wire pr=
otocol but a standard one, where accepting the request doesn&#39;t imply ac=
cepting or rejecting the SDP offer, since SDP has its own methods to accept=
 or reject the call as commented before.</div>
</div></blockquote><div><br></div><div style>I meant &#39;accept or reject =
the SDP offer or part of it&#39; =C2=A0where is said &#39;accept or reject =
the call&#39;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"im">
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">A --{&quot;=
action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quot;}-&gt; B<=
br>B --{&quot;action&quot;:&quot;ACCEPT CALL&quot;,&quot;sdp&quot;:&quot;v=
=3D0...&quot;}-&gt; A<br>

</div><div class=3D"gmail_extra"><br></div></div><div class=3D"gmail_extra"=
><div class=3D"im">A --{&quot;action&quot;:&quot;CALL&quot;,&quot;sdp&quot;=
:&quot;v=3D0...&quot;}-&gt; B<br></div>B --{&quot;action&quot;:&quot;ACCEPT=
 CALL&quot;,&quot;sdp&quot;:&quot;v=3D0\r\nm=3Dvideo 0...&quot;}-&gt; A<br>

</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra"><br><=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">2013/7/2 Enr=
ico Marocco <span dir=3D"ltr">&lt;<a href=3D"mailto:enrico.marocco@telecomi=
talia.it" target=3D"_blank">enrico.marocco@telecomitalia.it</a>&gt;</span><=
br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div><div class=3D"h5"><div>On 7/2/13 6:04 PM, Jos=C3=A9 L=
uis Mill=C3=A1n wrote:<br>


&gt; Hi,<br>
&gt;<br>
&gt; Please, let me know how this normal use case could be solved with the<=
br>
&gt; current API.<br>
<br>
</div>Just off the top of my head:<br>
<br>
Audio-only call establishment:<br>
<br>
A --{&quot;action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quo=
t;}-&gt; B<br>
B --{&quot;action&quot;:&quot;ACCEPT CALL&quot;,&quot;sdp&quot;:&quot;v=3D0=
...&quot;}-&gt; A<br>
<br>
Rejected audio+video upgrade proposal:<br>
<br>
A --{&quot;action&quot;:&quot;UPGRADE&quot;,&quot;sdp&quot;:&quot;v=3D0...&=
quot;}-&gt; B<br>
B --{&quot;action&quot;:&quot;REJECT UPGRADE&quot;}-&gt; A<br>
<br>
Alice rolls back and removes local video display.<br>
<br>
I may be missing something, but doesn&#39;t look like rocket science to me.=
..<br>
<span><font color=3D"#888888"><br>
Enrico<br>
<br>
<br>
</font></span><br></div></div>_____________________________________________=
__<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><span class=3D"HOEnZb"><font color=3D"#888888"><br><=
br clear=3D"all"><div><br></div>-- <br>Jos=C3=A9 Luis Mill=C3=A1n
</font></span></div></div>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>Jos=C3=A9 Lu=
is Mill=C3=A1n
</div></div>

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2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it>:
> Audio-only call establishment:
>
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>
> Rejected audio+video upgrade proposal:
>
> A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
> B --{"action":"REJECT UPGRADE"}-> A


SDP itself is a media negotiation protocol so SDP (and its WebRTC
"API") should provide a way to determine whether the remote has
accepted or not our new stream/track. But how? Current API lacks any
kind of event for the case in which the remote replies a SDP rejecting
our new local strem/track.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From jmillan@aliax.net  Tue Jul  2 11:05:28 2013
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Date: Tue, 2 Jul 2013 20:05:22 +0200
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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--20cf307d066a4fc81104e08b3049
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In order to avoid the exposed circumstance, where a user has to 'suppose'
she is sharing her media because there is no way to certainly know, yes,
the JS application should be able to, at least, check if a localMediaStream
is being transmitted out of the RTCPeerConnection.

This would be a simple task if each mediaStream, or mediaStreamTrack had
it's own associated (even if logical because of Boundle) transport. In that
case, such connectivity status would give the information and much more
control over each media element.


2013/7/2 Jim Barnett <Jim.Barnett@genesyslab.com>

> When the JS calls setRemote with the response (which rejects the video),
> the UA will know that the video has been rejected.  Is the idea that the =
UA
> will let the user know via the chrome?  Do we think it is the JS
> applications job to notify the user of success/failure, or the UA's?  I
> don't recall a discussion of this.
>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of I=C3=B1aki Baz Castillo
> Sent: Tuesday, July 02, 2013 12:48 PM
> To: Enrico Marocco
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the
> actual API
>
> 2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it>:
> > Audio-only call establishment:
> >
> > A --{"action":"CALL","sdp":"v=3D0..."}-> B B --{"action":"ACCEPT
> > CALL","sdp":"v=3D0..."}-> A
> >
> > Rejected audio+video upgrade proposal:
> >
> > A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B B --{"action":"REJECT
> > UPGRADE"}-> A
>
>
> SDP itself is a media negotiation protocol so SDP (and its WebRTC
> "API") should provide a way to determine whether the remote has accepted
> or not our new stream/track. But how? Current API lacks any kind of event
> for the case in which the remote replies a SDP rejecting our new local
> strem/track.
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

--20cf307d066a4fc81104e08b3049
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">In order to avoid the exposed circumstance, where a user h=
as to &#39;suppose&#39; she is sharing her media because there is no way to=
 certainly know, yes, the JS application should be able to, at least, check=
 if a localMediaStream is being transmitted out of the RTCPeerConnection.<d=
iv>
<br></div><div style>This would be a simple task if each mediaStream, or me=
diaStreamTrack had it&#39;s own associated (even if logical because of Boun=
dle) transport. In that case, such connectivity status would give the infor=
mation and much more control over each media element.</div>
</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">2013/7/=
2 Jim Barnett <span dir=3D"ltr">&lt;<a href=3D"mailto:Jim.Barnett@genesysla=
b.com" target=3D"_blank">Jim.Barnett@genesyslab.com</a>&gt;</span><br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">
When the JS calls setRemote with the response (which rejects the video), th=
e UA will know that the video has been rejected. =C2=A0Is the idea that the=
 UA will let the user know via the chrome? =C2=A0Do we think it is the JS a=
pplications job to notify the user of success/failure, or the UA&#39;s? =C2=
=A0I don&#39;t recall a discussion of this.<br>

<span class=3D"HOEnZb"><font color=3D"#888888"><br>
- Jim<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a=
> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.or=
g</a>] On Behalf Of I=C3=B1aki Baz Castillo<br>
Sent: Tuesday, July 02, 2013 12:48 PM<br>
To: Enrico Marocco<br>
Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: [rtcweb] Basic scenario &#39;impossible?&#39; to achieve with =
the actual API<br>
<br>
2013/7/2 Enrico Marocco &lt;<a href=3D"mailto:enrico.marocco@telecomitalia.=
it">enrico.marocco@telecomitalia.it</a>&gt;:<br>
&gt; Audio-only call establishment:<br>
&gt;<br>
&gt; A --{&quot;action&quot;:&quot;CALL&quot;,&quot;sdp&quot;:&quot;v=3D0..=
.&quot;}-&gt; B B --{&quot;action&quot;:&quot;ACCEPT<br>
&gt; CALL&quot;,&quot;sdp&quot;:&quot;v=3D0...&quot;}-&gt; A<br>
&gt;<br>
&gt; Rejected audio+video upgrade proposal:<br>
&gt;<br>
&gt; A --{&quot;action&quot;:&quot;UPGRADE&quot;,&quot;sdp&quot;:&quot;v=3D=
0...&quot;}-&gt; B B --{&quot;action&quot;:&quot;REJECT<br>
&gt; UPGRADE&quot;}-&gt; A<br>
<br>
<br>
SDP itself is a media negotiation protocol so SDP (and its WebRTC<br>
&quot;API&quot;) should provide a way to determine whether the remote has a=
ccepted or not our new stream/track. But how? Current API lacks any kind of=
 event for the case in which the remote replies a SDP rejecting our new loc=
al strem/track.<br>

<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
Jos=C3=A9 Luis Mill=C3=A1n
</div>

--20cf307d066a4fc81104e08b3049--

From martin.thomson@gmail.com  Tue Jul  2 11:05:53 2013
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On 2 July 2013 09:13, Enrico Marocco <enrico.marocco@telecomitalia.it> wrot=
e:
> On 7/2/13 6:04 PM, Jos=C3=A9 Luis Mill=C3=A1n wrote:
>> Hi,
>>
>> Please, let me know how this normal use case could be solved with the
>> current API.
>
> Just off the top of my head:
>
> Audio-only call establishment:
>
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>
> Rejected audio+video upgrade proposal:
>
> A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
> B --{"action":"REJECT UPGRADE"}-> A
>
> Alice rolls back and removes local video display.

This is fine for the case that has been proposed, but it doesn't work
as well if A and B are initiating a session.  In that case, you would
need two rounds of negotiation to get the functionality you describe.

There are two separate problems here:

1. How do I reject a proposed stream?

2. How do I learn that my proposed stream was rejected?

I'm sure that there are controls for this already...  Maybe.  You
could, as Enrico suggests, add streams one-by-one using multiple
offer/answer rounds - then you would be able to isolate what was
rejected. That doesn't really scale.

Note: These questions are only relevant with offer/answer.  Choosing
not to do offer/answer means that the application is in control, so
the question is moot.

From martin.thomson@gmail.com  Tue Jul  2 11:06:43 2013
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On 2 July 2013 10:20, Jim Barnett <Jim.Barnett@genesyslab.com> wrote:
> When the JS calls setRemote with the response (which rejects the video), =
the UA will know that the video has been rejected.  Is the idea that the UA=
 will let the user know via the chrome?  Do we think it is the JS applicati=
ons job to notify the user of success/failure, or the UA's?  I don't recall=
 a discussion of this.

Notifications in Chrome aren't sufficient.  The application needs to know t=
oo.

From lorenzo@meetecho.com  Tue Jul  2 11:08:02 2013
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Date: Tue, 2 Jul 2013 20:07:50 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Jim Barnett <Jim.Barnett@genesyslab.com>
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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On Tue, 2 Jul 2013 17:20:04 +0000
Jim Barnett <Jim.Barnett@genesyslab.com> wrote:

> When the JS calls setRemote with the response (which rejects the video), =
the UA will know that the video has been rejected.  Is the idea that the UA=
 will let the user know via the chrome?  Do we think it is the JS applicati=
ons job to notify the user of success/failure, or the UA's?  I don't recall=
 a discussion of this.
>=20
> - Jim
>=20


I think that success/failure in such a case would be a quite complex thing =
to notify, mostly because you'd first have to define/understand what you're=
 trying to change: are you just trying to add video? are you changing the p=
ort? are you muting one direction? what if something was accepted, and some=
thing else not? etc. etc.

Anyway, I agree with you that the UA knows about that. In the case of Alice=
 trying to add video and Bob rejecting it, there may be no callback as of n=
ow, but the JS application should already have a way to know the video has =
been rejected nevertheless. It might just look at the updated MediaStream i=
nstance: if nothing changed, video was rejected (unless in case Bob is only=
 receiving video and not sending it, e.g., because it has no webcam, there =
would be no video track as well?). Not saying it's not ugly, or that a call=
back isn't needed, but it may work.

Lorenzo


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of I=F1aki Baz Castillo
> Sent: Tuesday, July 02, 2013 12:48 PM
> To: Enrico Marocco
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the ac=
tual API
>=20
> 2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it>:
> > Audio-only call establishment:
> >
> > A --{"action":"CALL","sdp":"v=3D0..."}-> B B --{"action":"ACCEPT=20
> > CALL","sdp":"v=3D0..."}-> A
> >
> > Rejected audio+video upgrade proposal:
> >
> > A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B B --{"action":"REJECT=20
> > UPGRADE"}-> A
>=20
>=20
> SDP itself is a media negotiation protocol so SDP (and its WebRTC
> "API") should provide a way to determine whether the remote has accepted =
or not our new stream/track. But how? Current API lacks any kind of event f=
or the case in which the remote replies a SDP rejecting our new local strem=
/track.
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

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Date: Tue, 2 Jul 2013 20:11:27 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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On Tue, 2 Jul 2013 18:13:48 +0200
Enrico Marocco <enrico.marocco@telecomitalia.it> wrote:

> On 7/2/13 6:04 PM, Jos=E9 Luis Mill=E1n wrote:
> > Hi,
> >=20
> > Please, let me know how this normal use case could be solved with the
> > current API.
>=20
> Just off the top of my head:
>=20
> Audio-only call establishment:
>=20
> A --{"action":"CALL","sdp":"v=3D0..."}-> B
> B --{"action":"ACCEPT CALL","sdp":"v=3D0..."}-> A
>=20
> Rejected audio+video upgrade proposal:
>=20
> A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B
> B --{"action":"REJECT UPGRADE"}-> A
>=20
> Alice rolls back and removes local video display.
>=20
> I may be missing something, but doesn't look like rocket science to me...
>=20
> Enrico
>=20
>=20

For the same reasons I just wrote in response to Jim's mail, I'm not sure s=
ome kind of a REJECT-UPGRADE action would be enough to address such a scena=
rio, especially in case, for instance, part of the update attempt was accep=
ted and the rest rejected. Considering several different aspects of a sessi=
on may be subject to change, such a mechanism should be aware of that.

Lorenzo

--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

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--20cf3071c9a4f54d9704e08b8472
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2013/7/2 Martin Thomson <martin.thomson@gmail.com>
>
>
> This is fine for the case that has been proposed, but it doesn't work
> as well if A and B are initiating a session.  In that case, you would
> need two rounds of negotiation to get the functionality you describe.
>
> There are two separate problems here:
>
> 1. How do I reject a proposed stream?
>
> 2. How do I learn that my proposed stream was rejected?
>
> I'm sure that there are controls for this already...  Maybe.


Apart from the statistics API there aren' t, as far as I know. That would
solve the exposed issue.


> You
> could, as Enrico suggests, add streams one-by-one using multiple
> offer/answer rounds - then you would be able to isolate what was
> rejected. That doesn't really scale.


> Note: These questions are only relevant with offer/answer.  Choosing
> not to do offer/answer means that the application is in control, so
> the question is moot.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

--20cf3071c9a4f54d9704e08b8472
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">2013/7/2 Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:mar=
tin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</=
span><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:so=
lid;padding-left:1ex">
<div class=3D"im">
<br>
</div>This is fine for the case that has been proposed, but it doesn&#39;t =
work<br>
as well if A and B are initiating a session. =C2=A0In that case, you would<=
br>
need two rounds of negotiation to get the functionality you describe.<br>
<br>
There are two separate problems here:<br>
<br>
1. How do I reject a proposed stream?<br>
<br>
2. How do I learn that my proposed stream was rejected?<br>
<br>
I&#39;m sure that there are controls for this already... =C2=A0Maybe. =C2=
=A0</blockquote><div><br></div><div style>Apart from the statistics API the=
re aren&#39; t, as far as I know. That would solve the exposed issue.</div>=
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">You<br>
could, as Enrico suggests, add streams one-by-one using multiple<br>
offer/answer rounds - then you would be able to isolate what was<br>
rejected. That doesn&#39;t really scale.</blockquote><blockquote class=3D"g=
mail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-=
left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<br>
Note: These questions are only relevant with offer/answer. =C2=A0Choosing<b=
r>
not to do offer/answer means that the application is in control, so<br>
the question is moot.<br>
<div class=3D""><div class=3D"h5">_________________________________________=
______<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
Jos=C3=A9 Luis Mill=C3=A1n
</div></div>

--20cf3071c9a4f54d9704e08b8472--

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] More H.264 vs VP8 tests
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Encoder rate control is achieved primarily using adaptive quantization. H.2=
64 allows the quantization parameter to adapt at each 16x16 macroblock boun=
dary. VP8 allows the quantization parameter to adapt at each segment (typic=
ally much larger than a macroblock), and only allows 4 segments per frame. =
Theoretically, H.264 allows much finer rate control since it allows much fi=
ner adaptive quantization. For example, 720p video is 3600 16x16 macroblock=
s, so H.264 can adapt 3600 times versus 4 times for VP8. But practically, t=
his is not much of an advantage since quantization is rarely adapted more f=
requently than a few times per frame.

Good (and bad) rate controllers can be achieved using either H.264 or VP8 w=
ith almost equal ease.

Mo

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Stefan H=E5kansson LK
Sent: Tuesday, July 02, 2013 6:16 AM
To: Saverio Mascolo
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] More H.264 vs VP8 tests

On 2013-07-01 19:10, Saverio Mascolo wrote:
> an evaluation of how h264 and vp8 are able to match a changing quality
> would be interesting

Such an evaluation would be very hard to do. The H.264 standard does not=20
specify any encoder behavior, only the decoder is specified. This means=20
that the encoder can be implemented freely, as long as it produces a=20
bitstream compliant to the standard. Specifically, there is no=20
specification in H.264 on how to implement the rate controller. Thus the=20
rate control mechanisms of two different H.264 encoder implementations=20
could react differently to a change in scene complexity.

There is a wealth of H.264 implementations on the market today. Some may=20
have implemented a better rate control mechanism than x264, some may=20
have implemented a worse variant. Testing x264 versus the WebM vp8=20
implemenation with rate control turned on may give some information=20
about the relative merits of these two rate control implementations=20
(although the rate control gives rise to so much measurement noise so=20
even that is hard to do) but it says very little about how the H.264=20
standard compares to the vp8 specification in situations where the scene=20
complexity varies.


>
>
> On Sat, Jun 29, 2013 at 3:17 PM, Stefan H=E5kansson LK
> <stefan.lk.hakansson@ericsson.com
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>
>     On 6/25/13 12:17 PM, Harald Alvestrand wrote:
>      > Again - thanks for releasing this openly!
>      >
>      > I ran the scripts (with a few tweaks; you run on a system where sh=
 is
>      > bash, not dash, for instance), and got the same numbers within
>     +/- 0.5%
>      > (probably some binary version skew);
>
>     I re-run the scripts on another computer with another OS today, and I
>     get exactly the same results as Bo sent out. I noted however that if =
the
>     input clips are not cut at 10s (but used in their entire length) the
>     results get slightly different, but within +/- 0.5%. Can this be the
>     reason why you get slightly different numbers?
>
>
>      > we may have disagreements on the
>      > parameters to use, but we agree on the numbers those parameters
>     produce.
>      >
>      > (I have since modified the Google framework to include a script th=
at
>      > pulls in the sources for the needed binaries and compiles them -
>     if you
>      > want to make 100% sure people are working from the same sources,
>     you may
>      > want to rebase to a newer version of the comparision toolkit.)
>      >
>      > On 06/22/2013 03:41 PM, Bo Burman wrote:
>      >> Hi all,
>      >>
>      >> We have had a look at Google's comparison between VP8 and H.264
>     constrained baseline that was posted on April 3rd
>     (http://www.ietf.org/mail-archive/web/rtcweb/current/msg07028.html).
>     This post contains, as the one mentioned above (and if the
>     attachments make it to the list), information on the exact tools and
>     options used for encoding and should thus be repeatable by anyone
>     interested.
>      >>
>      >> As was already stated by others on this list, one major problem
>     is that Google's test involves the rate control mechanism. Typically
>     codecs are measured with rate control turned off, since it acts as a
>     huge noise on the measurement. Instead we propose to compare the
>     codecs using fixed qp-levels. The qp-level is the quantization
>     parameter that affects the rate/distortion tradeoff. Comparing using
>     fixed qp-levels is what has been used when benchmarking HEVC against
>     H.264 in the JCT-VC standardization, for instance. We are going to
>     select a codec (essentially bit stream format), not a rate control
>     mechanism: Once the codec is selected you can choose whatever rate
>     control mechanism you wish.
>      >>
>      >> We used Google's excellent framework as the baseline and changed
>     the parameter settings in order to make it possible to measure using
>     fixed qp. We used the same sequences, but limited them to the first
>     10 seconds since they varied from 10 seconds to minutes; this also
>     eased computation time.
>      >>
>      >> We used two H.264 encoder implementations: X264, which is an
>     open-source codec that can operate in everything from real-time to
>     slow, and JM which is the reference implementation that was used to
>     develop H.264. JM is very slow but attempts to be very efficient in
>     terms of bits per quality. The results were as follows:
>      >>
>      >> X264 baseline vs VP8: H.264 wins with 1%
>      >> JM baseline vs VP8: H.264 wins with 4%
>      >>
>      >> Running times:
>      >> X264: 1 hour 3 minutes
>      >> VP8: 2 hours 0 minutes
>      >> JM: order of magnitude slower
>      >>
>      >> It is interesting to note that the measurements are more stable
>     in the new test; the variance of the percentages for the sequences
>     is now around 70, down from around 700 in Google's test of April
>     3rd.  We believe this is due to the removal of the rate controller,
>     which acts like noise on the measurements.
>      >>
>      >> We also tried setting H.264 to constrained high (no interlace
>     and no B-pictures, compared to high). The results were then:
>      >>
>      >> X264 constrained high vs VP8: H.264 wins with 25%
>      >> JM constrained high vs VP8: H.264 wins with 24%
>      >>
>      >> We also note that the script that Google provided to calculate
>     the rate differences ("BD-rate") does not give exactly the same
>     numbers as the JCT-VC-way of calculating BD-rate. The main
>     difference is that the JM score for constrained high is better
>     (around 29%) if the JCT-VC way of calculating BD-rate is used.
>      >>
>      >> In summary we think that proper testing can conclude that there
>     is no clear performance advantage to any codec between VP8 and H.264
>     baseline. When comparing VP8 against H.264 constrained high on the
>     other hand, it seems like there is an advantage for H.264
>     constrained high.
>      >>
>      >> The attached file includes the files necessary to reproduce the
>     test.
>      >>
>      >> Best Regards,
>      >>
>      >> Bo Burman
>      >>
>      >>
>      >> _______________________________________________
>      >> rtcweb mailing list
>      >> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>      >> https://www.ietf.org/mailman/listinfo/rtcweb
>      >
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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On 7/2/13 8:05 PM, Martin Thomson wrote:
> There are two separate problems here:
>=20
> 1. How do I reject a proposed stream?
>=20
> 2. How do I learn that my proposed stream was rejected?
>=20
> I'm sure that there are controls for this already...  Maybe.  You
> could, as Enrico suggests, add streams one-by-one using multiple
> offer/answer rounds - then you would be able to isolate what was
> rejected. That doesn't really scale.

Come on, Martin, what Enrico suggested was one of a probably unlimited
number of possible approaches for one very specific supposedly
unaddressable use case. It was put together and sent within 9' since the
initial email hit the list -- don't build your case on it!

FWIW I agree that, in more complex scenarios where streams are
frequently added and/or removed, requiring a single O/A for each
operation is going to be a problem. My preference -- a reasonable
tradeoff at this point in time -- would be to limit SDP O/A to
negotiation of ICE params, crypto material and payload types, and do
stream manipulation in an app-specific way. The latter is no-plan, that
incidentally carries my name on it. The former looks pretty much like
plan A.

Enrico



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On Tue, Jul 2, 2013 at 4:11 PM, Enrico Marocco <
enrico.marocco@telecomitalia.it> wrote:

>
> FWIW I agree that, in more complex scenarios where streams are
> frequently added and/or removed, requiring a single O/A for each
> operation is going to be a problem. My preference -- a reasonable
> tradeoff at this point in time -- would be to limit SDP O/A to
> negotiation of ICE params, crypto material and payload types, and do
> stream manipulation in an app-specific way. The latter is no-plan, that
> incidentally carries my name on it. The former looks pretty much like
> plan A.
>
>
Since ICE needs to be set via API methods in order to support trickle ICE
the only thing that really needs to be in the initial O/A is crypto. Move
this into separate API methods and O/A can be gone.
_____________
Roman Shpount

--089e0112cf9a265e8104e08da2c3
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<div class=3D"gmail_quote">On Tue, Jul 2, 2013 at 4:11 PM, Enrico Marocco <=
span dir=3D"ltr">&lt;<a href=3D"mailto:enrico.marocco@telecomitalia.it" tar=
get=3D"_blank">enrico.marocco@telecomitalia.it</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">
<br>
FWIW I agree that, in more complex scenarios where streams are<br>
frequently added and/or removed, requiring a single O/A for each<br>
operation is going to be a problem. My preference -- a reasonable<br>
tradeoff at this point in time -- would be to limit SDP O/A to<br>
negotiation of ICE params, crypto material and payload types, and do<br>
stream manipulation in an app-specific way. The latter is no-plan, that<br>
incidentally carries my name on it. The former looks pretty much like<br>
plan A.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br></font></span></blockquo=
te><div><br></div><div>Since ICE needs to be set via API methods in order t=
o support trickle ICE the only thing that really needs to be in the initial=
 O/A is crypto. Move this into separate API methods and O/A can be gone.</d=
iv>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--089e0112cf9a265e8104e08da2c3--

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2013/7/2 Lorenzo Miniero <lorenzo@meetecho.com>:
> Anyway, I agree with you that the UA knows about that. In the case of Ali=
ce trying to add video and Bob rejecting it, there may be no callback as of=
 now, but the JS application should already have a way to know the video ha=
s been rejected nevertheless. It might just look at the updated MediaStream=
 instance: if nothing changed, video was rejected

That is *really* ugly, don't you think?


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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You have said: "For example, the only thing that should be needed is a flag=
 that says FEC is available in the signaling for the the RTP layer.=A0"=0A=
=0AI don't really think and this is why I already said the job is MUCH hard=
er than you think.=0A- You also need to say which king of FEC you want to u=
se. Let's say it's ulpfec (RFC 5109)=0AIf we are using ulpfec:=0AA- It is a=
llowed to send FEC packets in the same stream as the protected one or not.=
=0AB- When FEC uses different stream then, it is possible to secure it or n=
ot regardless the protect stream is secure or not. (e.g SRTP/FEC and RTP/Pr=
otected stream). All=A0combinations=A0are possible. When both are encrypted=
 they even could use different keys (if different channels).=0AC- FEC could=
 be bundled with RED or not=0AD- When FEC use different channel then, it co=
uld use different rate than the protected stream if bundled with RED=0AE- S=
ay which media (audio, video, application...) and stream you want to protec=
t=0AF-Whether only one level of protection will be used ("onelevelonly" par=
ameter)=0AG- etc etc...=0A=0ASDP=A0already=A0allow all these use-cases. Dep=
ending on the =A0implementations,=A0signalling=A0[A-F] could be mandatory.=
=0A=0AGood luck...=0A=0A=0A________________________________=0A De=A0: Robin=
 Raymond <robin@hookflash.com>=0A=C0=A0: Tim Panton <tim@phonefromhere.com>=
 =0ACc=A0: "rtcweb@ietf.org" <rtcweb@ietf.org> =0AEnvoy=E9 le : Mardi 2 jui=
llet 2013 14h22=0AObjet=A0: Re: [rtcweb] New Draft - WebRTC JS Object API M=
odel=0A =0A=0A=0A=0AYou are absolutely, there are=A0 a lot of details. But =
splitting the =0Adetails into logical separate components reduces the overa=
ll complexity =0Agreatly. Plus many of those details are only needed becaus=
e of a lack =0Aability to properly describe the concept of "streams" at the=
 lower level=0A RTP layer (which really only supports tracks at best, and l=
acks =0Asufficient details to coordinate any information between them to =
=0Areassembly remotely as a "stream"). So we are left in a situation where =
=0Athe high layers must coordinate a lot of information to be capable of =
=0Areassembling the streams when they really should be just negotiating the=
=0A capabilities of the RTP layer itself.=0A=0AFor example, the only thing =
that should be needed is a flag that says =0AFEC is available in the signal=
ing for the the RTP layer. If it is set =0Athen that would imply the engine=
 supports this feature and thus FEC =0Astreams may be created by the sendin=
g party. Alas, because RTP lacks =0Asufficient depth for such concepts like=
 FEC, you have to signal it. RTP =0Adoesn't even have enough information to=
 reassemble tracks into stream, =0Alet allow do things like FEC between the=
m.=0A=0AHaving said that, by making a scoped object that understands there =
is =0Asome definition of what a stream means needs to be describes, but =0A=
perhaps that the "future' may not require such crazy definitions like =0Aal=
l the stream/ssrc/track mappings, it allows for a short/long-term =0Atempor=
ary situation where stream definitions are passed until we are =0Aable move=
 to a model where that is no longer needed. When a theoretical =0Anext-gen =
RTP transport is capable of sending entire streams and =0Areassembling them=
 without explicit signaling from an upper layer, we can=0A stop defining al=
l that mapping stuff in signaling and remove that =0Acomplexity.=0A=0ABasic=
ally, it's about scoping the objects, and limiting the requirements=0A need=
ed to exchange per object, and defining exactly what is =0Aexpected/require=
d in those definitions and allowing for future =0Aalternative objects that =
could replace current ones (if they should =0Abecome available). The other =
important aspect is removing the definition=0A how this information must be=
 coordinated and exchanged to allow for a =0Avariety of signaling scenarios=
 and alternative state machines. And =0Akeeping it relatively simple is imp=
ortant for the simple use cases too =0Aso the burden is low. We are well on=
 our way to producing such a draft =0Ain short order that does exactly that=
.=0A=0AAll we ask is that people keep an open mind. We have no desire to =
=0Aprevent SDP or offer/answer, just not mandate its usage as an API model.=
=0A=0AMake sense?=0A=0A-Robin=0A=0A=0ATim Panton=0A>2 July, 2013 4:31 AM=0A=
>=0A>=0A>On 2 Jul 2013, at 08:21, Stefan H=E5kansson LK wrote:=0A>=0A>Conve=
rsely we =0Ashouldn't underestimate (again) the difficulty of describing al=
l the=0A>nitty gritty details in SDP.=A0=0A>=0A>=0A>I have the sense that =
=0Awe may have pushed the flexibility of SDP beyond it's elastic limit.=0A>=
=0A>=0A>I think that over the last year we have discovered that a dynamical=
ly =0Aprogrammable browser=0A>is a fundamentally different beast from a des=
kphone and that what suits one may well not suit the other.=0A>=0A>=0A>Tim.=
=0A>=0A>=0A>_______________________________________________=0A>rtcweb=0A ma=
iling list=0A>rtcweb@ietf.org=0A>https://www.ietf.org/mailman/listinfo/rtcw=
eb=0A>=0A>=0A_______________________________________________=0Artcweb maili=
ng list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/mailman/listinfo/rtcweb
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"font-fa=
mily: 'times new roman', 'new york', times, serif; font-size: 12pt;"><span>=
You have said: "</span><span style=3D"color: rgb(69, 69, 69); font-family: =
'Helvetica Neue', Helvetica, Arial, sans-serif; font-size: 12px;">For examp=
le, the only thing that should be needed is a flag that says FEC is availab=
le in the signaling for the the RTP layer.</span><span style=3D"color: rgb(=
69, 69, 69); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; f=
ont-size: 12px;">&nbsp;</span><span style=3D"font-size: 12pt;">"</span></di=
v><div style=3D"font-family: 'times new roman', 'new york', times, serif; f=
ont-size: 12pt; color: rgb(0, 0, 0); background-color: transparent; font-st=
yle: normal;"><span style=3D"font-size: 12pt;"><br></span></div><div style=
=3D"font-family: 'times new roman', 'new york', times, serif; font-size: 12=
pt; color:
 rgb(0, 0, 0); background-color: transparent; font-style: normal;"><span st=
yle=3D"font-size: 12pt;"><span style=3D"font-weight: bold;">I don't really =
think</span> and this is why I already said the job is <span style=3D"font-=
weight: bold;">MUCH</span> harder than you think.</span></div><div style=3D=
"font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;=
 color: rgb(0, 0, 0); background-color: transparent; font-style: normal;"><=
span style=3D"font-size: 12pt;">- You also need to say which king of FEC yo=
u want to use. Let's say it's ulpfec (RFC 5109)</span></div><div style=3D"f=
ont-family: 'times new roman', 'new york', times, serif; font-size: 12pt; c=
olor: rgb(0, 0, 0); background-color: transparent; font-style: normal;">If =
we are using ulpfec:</div><div style=3D"font-family: 'times new roman', 'ne=
w york', times, serif; font-size: 12pt; color: rgb(0, 0, 0); background-col=
or: transparent; font-style: normal;">A- It is allowed to send FEC packets =
in the
 same stream as the protected one or not.</div><div style=3D"background-col=
or: transparent;"><font size=3D"3">B- When FEC uses different stream then, =
it is possible to secure it or not regardless the protect stream is secure =
or not. (e.g SRTP/FEC and RTP/Protected stream). All&nbsp;</font>combinatio=
ns<font size=3D"3">&nbsp;are possible. When both are encrypted they even co=
uld use different keys (if different channels).</font></div><div style=3D"f=
ont-family: 'times new roman', 'new york', times, serif; font-size: 12pt; c=
olor: rgb(0, 0, 0); background-color: transparent; font-style: normal;">C- =
FEC could be bundled with RED or not</div><div style=3D"font-family: 'times=
 new roman', 'new york', times, serif; font-size: 12pt; color: rgb(0, 0, 0)=
; background-color: transparent; font-style: normal;">D- When FEC use diffe=
rent channel then, it could use different rate than the protected stream if=
 bundled with RED</div><div style=3D"font-family: 'times new roman', 'new y=
ork',
 times, serif; font-size: 12pt; color: rgb(0, 0, 0); background-color: tran=
sparent; font-style: normal;">E- Say which media (audio, video, application=
...) and stream you want to protect</div><div style=3D"font-family: 'times =
new roman', 'new york', times, serif; font-size: 12pt; color: rgb(0, 0, 0);=
 background-color: transparent; font-style: normal;">F-Whether only one lev=
el of protection will be used ("<span style=3D"font-size: 1em;">onelevelonl=
y" parameter</span><span style=3D"background-color: transparent; font-size:=
 12pt;">)</span></div><div style=3D"font-family: 'times new roman', 'new yo=
rk', times, serif; font-size: 12pt; color: rgb(0, 0, 0); background-color: =
transparent; font-style: normal;"><span style=3D"background-color: transpar=
ent; font-size: 12pt;">G- etc etc...</span></div><div style=3D"font-family:=
 'times new roman', 'new york', times, serif; font-size: 12pt; color: rgb(0=
, 0, 0); background-color: transparent; font-style: normal;"><span
 style=3D"background-color: transparent; font-size: 12pt;"><br></span></div=
><div style=3D"background-color: transparent;"><span style=3D"background-co=
lor: transparent;"><font size=3D"3">SDP&nbsp;</font>already<font size=3D"3"=
>&nbsp;allow all these use-cases. Depending on the &nbsp;implementations,&n=
bsp;signalling&nbsp;[A-F] could be mandatory.</font></span></div><div style=
=3D"background-color: transparent; color: rgb(0, 0, 0); font-size: 16px; fo=
nt-family: 'times new roman', 'new york', times, serif; font-style: normal;=
"><span style=3D"background-color: transparent;"><font size=3D"3"><br></fon=
t></span></div><div style=3D"background-color: transparent; color: rgb(0, 0=
, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, s=
erif; font-style: normal;"><span style=3D"background-color: transparent;"><=
font size=3D"3">Good luck...</font></span></div><div style=3D"font-family: =
'times new roman', 'new york', times, serif; font-size: 12pt;"><br></div>  =
<div
 style=3D"font-family: 'times new roman', 'new york', times, serif; font-si=
ze: 12pt;"> <div style=3D"font-family: 'times new roman', 'new york', times=
, serif; font-size: 12pt;"> <div dir=3D"ltr"> <hr size=3D"1">  <font size=
=3D"2" face=3D"Arial"> <b><span style=3D"font-weight:bold;">De&nbsp;:</span=
></b> Robin Raymond &lt;robin@hookflash.com&gt;<br> <b><span style=3D"font-=
weight: bold;">=C0&nbsp;:</span></b> Tim Panton &lt;tim@phonefromhere.com&g=
t; <br><b><span style=3D"font-weight: bold;">Cc&nbsp;:</span></b> "rtcweb@i=
etf.org" &lt;rtcweb@ietf.org&gt; <br> <b><span style=3D"font-weight: bold;"=
>Envoy=E9 le :</span></b> Mardi 2 juillet 2013 14h22<br> <b><span style=3D"=
font-weight: bold;">Objet&nbsp;:</span></b> Re: [rtcweb] New Draft - WebRTC=
 JS Object API Model<br> </font> </div> <div class=3D"y_msg_container"><br>=
<div id=3D"yiv6733358432">=0A =0A<div><br>=0AYou are absolutely, there are&=
nbsp; a lot of details. But splitting the =0Adetails into logical separate =
components reduces the overall complexity =0Agreatly. Plus many of those de=
tails are only needed because of a lack =0Aability to properly describe the=
 concept of "streams" at the lower level=0A RTP layer (which really only su=
pports tracks at best, and lacks =0Asufficient details to coordinate any in=
formation between them to =0Areassembly remotely as a "stream"). So we are =
left in a situation where =0Athe high layers must coordinate a lot of infor=
mation to be capable of =0Areassembling the streams when they really should=
 be just negotiating the=0A capabilities of the RTP layer itself.<br>=0A<br=
>=0AFor example, the only thing that should be needed is a flag that says =
=0AFEC is available in the signaling for the the RTP layer. If it is set =
=0Athen that would imply the engine supports this feature and thus FEC =0As=
treams may be created by the sending party. Alas, because RTP lacks =0Asuff=
icient depth for such concepts like FEC, you have to signal it. RTP =0Adoes=
n't even have enough information to reassemble tracks into stream, =0Alet a=
llow do things like FEC between them.<br>=0A<br>=0AHaving said that, by mak=
ing a scoped object that understands there is =0Asome definition of what a =
stream means needs to be describes, but =0Aperhaps that the "future' may no=
t require such crazy definitions like =0Aall the stream/ssrc/track mappings=
, it allows for a short/long-term =0Atemporary situation where stream defin=
itions are passed until we are =0Aable move to a model where that is no lon=
ger needed. When a theoretical =0Anext-gen RTP transport is capable of send=
ing entire streams and =0Areassembling them without explicit signaling from=
 an upper layer, we can=0A stop defining all that mapping stuff in signalin=
g and remove that =0Acomplexity.<br>=0A<br>=0ABasically, it's about scoping=
 the objects, and limiting the requirements=0A needed to exchange per objec=
t, and defining exactly what is =0Aexpected/required in those definitions a=
nd allowing for future =0Aalternative objects that could replace current on=
es (if they should =0Abecome available). The other important aspect is remo=
ving the definition=0A how this information must be coordinated and exchang=
ed to allow for a =0Avariety of signaling scenarios and alternative state m=
achines. And =0Akeeping it relatively simple is important for the simple us=
e cases too =0Aso the burden is low. We are well on our way to producing su=
ch a draft =0Ain short order that does exactly that.<br>=0A<br>=0AAll we as=
k is that people keep an open mind. We have no desire to =0Aprevent SDP or =
offer/answer, just not mandate its usage as an API model.<br>=0A<br>=0AMake=
 sense?<br>=0A<br>=0A-Robin<br>=0A<br>=0A<blockquote style=3D"border:0px no=
ne;" type=3D"cite">=0A  <div style=3D"margin:30px 25px 10px 25px;" class=3D=
"yiv6733358432__pbConvHr"><div style=3D"display:table;width:100%;border-top=
:1px solid #EDEEF0;padding-top:5px;"> =09<div style=3D"display:table-cell;v=
ertical-align:middle;padding-right:6px;"><img src=3D"cid:1.3919835227@web17=
1301.mail.ir2.yahoo.com" name=3D"postbox-contact.jpg" height=3D"25px" width=
=3D"25px"></div>   <div style=3D"display:table-cell;white-space:nowrap;vert=
ical-align:middle;width:100%;">=0A   =09<a rel=3D"nofollow" ymailto=3D"mail=
to:tim@phonefromhere.com" target=3D"_blank" href=3D"mailto:tim@phonefromher=
e.com" style=3D"color:#737F92;padding-right:6px;font-weight:bold;text-decor=
ation:none;">Tim Panton</a></div>   <div style=3D"display:table-cell;white-=
space:nowrap;vertical-align:middle;">   =0A  <font color=3D"#9FA2A5"><span =
style=3D"padding-left:6px;">2 July, 2013 4:31=0A AM</span></font></div></di=
v></div>=0A  <div style=3D"color:#888888;margin-left:24px;margin-right:24px=
;" class=3D"yiv6733358432__pbConvBody"><br><div><div>On 2 Jul 2013, at=0A 0=
8:21, Stefan H=E5kansson LK wrote:</div></div><br><div>Conversely we =0Asho=
uldn't underestimate (again) the difficulty of describing all the</div><div=
>nitty=0A gritty details in SDP.&nbsp;</div><div><br></div><div>I have the =
sense that =0Awe may have pushed the flexibility of SDP beyond it's elastic=
 limit.</div><div><br></div><div>I=0A think that over the last year we have=
 discovered that a dynamically =0Aprogrammable browser</div><div>is a funda=
mentally different beast from a=0A deskphone and that what suits one may we=
ll not suit the other.</div><div><br></div><div>Tim.</div><div><br></div><d=
iv>_______________________________________________<br>rtcweb=0A mailing lis=
t<br><a rel=3D"nofollow" class=3D"yiv6733358432moz-txt-link-abbreviated" ym=
ailto=3D"mailto:rtcweb@ietf.org" target=3D"_blank" href=3D"mailto:rtcweb@ie=
tf.org">rtcweb@ietf.org</a><br><a rel=3D"nofollow" class=3D"yiv6733358432mo=
z-txt-link-freetext" target=3D"_blank" href=3D"https://www.ietf.org/mailman=
/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></div=
></div>=0A  <br>=0A</blockquote>=0A</div>=0A</div><br>_____________________=
__________________________<br>rtcweb mailing list<br><a ymailto=3D"mailto:r=
tcweb@ietf.org" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a h=
ref=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">http=
s://www.ietf.org/mailman/listinfo/rtcweb</a><br><br><br></div> </div> </div=
>  </div></body></html>
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: =?iso-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
Thread-Topic: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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Date: Wed, 3 Jul 2013 07:30:00 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the	actual API
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On 7/2/13 8:05 PM, Jos=E9 Luis Mill=E1n wrote:=0A=
> In order to avoid the exposed circumstance, where a user has to=0A=
> 'suppose' she is sharing her media because there is no way to certainly=
=0A=
> know, yes, the JS application should be able to, at least, check if a=0A=
> localMediaStream is being transmitted out of the RTCPeerConnection.=0A=
=0A=
There is a proposal that addresses this:=0A=
=0A=
http://lists.w3.org/Archives/Public/public-webrtc/2013Jan/att-0005/PrioAPI.=
pdf=0A=
=0A=
(Note that the constraint part would be simplified per the recent =0A=
agreements in the Media Capture TF)=0A=
=0A=
Stefan=0A=
=0A=
=0A=
=0A=
>=0A=
> This would be a simple task if each mediaStream, or mediaStreamTrack had=
=0A=
> it's own associated (even if logical because of Boundle) transport. In=0A=
> that case, such connectivity status would give the information and much=
=0A=
> more control over each media element.=0A=
>=0A=
>=0A=
> 2013/7/2 Jim Barnett <Jim.Barnett@genesyslab.com=0A=
> <mailto:Jim.Barnett@genesyslab.com>>=0A=
>=0A=
>     When the JS calls setRemote with the response (which rejects the=0A=
>     video), the UA will know that the video has been rejected.  Is the=0A=
>     idea that the UA will let the user know via the chrome?  Do we think=
=0A=
>     it is the JS applications job to notify the user of success/failure,=
=0A=
>     or the UA's?  I don't recall a discussion of this.=0A=
>=0A=
>     - Jim=0A=
>=0A=
>     -----Original Message-----=0A=
>     From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>=0A=
>     [mailto:rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>] On=
=0A=
>     Behalf Of I=F1aki Baz Castillo=0A=
>     Sent: Tuesday, July 02, 2013 12:48 PM=0A=
>     To: Enrico Marocco=0A=
>     Cc: rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with=0A=
>     the actual API=0A=
>=0A=
>     2013/7/2 Enrico Marocco <enrico.marocco@telecomitalia.it=0A=
>     <mailto:enrico.marocco@telecomitalia.it>>:=0A=
>      > Audio-only call establishment:=0A=
>      >=0A=
>      > A --{"action":"CALL","sdp":"v=3D0..."}-> B B --{"action":"ACCEPT=
=0A=
>      > CALL","sdp":"v=3D0..."}-> A=0A=
>      >=0A=
>      > Rejected audio+video upgrade proposal:=0A=
>      >=0A=
>      > A --{"action":"UPGRADE","sdp":"v=3D0..."}-> B B --{"action":"REJEC=
T=0A=
>      > UPGRADE"}-> A=0A=
>=0A=
>=0A=
>     SDP itself is a media negotiation protocol so SDP (and its WebRTC=0A=
>     "API") should provide a way to determine whether the remote has=0A=
>     accepted or not our new stream/track. But how? Current API lacks any=
=0A=
>     kind of event for the case in which the remote replies a SDP=0A=
>     rejecting our new local strem/track.=0A=
>=0A=
>     --=0A=
>     I=F1aki Baz Castillo=0A=
>     <ibc@aliax.net <mailto:ibc@aliax.net>>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
>=0A=
>=0A=
> --=0A=
> Jos=E9 Luis Mill=E1n=0A=
=0A=

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To: =?iso-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
Thread-Topic: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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On 7/2/13 6:04 PM, Jos=E9 Luis Mill=E1n wrote:=0A=
> Hi,=0A=
>=0A=
> Please, let me know how this normal use case could be solved with the=0A=
> current API.=0A=
>=0A=
> Alice and Bob have accessed to a Web page offering a basic WebRTC=0A=
> application   that they use to speak and see each other. They are in the=
=0A=
> middle of a media session right now.=0A=
>=0A=
> Alice presses the "Video" button in her web application. A SDP offer=0A=
> with video description is sent to Bob:=0A=
>=0A=
> Bob doesn't want to see her, so his web application rejects the media by=
=0A=
> using any valid method for that:=0A=
> -- a=3Dinactive=0A=
> -- m=3Dvideo 0=0A=
=0A=
My personal view is that even if we use SDP for carrying the association =
=0A=
between RTP SSRCs and the MediaStream and MediaStreamTrack structures as =
=0A=
well as negotiating codecs, we should not use it to negotiate what =0A=
streams and tracks to use. It should only be used to set up the streams =0A=
and tracks over the networks that is decided by the application.=0A=
=0A=
One way in this specific case could be that Bob's application (since Bob =
=0A=
does not want to see Alice) simply does not show the video. Bob's =0A=
application could in addition signal this to Alice's application (in =0A=
some app proprietary way) and Alice's app could remove the video stream =0A=
(or track) from PeerConnection to save bandwidth and processing.=0A=
=0A=
Stefan=0A=
=0A=
>=0A=
> Alice web application received a SDP answer which was consumed=0A=
> successfully by the RTCPeerConnection. Alice thinks she is sharing her=0A=
> video, but she is not.=0A=
>=0A=
> The application "Video" button is disabled since the video is already=0A=
> being shared (theoretically)=0A=
>=0A=
> The application "self-Video" HTML5 video element displays Alice's face.=
=0A=
> She thinks that Bob is seeing her face too, but he did indeed rejected so=
.=0A=
>=0A=
> I hope this common scenario can be achieved somehow with the current=0A=
> API. I can't find how to do it though.=0A=
>=0A=
> Thanks in advance=0A=
>=0A=
> --=0A=
> Jos=E9 Luis Mill=E1n=0A=
=0A=

From enrico.marocco@telecomitalia.it  Wed Jul  3 01:23:40 2013
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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--------------ms090906030002040805000804
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On 7/2/13 6:30 PM, Jos=C3=A9 Luis Mill=C3=A1n wrote:
> Thanks Enrico,
>=20
> Yes, you are right. I was missing a detail.
>=20
> Now imagine you are not using your own wire protocol but a standard one=
,
> where accepting the request doesn't imply accepting or rejecting the SD=
P
> offer, since SDP has its own methods to accept or reject the call as
> commented before.

I assume you are talking about browser-to-browser signalling. If it was
a deliberate choice to pick SIP rather than a custom protocol properly
designed for the semantics of you application, then it is a
self-inflicted pain you cannot blame on the API.

If on the other hand you are talking about interworking with SIP
networks, I'm sympathetic with you, as there is no way for your
gatewaying function (regardless of whether it is on an external element
or wired directly into the JS) to avoid SDP parsing and handling of the
42 different ways it defines for signalling stream rejection. My advice
in such case: hang on in there! Eventually your customers will be happy
to pay you good money for the commendable effort.

Enrico



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From ibc@aliax.net  Wed Jul  3 01:59:18 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 3 Jul 2013 10:58:52 +0200
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2013/7/3 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> One way in this specific case could be that Bob's application (since Bob
> does not want to see Alice) simply does not show the video. Bob's
> application could in addition signal this to Alice's application (in
> some app proprietary way) and Alice's app could remove the video stream
> (or track) from PeerConnection to save bandwidth and processing.

Let me understand it:

So we are forced to use SDP in any WebRTC application but we cannot
use SDP for common features (i.e. detecting that the remote peer has
rejected our new offered stream/track). Thus regardless we use SDP we
cannot do SIP, right? and instead we must use a proprietary signaling
mechanism for those so common features already present in SIP and SDP,
right?

So live with SDP but don't use SDP, am I wrong?

So be free to implement any custom signaling protocol (proprietary
protocol) but not a standard like SIP since SIP's SDP usage cannot be
implemented in WebRTC, right?

Opss, I forgot, I can parse my new generated SDP at JS level (the SDP
with my new video stream/track offer) and then also parse the received
SDP, and detect wheter there is "a=3Dinactive" or "m" line with port 0
to detect that the remote has rejected my stream. Am I really right?

Regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2013/7/2 Lorenzo Miniero <lorenzo@meetecho.com>:
> In the case of Alice trying to add video and Bob rejecting it, there may =
be no callback as of now, but the JS application should already have a way =
to know the video has been rejected nevertheless. It might just look at the=
 updated MediaStream instance: if nothing changed, video was rejected (unle=
ss in case Bob is only receiving video and not sending it, e.g., because it=
 has no webcam, there would be no video track as well?).

One more comment about this:

The fact that Alice offers video to Bob does not mean that Bob also
should offer video to Alice. It could perfectly occur that Alice
offers video to Bob, Bob accepts receiving it but Bob does not send
video to Alice (i.e. "a=3Drecvonly").

The problem described in this issue is the fact that Alice has no way
to realize whether Bob has accepted receiving her video or not. Bob
has replied to the video offer with "m=3Dvideo 0" or "a=3Dinactive", which
means that there won't be video RTP from Alice to Bob. But there is no
way for Alice's JS to realize of that so Alice's will expect that Bob
is receiving her video (when he is not). There is no API callback to
know whether the remote has accepted our flows or not.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From lorenzo@meetecho.com  Wed Jul  3 05:32:10 2013
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Subject: Re: [rtcweb] Basic scenario 'impossible?' to achieve with the actual API
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Il giorno Wed, 3 Jul 2013 13:58:54 +0200
I=C3=B1aki Baz Castillo <ibc@aliax.net> ha scritto:

> 2013/7/2 Lorenzo Miniero <lorenzo@meetecho.com>:
> > In the case of Alice trying to add video and Bob rejecting it,
> > there may be no callback as of now, but the JS application should
> > already have a way to know the video has been rejected
> > nevertheless. It might just look at the updated MediaStream
> > instance: if nothing changed, video was rejected (unless in case
> > Bob is only receiving video and not sending it, e.g., because it
> > has no webcam, there would be no video track as well?).
>=20
> One more comment about this:
>=20
> The fact that Alice offers video to Bob does not mean that Bob also
> should offer video to Alice. It could perfectly occur that Alice
> offers video to Bob, Bob accepts receiving it but Bob does not send
> video to Alice (i.e. "a=3Drecvonly").
>=20


Yes, that's what I meant by the "unless...": I'm not sure a remote
MediaStream would look different in case of no video at all or a
recvonly video. If not, I agree that not even the ugly approach I
mentioned would work.

L.


> The problem described in this issue is the fact that Alice has no way
> to realize whether Bob has accepted receiving her video or not. Bob
> has replied to the video offer with "m=3Dvideo 0" or "a=3Dinactive", which
> means that there won't be video RTP from Alice to Bob. But there is no
> way for Alice's JS to realize of that so Alice's will expect that Bob
> is receiving her video (when he is not). There is no API callback to
> know whether the remote has accepted our flows or not.
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Robin Raymond <robin@hookflash.com>
Thread-Topic: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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Subject: Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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On Jun 29, 2013, at 8:28 PM, Robin Raymond <robin@hookflash.com> wrote:

> Versus this SDP blob that many of us must parse/mangle/generate from our =
code if we want to go beyond a basic demo:

My assertion has always been we should avoid any JS app needing to touch th=
e SDP. Every example I have heard of why people are doing (with exception o=
f one) it is ones that there has been discussion of providing an API to do =
that and/or to fix a bug in browsers. (Just as FYI, the one exception is Ka=
ufrman's use of ICE to put things in what is usual done as a MPLS tunnel. T=
here are probably other common ways to solve that problem).=20

I like Peter's idea of trying to capture what are the reason people think t=
hey need to look at the SDP.=20





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From: Peter Thatcher <pthatcher@google.com>
Date: Wed, 3 Jul 2013 12:34:02 -0700
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Subject: Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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--047d7b6d8d7cb7ac6a04e0a08d4d
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We already have some in the spreadsheet that I previously linked (
https://docs.google.com/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhy=
bVl4anJLRkVPV0E#gid=3D0
)

Wolfgang Beck: Muting a call
Lance Stout: Jingle signalling
Gustavo Garc=C3=ADa: Set "inactive", control crypto, modifying codecs, modi=
fying
bitrate,
Philipp Hancke: Jingle signalling
Alexandre Gouaillard: Knowing the codec list, the tranport details, the
bandwidth options, which ice candidates are used or failing
Roman Shpount: Interop between browser SDP and MCU SDP
Silvia : Not trying to mangle SDP yet, but would like to control bandwidth
requests/limitations; bitrate requests; codec settings
I=C3=B1aki Baz Castillo: Mangling it, but didn't say for what purpose.


And I could give a very long list of things, but I'll continue to keep
myself disqualified :).




On Wed, Jul 3, 2013 at 12:16 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Jun 29, 2013, at 8:28 PM, Robin Raymond <robin@hookflash.com> wrote:
>
> > Versus this SDP blob that many of us must parse/mangle/generate from ou=
r
> code if we want to go beyond a basic demo:
>
> My assertion has always been we should avoid any JS app needing to touch
> the SDP. Every example I have heard of why people are doing (with excepti=
on
> of one) it is ones that there has been discussion of providing an API to =
do
> that and/or to fix a bug in browsers. (Just as FYI, the one exception is
> Kaufrman's use of ICE to put things in what is usual done as a MPLS tunne=
l.
> There are probably other common ways to solve that problem).
>
> I like Peter's idea of trying to capture what are the reason people think
> they need to look at the SDP.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b6d8d7cb7ac6a04e0a08d4d
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<div dir=3D"ltr">We already have some in the spreadsheet that I previously =
linked (<a href=3D"https://docs.google.com/spreadsheet/ccc?key=3D0AuaKXw3Sk=
HMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D0">https://docs.google.com/spreads=
heet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D0</a>)<di=
v>

<br></div><div>Wolfgang Beck: Muting a call<br></div><div><span style=3D"co=
lor:rgb(0,0,0);font-family:arial,sans,sans-serif;font-size:13px;white-space=
:pre-wrap">Lance Stout: Jingle signalling</span><br></div><div><span style=
=3D"color:rgb(0,0,0);font-family:arial,sans,sans-serif;font-size:13px;white=
-space:pre-wrap">Gustavo Garc=C3=ADa: Set &quot;inactive&quot;, control cry=
pto, modifying codecs, modifying bitrate, </span><br>

</div><div><span style=3D"color:rgb(0,0,0);font-family:arial,sans,sans-seri=
f;font-size:13px;white-space:pre-wrap">Philipp Hancke: Jingle signalling</s=
pan><span style=3D"color:rgb(0,0,0);font-family:arial,sans,sans-serif;font-=
size:13px;white-space:pre-wrap"><br>

</span></div><div><span style=3D"color:rgb(0,0,0);font-family:arial,sans,sa=
ns-serif;font-size:13px;white-space:pre-wrap">Alexandre Gouaillard: Knowing=
 the codec list, the tranport details, the bandwidth options, which ice can=
didates are used or failing</span><span style=3D"color:rgb(0,0,0);font-fami=
ly:arial,sans,sans-serif;font-size:13px;white-space:pre-wrap"><br>

</span></div><div><span style=3D"color:rgb(0,0,0);font-family:arial,sans,sa=
ns-serif;font-size:13px;white-space:pre-wrap">Roman Shpount: Interop betwee=
n browser SDP and MCU SDP</span><span style=3D"color:rgb(0,0,0);font-family=
:arial,sans,sans-serif;font-size:13px;white-space:pre-wrap"><br>

</span></div><div><span style=3D"color:rgb(0,0,0);font-family:arial,sans,sa=
ns-serif;font-size:13px;white-space:pre-wrap">Silvia : Not trying to mangle=
 SDP yet, but would like to control </span><span style=3D"color:rgb(0,0,0);=
font-family:arial,sans,sans-serif;font-size:13px;white-space:pre-wrap">band=
width requests/limitations; bitrate requests; codec settings</span></div>

<div><font color=3D"#000000" face=3D"arial, sans, sans-serif"><span style=
=3D"white-space:pre-wrap">I=C3=B1aki Baz Castillo: Mangling it, but didn&#3=
9;t say for what purpose.</span></font></div><div><font color=3D"#000000" f=
ace=3D"arial, sans, sans-serif"><span style=3D"white-space:pre-wrap"><br>

</span></font></div><div><div><font color=3D"#000000" face=3D"arial, sans, =
sans-serif"><span style=3D"white-space:pre-wrap"><br class=3D"">And I could=
 give a very long list of things, but I&#39;ll continue to keep myself disq=
ualified :).</span></font></div>

</div><div><font color=3D"#000000" face=3D"arial, sans, sans-serif"><span s=
tyle=3D"white-space:pre-wrap"><br></span></font></div><div><font color=3D"#=
000000" face=3D"arial, sans, sans-serif"><span style=3D"white-space:pre-wra=
p"><br>
</span></font></div>
<div><font color=3D"#000000" face=3D"arial, sans, sans-serif"><span style=
=3D"white-space:pre-wrap"> </span></font></div></div><div class=3D"gmail_ex=
tra"><br><br><div class=3D"gmail_quote">On Wed, Jul 3, 2013 at 12:16 PM, Cu=
llen Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco=
.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Jun 29, 2013, at 8:28 PM, Robin Raymond &lt;<a href=3D"mailto:robin@hook=
flash.com">robin@hookflash.com</a>&gt; wrote:<br>
<br>
&gt; Versus this SDP blob that many of us must parse/mangle/generate from o=
ur code if we want to go beyond a basic demo:<br>
<br>
</div>My assertion has always been we should avoid any JS app needing to to=
uch the SDP. Every example I have heard of why people are doing (with excep=
tion of one) it is ones that there has been discussion of providing an API =
to do that and/or to fix a bug in browsers. (Just as FYI, the one exception=
 is Kaufrman&#39;s use of ICE to put things in what is usual done as a MPLS=
 tunnel. There are probably other common ways to solve that problem).<br>


<br>
I like Peter&#39;s idea of trying to capture what are the reason people thi=
nk they need to look at the SDP.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7b6d8d7cb7ac6a04e0a08d4d--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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Date: Wed, 3 Jul 2013 19:48:58 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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On Jul 3, 2013, at 12:34 PM, Peter Thatcher <pthatcher@google.com> wrote:

> And I could give a very long list of things, but I'll continue to keep my=
self disqualified :).
>=20
>=20

Please bring up the long list now - later just delays the discussion =85


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Subject: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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--089e0160cb9a625ba004e0a1d8b2
Content-Type: text/plain; charset=UTF-8

After about a week of input, and feedback from 19 developers doing a wide
range of things using the WebRTC API, we have some data about their opinion
of the current API.

*Short answer*:

61% "Good for simple Stuff"    49% "Bad for simple stuff"
 6% "Good for advanced stuff"      94% "Bad for advanced stuff"
46% "Good/OK/tolerable for 1.0"    54% "Horrible/intolerable for 1.0"
 7% "Good/OK/tolerable long-term"  93% "Need a better API long-term"


In general the attitude is *"It's OK for simple stuff but bad for advanced
stuff",  *and *"We can tolerate it for 1.0, but we need something better
for 2.0".  *But the majority of the feedback was against it even for the
1.0, so the last one is a bit of a stretch.


*Long Answer*:

I had to interpret the results a little to fit into these buckets, so if
you don't like the interpretation, you're more than welcome to interpret
your own results from the raw data:
https://docs.google.com/spreadsheet/ccc?key=0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=1


There seem to be 4 or 5 different camps:

Camp #1:  Doing very simple things, doesn't touch the SDP.  Everything is
fine, except they might have things they want to control but can't.

Camp #2:  Does somewhat more advanced things, touches the SDP a little, has
pain, anticipates more pain, would like something better, but thinks this
is OK for 1.0.

Camp #3:  Is trying advanced things, is touching the SDP a lot, and really
feels pain.  Wants something better, and might tolerate this API for 1.0,
but not for long.

Camp #4:  Doesn't want SDP or O/A in the API at all.  Has a very strong
dislike for it.  Won't tolerate it, even for 1.0.  The IETF and W3C should
be embarrassed for even suggesting such a terrible API.  It's better to go
back to the drawing board (if you think those words are strong, go read the
spreadsheet or mailing list!)

Oh, and of course, for completeness, we should describe Camp #0, which
didn't have any input in this feedback:

Camp #0:  I've used SDP for years and I'm very comfortable with it.  Using
SDP as the control surface really helps my use case, which is legacy
interop.  Defining an API without SDP would be too much work, and probably
fail.  Look at what happened with SDPng!  Supporting all these advanced use
cases doesn't seem worth it.   If developers are doing that much advanced
stuff, they can learn to munge SDP.  It isn't that bad.

(By the way, I'm not a member of Camp #0, so I may be a little off in my
description of it.  If any member of Camp #0 would like to adjust that
description, please feel free).




*Suggested Next Step: Figure out what is needed for WebRTC++*
*
*
As Cullen recently pointed out, we need to get a list of things developers
want to be able to control in a future version of the API which could
hopefully make developers happier Whether that future API is 1.0, 2.0, or
1.1 is an orthogonal question, but let's call it WebRTC++ for now.
Clearly, they want a way to control things without SDP, and perhaps without
O/A at all.  But what do they want to control?  We need a good list.

I suggest we reach out more closely to the developers, and use their input
to create not only a list of things they want to do, but a WebRTC++ API
that fulfills those needs without requiring SDP munging.  I think we can do
this in parallel with the current  API work without detracting from it
(from the current work, that is).    Certainly there is plenty of energy
from people interested in working on it (WebRTC++, that is).  Such parallel
work would not only better capture the needs of developers, but could also
improve the WebRTC (not ++) API  by incorporating ideas from it.

So far, such activity has been frowned on, perhaps out of fear that it's a
threat to the current work.  But I think it can be done in a way that
doesn't threaten anything, and even learns from and builds on the valuable
input we are receiving from developers.  The alternative is to ask all
these developers to wait an indefinite amount of time and munge SDP like
crazy until then, and that seems like a rather poor answer.  Therefore, I
would support an exploratory effort at a "WebRTC++ API".

--089e0160cb9a625ba004e0a1d8b2
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">After about a week of input, and feedback from 19 develope=
rs doing a wide range of things using the WebRTC API, we have some data abo=
ut their opinion of the current API.<div><br></div><div><b>Short answer</b>=
:</div>

<div><br></div><div><div><span style=3D"font-family:&#39;courier new&#39;,m=
onospace">61%=C2=A0</span><font face=3D"courier new, monospace">&quot;Good =
for simple Stuff&quot;<span style=3D"white-space:pre">=C2=A0     </span>=C2=
=A0 49% &quot;Bad for simple stuff&quot;</font></div>

<div><span style=3D"font-family:&#39;courier new&#39;,monospace">=C2=A06%=
=C2=A0</span><font face=3D"courier new, monospace">&quot;Good for advanced =
stuff&quot; =C2=A0 =C2=A0 =C2=A094% &quot;Bad for advanced stuff&quot;</fon=
t></div><div><span style=3D"font-family:&#39;courier new&#39;,monospace">46=
%=C2=A0</span><font face=3D"courier new, monospace">&quot;Good/OK/tolerable=
 for 1.0&quot; =C2=A0 =C2=A054% &quot;Horrible/intolerable for 1.0&quot;</f=
ont></div>

<div><span style=3D"font-family:&#39;courier new&#39;,monospace">=C2=A07%=
=C2=A0</span><font face=3D"courier new, monospace">&quot;Good/OK/tolerable =
long-term&quot; =C2=A093</font><span style=3D"font-family:&#39;courier new&=
#39;,monospace">% &quot;Need a better API long-term&quot;</span><span style=
=3D"font-family:&#39;courier new&#39;,monospace">=C2=A0</span></div>

</div><div><br></div><div><br></div><div>In general the attitude is <b>&quo=
t;It&#39;s OK for simple stuff but bad for advanced stuff&quot;, =C2=A0</b>=
and <b>&quot;We can tolerate it for 1.0, but we need something better for 2=
.0&quot;. =C2=A0</b>But the majority of the feedback was against it even fo=
r the 1.0, so the last one is a bit of a stretch.</div>

<div><br></div><div><br></div><div><b>Long Answer</b>:</div><div><br></div>=
<div><div>I had to interpret the results a little to fit into these buckets=
, so if you don&#39;t like the interpretation, you&#39;re more than welcome=
 to interpret your own results from the raw data: =C2=A0<a href=3D"https://=
docs.google.com/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJL=
RkVPV0E#gid=3D1">https://docs.google.com/spreadsheet/ccc?key=3D0AuaKXw3SkHM=
SdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1</a></div>

</div><div><br></div><div><br></div><div>There seem to be 4 or 5 different =
camps:</div><div><br></div><div>Camp #1: =C2=A0Doing very simple things, do=
esn&#39;t touch the SDP. =C2=A0Everything is fine, except they might have t=
hings they want to control but can&#39;t.</div>

<div><br></div><div>Camp #2: =C2=A0Does somewhat more advanced things, touc=
hes the SDP a little, has pain, anticipates more pain, would like something=
 better, but thinks this is OK for 1.0.</div><div><br></div><div>Camp #3: =
=C2=A0Is trying advanced things, is touching the SDP a lot, and really feel=
s pain. =C2=A0Wants something better, and might tolerate this API for 1.0, =
but not for long.</div>

<div><br></div><div>Camp #4: =C2=A0Doesn&#39;t want SDP or O/A in the API a=
t all. =C2=A0Has a very strong dislike for it. =C2=A0Won&#39;t tolerate it,=
 even for 1.0. =C2=A0The IETF and W3C should be embarrassed for even sugges=
ting such a terrible API. =C2=A0It&#39;s better to go back to the drawing b=
oard (if you think those words are strong, go read the spreadsheet or maili=
ng list!)</div>

<div><br></div><div>Oh, and of course, for completeness, we should describe=
 Camp #0, which didn&#39;t have any input in this feedback:</div><div><br><=
/div><div>Camp #0: =C2=A0I&#39;ve used SDP for years and I&#39;m very comfo=
rtable with it. =C2=A0Using SDP as the control surface really helps my use =
case, which is legacy interop. =C2=A0Defining an API without SDP would be t=
oo much work, and probably fail. =C2=A0Look at what happened with SDPng! =
=C2=A0Supporting all these advanced use cases doesn&#39;t seem worth it. =
=C2=A0 If developers are doing that much advanced stuff, they can learn to =
munge SDP. =C2=A0It isn&#39;t that bad.=C2=A0</div>

<div><br></div><div>(By the way, I&#39;m not a member of Camp #0, so I may =
be a little off in my description of it. =C2=A0If any member of Camp #0 wou=
ld like to adjust that description, please feel free).</div><div><br></div>
<div>
<br></div><div><br></div><div><br></div><div><b>Suggested Next Step: Figure=
 out what is needed for WebRTC++</b></div><div><b><br></b></div><div>As Cul=
len recently pointed out, we need to get a list of things developers want t=
o be able to control in a future version of the API which could hopefully m=
ake developers happier Whether that future API is 1.0, 2.0, or 1.1 is an or=
thogonal question, but let&#39;s call it WebRTC++ for now. =C2=A0 Clearly, =
they want a way to control things without SDP, and perhaps without O/A at a=
ll. =C2=A0But what do they want to control? =C2=A0We need a good list.</div=
>

<div><br></div><div>I suggest we reach out more closely to the developers, =
and use their input to create not only a list of things they want to do, bu=
t a WebRTC++ API that fulfills those needs without requiring SDP munging. =
=C2=A0I think we can do this in parallel with the current =C2=A0API work wi=
thout detracting from it (from the current work, that is). =C2=A0 =C2=A0Cer=
tainly there is plenty of energy from people interested in working on it (W=
ebRTC++, that is). =C2=A0Such parallel work would not only better capture t=
he needs of developers, but could also improve the WebRTC (not ++) API =C2=
=A0by incorporating ideas from it. =C2=A0</div>

<div><br></div><div>So far, such activity has been frowned on, perhaps out =
of fear that it&#39;s a threat to the current work. =C2=A0But I think it ca=
n be done in a way that doesn&#39;t threaten anything, and even learns from=
 and builds on the valuable input we are receiving from developers. =C2=A0T=
he alternative is to ask all these developers to wait an indefinite amount =
of time and munge SDP like crazy until then, and that seems like a rather p=
oor answer. =C2=A0Therefore, I would support an exploratory effort at a &qu=
ot;WebRTC++ API&quot;.</div>

<div><br></div><div><br></div><div><br></div><div><br></div><div><br></div>=
<div><br></div><div>=C2=A0=C2=A0</div></div>

--089e0160cb9a625ba004e0a1d8b2--

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From: Peter Thatcher <pthatcher@google.com>
Date: Wed, 3 Jul 2013 14:10:27 -0700
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Subject: Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
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The more I think about it, the more I think I'll refrain from posting what
I need.  I don't want to make this API just for myself.  I want to make it
for all the developers that are currently using it and the those that will
use it in the future, so I think we should focus on them first.   Perhaps
if we use their input and begin incorporating it into the API, I can
provide secondary input to refine what we work on, but I'd like the primary
input to be from other developers.

I don't want to make the mistake of building an API just for myself and
leaving the developers suffering because I over-focused on myself.   To
avoid that, I think I'll focus on them first.


On Wed, Jul 3, 2013 at 12:48 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Jul 3, 2013, at 12:34 PM, Peter Thatcher <pthatcher@google.com> wrote:
>
> > And I could give a very long list of things, but I'll continue to keep
> myself disqualified :).
> >
> >
>
> Please bring up the long list now - later just delays the discussion =E2=
=80=A6
>
>

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<div dir=3D"ltr">The more I think about it, the more I think I&#39;ll refra=
in from posting what I need. =C2=A0I don&#39;t want to make this API just f=
or myself. =C2=A0I want to make it for all the developers that are currentl=
y using it and the those that will use it in the future, so I think we shou=
ld focus on them first. =C2=A0 Perhaps if we use their input and begin inco=
rporating it into the API, I can provide secondary input to refine what we =
work on, but I&#39;d like the primary input to be from other developers. =
=C2=A0<div>

<br></div><div>I don&#39;t want to make the mistake of building an API just=
 for myself and leaving the developers suffering because I over-focused on =
myself. =C2=A0 To avoid that, I think I&#39;ll focus on them first.=C2=A0</=
div></div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Jul 3=
, 2013 at 12:48 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=
=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Jul 3, 2013, at 12:34 PM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt; And I could give a very long list of things, but I&#39;ll continue to =
keep myself disqualified :).<br>
&gt;<br>
&gt;<br>
<br>
</div>Please bring up the long list now - later just delays the discussion =
=E2=80=A6<br>
<br>
</blockquote></div><br></div>

--bcaec5015cf77e374604e0a1e677--

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 Jul 2013 14:20:41 -0700
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On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <pthatcher@google.com> wrote:

> Camp #0:  I've used SDP for years and I'm very comfortable with it.  Using
> SDP as the control surface really helps my use case, which is legacy
> interop.  Defining an API without SDP would be too much work, and probably
> fail.  Look at what happened with SDPng!  Supporting all these advanced use
> cases doesn't seem worth it.   If developers are doing that much advanced
> stuff, they can learn to munge SDP.  It isn't that bad.
>

Hmm... That's not my understanding of the situation at all.

Rather, I believe the expectation is that you shouldn't have to modify the
SDP but rather there should be API points to cover most of the use cases
that people want. This isn't to say that all those API points exist or that
they work or whatever.

-Ekr

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<div dir=3D"ltr">On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">ptha=
tcher@google.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div c=
lass=3D"gmail_quote">


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>Camp #0: =A0I&#39;ve u=
sed SDP for years and I&#39;m very comfortable with it. =A0Using SDP as the=
 control surface really helps my use case, which is legacy interop. =A0Defi=
ning an API without SDP would be too much work, and probably fail. =A0Look =
at what happened with SDPng! =A0Supporting all these advanced use cases doe=
sn&#39;t seem worth it. =A0 If developers are doing that much advanced stuf=
f, they can learn to munge SDP. =A0It isn&#39;t that bad.=A0</div>



</div></blockquote><div><br></div><div>Hmm... That&#39;s not my understandi=
ng of the situation at all.</div><div><br></div><div>Rather, I believe the =
expectation is that you shouldn&#39;t have to modify the</div>
<div>SDP but rather there should be API points to cover most of the use cas=
es</div><div>that people want. This isn&#39;t to say that all those API poi=
nts exist or that</div><div style>they work or whatever.</div><div style>

<br></div><div>-Ekr</div><div><br></div></div><br></div></div>

--0023544707ac0f40f404e0a20b0f--

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From: Peter Thatcher <pthatcher@google.com>
Date: Wed, 3 Jul 2013 14:25:59 -0700
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On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <pthatcher@google.com>wrote:
>
>> Camp #0:  I've used SDP for years and I'm very comfortable with it.
>>  Using SDP as the control surface really helps my use case, which is legacy
>> interop.  Defining an API without SDP would be too much work, and probably
>> fail.  Look at what happened with SDPng!  Supporting all these advanced use
>> cases doesn't seem worth it.   If developers are doing that much advanced
>> stuff, they can learn to munge SDP.  It isn't that bad.
>>
>
> Hmm... That's not my understanding of the situation at all.
>
> Rather, I believe the expectation is that you shouldn't have to modify the
> SDP but rather there should be API points to cover most of the use cases
> that people want. This isn't to say that all those API points exist or that
> they work or whatever.
>
> -Ekr
>
>
I'm glad to be wrong here.  Is the phrase "you shouldn't have to modify the
SDP but rather there should be API points to cover most of the use cases"
 the consensus of the working group?  That would be (good) news to me.
 Because if it's true, then we can just go ahead and add those API points
and make the developers happy.

Along with that, is "use Jingle for signalling" included in your set of
"most of the use cases"?

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<div dir=3D"ltr">On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <span dir=3D=
"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a=
>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid=
;padding-left:1ex">


<div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <span =
dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">p=
thatcher@google.com</a>&gt;</span> wrote:<br></div><div class=3D"gmail_extr=
a">
<div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div>Camp #0: =C2=A0I&#39;ve used SDP for=
 years and I&#39;m very comfortable with it. =C2=A0Using SDP as the control=
 surface really helps my use case, which is legacy interop. =C2=A0Defining =
an API without SDP would be too much work, and probably fail. =C2=A0Look at=
 what happened with SDPng! =C2=A0Supporting all these advanced use cases do=
esn&#39;t seem worth it. =C2=A0 If developers are doing that much advanced =
stuff, they can learn to munge SDP. =C2=A0It isn&#39;t that bad.=C2=A0</div=
>






</div></blockquote><div><br></div></div><div>Hmm... That&#39;s not my under=
standing of the situation at all.</div><div><br></div><div>Rather, I believ=
e the expectation is that you shouldn&#39;t have to modify the</div>
<div>SDP but rather there should be API points to cover most of the use cas=
es</div><div>that people want. This isn&#39;t to say that all those API poi=
nts exist or that</div><div>they work or whatever.</div><div>

<br></div><div>-Ekr</div><div><br></div></div></div></div></blockquote><div=
><br></div><div>I&#39;m glad to be wrong here. =C2=A0Is the phrase &quot;yo=
u shouldn&#39;t have to modify the SDP but rather there should be API point=
s to cover most of the use cases&quot; =C2=A0the consensus of the working g=
roup? =C2=A0That would be (good) news to me. =C2=A0Because if it&#39;s true=
, then we can just go ahead and add those API points and make the developer=
s happy.</div>


<div><br></div><div>Along with that, is &quot;use Jingle for signalling&quo=
t; included in your set of &quot;most of the use cases&quot;?=C2=A0</div></=
div><br></div></div>

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El 03/07/2013 23:26, "Peter Thatcher" <pthatcher@google.com> escribi=C3=B3:
>

> Along with that, is "use Jingle for signalling" included in your set of
"most of the use cases"?

+1 good question. Hope there is a so good answer for it. Is it?

BTW not just Jingle but any (future?) media signaling protocol not based on
atomic full SDP blob exchange. Could somebody explain me how to do that
without mangling the SDP blob generated by PC?

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<p dir=3D"ltr"><br>
El 03/07/2013 23:26, &quot;Peter Thatcher&quot; &lt;<a href=3D"mailto:pthat=
cher@google.com">pthatcher@google.com</a>&gt; escribi=C3=B3:<br>
&gt;</p>
<p dir=3D"ltr">&gt; Along with that, is &quot;use Jingle for signalling&quo=
t; included in your set of &quot;most of the use cases&quot;?=C2=A0</p>
<p dir=3D"ltr">+1 good question. Hope there is a so good answer for it. Is =
it?</p>
<p dir=3D"ltr">BTW not just Jingle but any (future?) media signaling protoc=
ol not based on atomic full SDP blob exchange. Could somebody explain me ho=
w to do that without mangling the SDP blob generated by PC?</p>

--047d7bdc7de8c04ba904e0a23e2d--

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On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <pthatcher@google.com> wrote:

> On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <pthatcher@google.com>wrote:
>>
>>> Camp #0:  I've used SDP for years and I'm very comfortable with it.
>>>  Using SDP as the control surface really helps my use case, which is legacy
>>> interop.  Defining an API without SDP would be too much work, and probably
>>> fail.  Look at what happened with SDPng!  Supporting all these advanced use
>>> cases doesn't seem worth it.   If developers are doing that much advanced
>>> stuff, they can learn to munge SDP.  It isn't that bad.
>>>
>>
>> Hmm... That's not my understanding of the situation at all.
>>
>> Rather, I believe the expectation is that you shouldn't have to modify the
>> SDP but rather there should be API points to cover most of the use cases
>> that people want. This isn't to say that all those API points exist or
>> that
>> they work or whatever.
>>
>> -Ekr
>>
>>
> I'm glad to be wrong here.  Is the phrase "you shouldn't have to modify
> the SDP but rather there should be API points to cover most of the use
> cases"  the consensus of the working group?
>

I thought it was, but I'm not the chair, so maybe you could ask Harald or
Stefan.


 Along with that, is "use Jingle for signalling" included in your set of
> "most of the use cases"?
>

No, I don't think it is, since it's not SDP.

Maybe I could phrase this differently: It was my understanding that you
should have
API points to get the PC to emit SDP that would express the policies,
preferences,
actions, etc. that the application wants. If you want something that's not
SDP you would
need to gateway.

This may or may not be a satisfactory state of affairs, but it was the rule
I was using
(based on what I thought the WG expected) for when use cases needed new API
points.

-Ekr

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<div dir=3D"ltr">On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">ptha=
tcher@google.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div c=
lass=3D"gmail_quote">


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>On Wed, Jul 3, 20=
13 at 2:20 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>


</div></div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><div=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;=
padding-left:1ex">





<div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <span =
dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">p=
thatcher@google.com</a>&gt;</span> wrote:<br></div><div class=3D"gmail_extr=
a">
<div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div>Camp #0: =A0I&#39;ve used SDP for ye=
ars and I&#39;m very comfortable with it. =A0Using SDP as the control surfa=
ce really helps my use case, which is legacy interop. =A0Defining an API wi=
thout SDP would be too much work, and probably fail. =A0Look at what happen=
ed with SDPng! =A0Supporting all these advanced use cases doesn&#39;t seem =
worth it. =A0 If developers are doing that much advanced stuff, they can le=
arn to munge SDP. =A0It isn&#39;t that bad.=A0</div>









</div></blockquote><div><br></div></div><div>Hmm... That&#39;s not my under=
standing of the situation at all.</div><div><br></div><div>Rather, I believ=
e the expectation is that you shouldn&#39;t have to modify the</div>
<div>SDP but rather there should be API points to cover most of the use cas=
es</div><div>that people want. This isn&#39;t to say that all those API poi=
nts exist or that</div><div>they work or whatever.</div><div>

<br></div><div>-Ekr</div><div><br></div></div></div></div></blockquote><div=
><br></div></div></div><div>I&#39;m glad to be wrong here. =A0Is the phrase=
 &quot;you shouldn&#39;t have to modify the SDP but rather there should be =
API points to cover most of the use cases&quot; =A0the consensus of the wor=
king group? =A0</div>


</div></div></div></blockquote><div><br></div><div>I thought it was, but I&=
#39;m not the chair, so maybe you could ask Harald or</div><div>Stefan.</di=
v><div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>
Along with that, is &quot;use Jingle for signalling&quot; included in your =
set of &quot;most of the use cases&quot;?=A0</div></div></div></div></block=
quote><div><br></div><div style>No, I don&#39;t think it is, since it&#39;s=
 not SDP.</div>

<div style><br></div><div style>Maybe I could phrase this differently: It w=
as my understanding that you should have</div><div style>API points to get =
the PC to emit SDP that would express the policies, preferences,</div>
<div style>
actions, etc. that the application wants. If you want something that&#39;s =
not SDP you would</div><div style>need to gateway.</div><div style><br></di=
v><div style>This may or may not be a satisfactory state of affairs, but it=
 was the rule I was using</div>

<div style>(based on what I thought the WG expected) for when use cases nee=
ded new API</div><div style>points.</div><div style><br></div><div>-Ekr</di=
v><div>=A0</div></div><br></div></div>

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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So compatibility with SIP is important but compatibility with Jingle is
just impossible. And this is supposed to be the API proposed to W3C...

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 03/07/2013 23:37, "Eric Rescorla" <ekr@rtfm.com> escribi=C3=B3:

> On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <pthatcher@google.com>wrot=
e:
>
>> On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>>> On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <pthatcher@google.com>wr=
ote:
>>>
>>>> Camp #0:  I've used SDP for years and I'm very comfortable with it.
>>>>  Using SDP as the control surface really helps my use case, which is l=
egacy
>>>> interop.  Defining an API without SDP would be too much work, and prob=
ably
>>>> fail.  Look at what happened with SDPng!  Supporting all these advance=
d use
>>>> cases doesn't seem worth it.   If developers are doing that much advan=
ced
>>>> stuff, they can learn to munge SDP.  It isn't that bad.
>>>>
>>>
>>> Hmm... That's not my understanding of the situation at all.
>>>
>>> Rather, I believe the expectation is that you shouldn't have to modify
>>> the
>>> SDP but rather there should be API points to cover most of the use case=
s
>>> that people want. This isn't to say that all those API points exist or
>>> that
>>> they work or whatever.
>>>
>>> -Ekr
>>>
>>>
>> I'm glad to be wrong here.  Is the phrase "you shouldn't have to modify
>> the SDP but rather there should be API points to cover most of the use
>> cases"  the consensus of the working group?
>>
>
> I thought it was, but I'm not the chair, so maybe you could ask Harald or
> Stefan.
>
>
>  Along with that, is "use Jingle for signalling" included in your set of
>> "most of the use cases"?
>>
>
> No, I don't think it is, since it's not SDP.
>
> Maybe I could phrase this differently: It was my understanding that you
> should have
> API points to get the PC to emit SDP that would express the policies,
> preferences,
>  actions, etc. that the application wants. If you want something that's
> not SDP you would
> need to gateway.
>
> This may or may not be a satisfactory state of affairs, but it was the
> rule I was using
> (based on what I thought the WG expected) for when use cases needed new A=
PI
> points.
>
> -Ekr
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<p dir=3D"ltr">So compatibility with SIP is important but compatibility wit=
h Jingle is just impossible. And this is supposed to be the API proposed to=
 W3C...<br></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 03/07/2013 23:37, &quot;Eric Rescorla&quot; &=
lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; escribi=C3=B3:<br t=
ype=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr">On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">ptha=
tcher@google.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div c=
lass=3D"gmail_quote">



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>On Wed, Jul 3, 20=
13 at 2:20 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>



</div></div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><div=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;=
padding-left:1ex">






<div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <span =
dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">p=
thatcher@google.com</a>&gt;</span> wrote:<br></div><div class=3D"gmail_extr=
a">
<div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div>Camp #0: =C2=A0I&#39;ve used SDP for=
 years and I&#39;m very comfortable with it. =C2=A0Using SDP as the control=
 surface really helps my use case, which is legacy interop. =C2=A0Defining =
an API without SDP would be too much work, and probably fail. =C2=A0Look at=
 what happened with SDPng! =C2=A0Supporting all these advanced use cases do=
esn&#39;t seem worth it. =C2=A0 If developers are doing that much advanced =
stuff, they can learn to munge SDP. =C2=A0It isn&#39;t that bad.=C2=A0</div=
>










</div></blockquote><div><br></div></div><div>Hmm... That&#39;s not my under=
standing of the situation at all.</div><div><br></div><div>Rather, I believ=
e the expectation is that you shouldn&#39;t have to modify the</div>
<div>SDP but rather there should be API points to cover most of the use cas=
es</div><div>that people want. This isn&#39;t to say that all those API poi=
nts exist or that</div><div>they work or whatever.</div><div>

<br></div><div>-Ekr</div><div><br></div></div></div></div></blockquote><div=
><br></div></div></div><div>I&#39;m glad to be wrong here. =C2=A0Is the phr=
ase &quot;you shouldn&#39;t have to modify the SDP but rather there should =
be API points to cover most of the use cases&quot; =C2=A0the consensus of t=
he working group? =C2=A0</div>



</div></div></div></blockquote><div><br></div><div>I thought it was, but I&=
#39;m not the chair, so maybe you could ask Harald or</div><div>Stefan.</di=
v><div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>
Along with that, is &quot;use Jingle for signalling&quot; included in your =
set of &quot;most of the use cases&quot;?=C2=A0</div></div></div></div></bl=
ockquote><div><br></div><div>No, I don&#39;t think it is, since it&#39;s no=
t SDP.</div>


<div><br></div><div>Maybe I could phrase this differently: It was my unders=
tanding that you should have</div><div>API points to get the PC to emit SDP=
 that would express the policies, preferences,</div>
<div>
actions, etc. that the application wants. If you want something that&#39;s =
not SDP you would</div><div>need to gateway.</div><div><br></div><div>This =
may or may not be a satisfactory state of affairs, but it was the rule I wa=
s using</div>


<div>(based on what I thought the WG expected) for when use cases needed ne=
w API</div><div>points.</div><div><br></div><div>-Ekr</div><div>=C2=A0</div=
></div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--047d7b6d8b765c1da804e0a25760--

From hadriel.kaplan@oracle.com  Wed Jul  3 14:43:27 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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You might also want to cite some of the reasons noted in this old draft:
http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01

-hadriel


On Jun 26, 2013, at 12:37 AM, Robin Raymond <robin@hookflash.com> wrote:

>=20
> According
>  to many, real adoption of WebRTC will not happen if we continue to=20
> force everyone to use this SDP Offer/Answer methodology.  It is =
clearly=20
> blocking our way forward and the amount of specification documentation=20=

> remaining needed for the browser vendors to produce a compatible SDP=20=

> based WebRTC engine in a browser is much more daunting than most are=20=

> willing to admit.
>=20
>=20
> The
>  draft rationale behind a JavaScript Object API model:
>=20
> =
http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api=
-rationale-00.txt
>=20
> =20
>=20
> Whereas:
> The
>  WebRTC JavaScript Object API & Shim document will be along shortly.
>=20
>=20
> We
>  wanted to get this initial rationale draft informational document out=20=

> right away. Everyone that has any interest should review the reasons =
and
>  concepts and comment before we get too far along on the details of an=20=

> actual API, the initial API will follow in the coming days.
>=20
>=20
> Best
>  regards,
>=20
> Robin
>  Raymond
> =20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On Wed, Jul 3, 2013 at 2:42 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> So compatibility with SIP is important but compatibility with Jingle is
> just impossible. And this is supposed to be the API proposed to W3C...
>
Who said anything about impossible? It's a mechanical transformation
(see: http://xmpp.org/extensions/xep-0167.html).

Anyway, it's not like this feature is a surprise to anyone--well at
least anyone who was paying attention--it's been a feature of the
specification since before the WG was even formed. As I said
earlier, it was in the original WHATWG spec that Ian Hickson
wrote.

-Ekr

--001a1132eee653848704e0a2a900
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<div dir=3D"ltr">On Wed, Jul 3, 2013 at 2:42 PM, I=F1aki Baz Castillo <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@ali=
ax.net</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=3D"gm=
ail_quote">


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><p dir=3D"ltr">So compatibility with SIP is important but =
compatibility with Jingle is just impossible. And this is supposed to be th=
e API proposed to W3C...</p>


</blockquote><div>Who said anything about impossible? It&#39;s a mechanical=
 transformation<br></div><div>(see:=A0<a href=3D"http://xmpp.org/extensions=
/xep-0167.html" target=3D"_blank">http://xmpp.org/extensions/xep-0167.html<=
/a>).<br>

</div><div><br></div>
<div style>Anyway, it&#39;s not like this feature is a surprise to anyone--=
well at</div><div style>least anyone who was paying attention--it&#39;s bee=
n a feature of the</div><div style>specification since before the WG was ev=
en formed. As I said</div>

<div style>earlier, it was in the original WHATWG spec that Ian Hickson</di=
v><div style>wrote.</div><div style><br></div><div>-Ekr<br></div><div><br><=
/div><div><br></div><div><br></div></div></div></div>

--001a1132eee653848704e0a2a900--

From martin.thomson@gmail.com  Wed Jul  3 15:07:47 2013
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On 3 July 2013 14:43, Hadriel Kaplan <hadriel.kaplan@oracle.com> wrote:
> You might also want to cite some of the reasons noted in this old draft:
> http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01

Yes. This is an excellent draft that covers a lot of the same issues.
I especially like:

   4) That when a IETF documents start telling you how to build
     Javascript APIs, you should run far away... quickly.  :)

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On Wed, Jul 3, 2013 at 3:04 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> On Wed, Jul 3, 2013 at 2:42 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
>
>> So compatibility with SIP is important but compatibility with Jingle is
>> just impossible. And this is supposed to be the API proposed to W3C...
>>
> Who said anything about impossible? It's a mechanical transformation
> (see: http://xmpp.org/extensions/xep-0167.html).
>
> Anyway, it's not like this feature is a surprise to anyone--well at
> least anyone who was paying attention--it's been a feature of the
> specification since before the WG was even formed. As I said
> earlier, it was in the original WHATWG spec that Ian Hickson
> wrote.
>
> -Ekr
>
>
There are lots of application developers who weren't around back then, so
it's not reasonable to expect them to have been paying attention back then.
 Many are very new to this, and there are lots of surprises for them.

And as many developers are now trying to use the WebRTC API, we're now
getting feedback on how well it works for them.  The point of my original
post in this thread was to summarize the feedback we've received so far for
the WG.

It wasn't my intention to re-start arguments between one camp and another.
  But your clarification of "Camp #0" was helpful, so thank for you for
that.

--bcaec51f99e513f0f804e0a2d43d
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:04 PM, Eric Rescorla <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</sp=
an> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at=
 2:42 PM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:i=
bc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>


</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><p dir=3D"ltr">So compatibility with SIP is important but =
compatibility with Jingle is just impossible. And this is supposed to be th=
e API proposed to W3C...</p>





</blockquote></div><div>Who said anything about impossible? It&#39;s a mech=
anical transformation<br></div><div>(see:=C2=A0<a href=3D"http://xmpp.org/e=
xtensions/xep-0167.html" target=3D"_blank">http://xmpp.org/extensions/xep-0=
167.html</a>).<br>




</div><div><br></div>
<div>Anyway, it&#39;s not like this feature is a surprise to anyone--well a=
t</div><div>least anyone who was paying attention--it&#39;s been a feature =
of the</div><div>specification since before the WG was even formed. As I sa=
id</div>




<div>earlier, it was in the original WHATWG spec that Ian Hickson</div><div=
>wrote.</div><div><br></div><div>-Ekr<br></div><div><br></div></div></div><=
/div></blockquote><div><br></div><div>There are lots of application develop=
ers who weren&#39;t around back then, so it&#39;s not reasonable to expect =
them to have been paying attention back then. =C2=A0Many are very new to th=
is, and there are lots of surprises for them.<br>


</div><div><br></div><div>And as many developers are now trying to use the =
WebRTC API, we&#39;re now getting feedback on how well it works for them. =
=C2=A0The point of my original post in this thread was to summarize the fee=
dback we&#39;ve received so far for the WG.</div>

<div><br></div><div>It wasn&#39;t my intention to re-start arguments betwee=
n one camp and another. =C2=A0 But your clarification of &quot;Camp #0&quot=
; was helpful, so thank for you for that.<br></div>
<div><br></div></div><br></div></div>

--bcaec51f99e513f0f804e0a2d43d--

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2013/7/4 Eric Rescorla <ekr@rtfm.com>:
> Anyway, it's not like this feature is a surprise to anyone--well at
> least anyone who was paying attention--it's been a feature of the
> specification since before the WG was even formed. As I said
> earlier, it was in the original WHATWG spec that Ian Hickson
> wrote.

I know, and I was a  "SDP supporter" at the beginning (and much others
like me who, after dealing with it for something more than just single
audio+video phone calls, have changed their mind). I don't consider
that such a "original WHATWG spec" is written in a stone. It was an
initial proposal after all. Nothing prevents it to be radically
modified/improved now.

Best regards.




--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 4 Jul 2013 00:20:26 +0200
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2013/7/3 Hadriel Kaplan <hadriel.kaplan@oracle.com>:
> You might also want to cite some of the reasons noted in this old draft:
> http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01

Well, section "3. Defining a WebRTC Protocol in the Browser" says:


   9) Using the SDP offer/answer model provides a more rigid API
     interaction model, enabling Browser vendors to perform less
     testing and provide more robust implementations than exposing all
     discrete components to a Javascript API would.

   10)   Using a higher-level API model, such as would be done with an
     SDP offer/answer model, means the cross-browser vendor-specific
     variances would be reduced.  Exposing a lower-level API would
     inevitably lead to some differences in different browsers due to
     differences in their architectures/implementation.


Time to reconsider those assertions? ;)


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 Jul 2013 15:20:49 -0700
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On Wed, Jul 3, 2013 at 3:16 PM, Peter Thatcher <pthatcher@google.com> wrote=
:

>
>
>
> On Wed, Jul 3, 2013 at 3:04 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> On Wed, Jul 3, 2013 at 2:42 PM, I=F1aki Baz Castillo <ibc@aliax.net> wro=
te:
>>
>>> So compatibility with SIP is important but compatibility with Jingle is
>>> just impossible. And this is supposed to be the API proposed to W3C...
>>>
>> Who said anything about impossible? It's a mechanical transformation
>> (see: http://xmpp.org/extensions/xep-0167.html).
>>
>> Anyway, it's not like this feature is a surprise to anyone--well at
>> least anyone who was paying attention--it's been a feature of the
>> specification since before the WG was even formed. As I said
>> earlier, it was in the original WHATWG spec that Ian Hickson
>> wrote.
>>
>>

>  There are lots of application developers who weren't around back then,
> so it's not reasonable to expect them to have been paying attention back
> then.  Many are very new to this, and there are lots of surprises for the=
m.
>

The person I was responding to has posts on this mailing list going back as
far as September 2011.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:16 PM, Peter Thatcher <span dir=3D"ltr">&l=
t;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@googl=
e.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Wed, Jul 3=
, 2013 at 3:04 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ek=
r@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>




<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at=
 2:42 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@=
aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>




</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><p dir=3D"ltr">So compatibility with SIP is important but =
compatibility with Jingle is just impossible. And this is supposed to be th=
e API proposed to W3C...</p>







</blockquote></div><div>Who said anything about impossible? It&#39;s a mech=
anical transformation<br></div><div>(see:=A0<a href=3D"http://xmpp.org/exte=
nsions/xep-0167.html" target=3D"_blank">http://xmpp.org/extensions/xep-0167=
.html</a>).<br>






</div><div><br></div>
<div>Anyway, it&#39;s not like this feature is a surprise to anyone--well a=
t</div><div>least anyone who was paying attention--it&#39;s been a feature =
of the</div><div>specification since before the WG was even formed. As I sa=
id</div>






<div>earlier, it was in the original WHATWG spec that Ian Hickson</div><div=
>wrote.</div><div><br></div></div></div></div></blockquote></div></div></di=
v></div></div></blockquote><div>=A0</div><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><div class=3D"h5"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div cla=
ss=3D"gmail_extra">

<div class=3D"gmail_quote"><div></div></div></div></div></blockquote></div>=
</div><div>There are lots of application developers who weren&#39;t around =
back then, so it&#39;s not reasonable to expect them to have been paying at=
tention back then. =A0Many are very new to this, and there are lots of surp=
rises for them.</div>

</div></div></div></blockquote><div><br></div><div style>The person I was r=
esponding to has posts on this mailing list going back as far as September =
2011.</div><div style><br></div><div style>-Ekr</div><div style><br></div>

</div></div></div>

--485b397dd205192c6d04e0a2e24a--

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 Jul 2013 15:27:43 -0700
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On Wed, Jul 3, 2013 at 3:17 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/4 Eric Rescorla <ekr@rtfm.com>:
> > Anyway, it's not like this feature is a surprise to anyone--well at
> > least anyone who was paying attention--it's been a feature of the
> > specification since before the WG was even formed. As I said
> > earlier, it was in the original WHATWG spec that Ian Hickson
> > wrote.
>
> I know, and I was a  "SDP supporter" at the beginning (and much others
> like me who, after dealing with it for something more than just single
> audio+video phone calls, have changed their mind).


This leaves me puzzled by your complaint that the current specification
doesn't support Jingle out of the box, since that's been obvious from the
very beginning and it's not like that's something that required extensive
developer experience to discern. If this is important, why didn't you
complain then?

-Ekr

--047d7b677510c4773c04e0a2fa8a
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:17 PM, I=F1aki Baz Castillo <span dir=3D"l=
tr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a=
>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">2013/7/4 Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com"=
 target=3D"_blank">ekr@rtfm.com</a>&gt;:<br>



<div>&gt; Anyway, it&#39;s not like this feature is a surprise to anyone--w=
ell at<br>
&gt; least anyone who was paying attention--it&#39;s been a feature of the<=
br>
&gt; specification since before the WG was even formed. As I said<br>
&gt; earlier, it was in the original WHATWG spec that Ian Hickson<br>
&gt; wrote.<br>
<br>
</div>I know, and I was a =A0&quot;SDP supporter&quot; at the beginning (an=
d much others<br>
like me who, after dealing with it for something more than just single<br>
audio+video phone calls, have changed their mind).=A0</blockquote><div><br>=
</div><div><div>This leaves me puzzled by your complaint that the current s=
pecification</div><div>doesn&#39;t support Jingle out of the box, since tha=
t&#39;s been obvious from the</div>

<div>very beginning and it&#39;s not like that&#39;s something that require=
d extensive</div><div>developer experience to discern. If this is important=
, why didn&#39;t you</div><div>complain then?</div></div><div><br></div>

<div>-Ekr</div></div></div></div>

--047d7b677510c4773c04e0a2fa8a--

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From: Peter Thatcher <pthatcher@google.com>
Date: Wed, 3 Jul 2013 15:28:07 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Wed, Jul 3, 2013 at 3:20 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
>
> On Wed, Jul 3, 2013 at 3:16 PM, Peter Thatcher <pthatcher@google.com>wrot=
e:
>
>>
>>
>>
>> On Wed, Jul 3, 2013 at 3:04 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>>> On Wed, Jul 3, 2013 at 2:42 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net>=
wrote:
>>>
>>>> So compatibility with SIP is important but compatibility with Jingle i=
s
>>>> just impossible. And this is supposed to be the API proposed to W3C...
>>>>
>>> Who said anything about impossible? It's a mechanical transformation
>>> (see: http://xmpp.org/extensions/xep-0167.html).
>>>
>>> Anyway, it's not like this feature is a surprise to anyone--well at
>>> least anyone who was paying attention--it's been a feature of the
>>> specification since before the WG was even formed. As I said
>>> earlier, it was in the original WHATWG spec that Ian Hickson
>>> wrote.
>>>
>>>
>
>>  There are lots of application developers who weren't around back then,
>> so it's not reasonable to expect them to have been paying attention back
>> then.  Many are very new to this, and there are lots of surprises for th=
em.
>>
>
> The person I was responding to has posts on this mailing list going back
> as far as September 2011.
>
> -Ekr
>

You said "it's not like this feature is a surprise to anyone",  and I
thought "anyone" was a bit broader than that.  Sorry for the
misunderstanding.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:20 PM, Eric Rescorla <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><d=
iv class=3D"gmail_quote">

<div class=3D"im">On Wed, Jul 3, 2013 at 3:16 PM, Peter Thatcher <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">ptha=
tcher@google.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><d=
iv class=3D"gmail_quote">

<div><div>On Wed, Jul 3, 2013 at 3:04 PM, Eric Rescorla <span dir=3D"ltr">&=
lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</=
span> wrote:<br>




<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div>On Wed, Jul 3, 2013 at 2:42 PM, I=C3=
=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" =
target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>






</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><p dir=3D"ltr">So compatibility with SIP is important but =
compatibility with Jingle is just impossible. And this is supposed to be th=
e API proposed to W3C...</p>









</blockquote></div><div>Who said anything about impossible? It&#39;s a mech=
anical transformation<br></div><div>(see:=C2=A0<a href=3D"http://xmpp.org/e=
xtensions/xep-0167.html" target=3D"_blank">http://xmpp.org/extensions/xep-0=
167.html</a>).<br>








</div><div><br></div>
<div>Anyway, it&#39;s not like this feature is a surprise to anyone--well a=
t</div><div>least anyone who was paying attention--it&#39;s been a feature =
of the</div><div>specification since before the WG was even formed. As I sa=
id</div>








<div>earlier, it was in the original WHATWG spec that Ian Hickson</div><div=
>wrote.</div><div><br></div></div></div></div></blockquote></div></div></di=
v></div></div></blockquote><div>=C2=A0</div></div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-lef=
t-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">



<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:s=
olid;padding-left:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra">

<div class=3D"gmail_quote"><div></div></div></div></div></blockquote></div>=
</div><div>There are lots of application developers who weren&#39;t around =
back then, so it&#39;s not reasonable to expect them to have been paying at=
tention back then. =C2=A0Many are very new to this, and there are lots of s=
urprises for them.</div>



</div></div></div></blockquote><div><br></div><div>The person I was respond=
ing to has posts on this mailing list going back as far as September 2011.<=
/div><div><br></div><div>-Ekr</div></div></div></div></blockquote><div>

<br></div><div>You said &quot;<span style=3D"color:rgb(80,0,80)">it&#39;s n=
ot like this feature is a surprise to anyone&quot;, =C2=A0and I thought &qu=
ot;anyone&quot; was a bit broader than that. =C2=A0Sorry for the misunderst=
anding.</span></div>

<div><br></div><div>=C2=A0</div></div><br></div></div>

--e89a8fb208343dbc9c04e0a2fc14--

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 Jul 2013 15:34:04 -0700
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On Wed, Jul 3, 2013 at 3:27 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
>
> On Wed, Jul 3, 2013 at 3:17 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:
>
>> 2013/7/4 Eric Rescorla <ekr@rtfm.com>:
>> > Anyway, it's not like this feature is a surprise to anyone--well at
>> > least anyone who was paying attention--it's been a feature of the
>> > specification since before the WG was even formed. As I said
>> > earlier, it was in the original WHATWG spec that Ian Hickson
>> > wrote.
>>
>> I know, and I was a  "SDP supporter" at the beginning (and much others
>> like me who, after dealing with it for something more than just single
>> audio+video phone calls, have changed their mind).
>
>
> This leaves me puzzled by your complaint that the current specification
> doesn't support Jingle out of the box, since that's been obvious from the
> very beginning and it's not like that's something that required extensive
> developer experience to discern. If this is important, why didn't you
> complain then?
>

Looking back at your posts from back then, I see you did in fact suggest
a set of API modifications that would have included an abstract session
description. That would in fact have made Jingle somewhat (though not
really that much) easier. So my last sentence here is unfair. My apologies.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:27 PM, Eric Rescorla <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"im">On Wed, Jul 3, 201=
3 at 3:17 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:=
ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">2013/7/4 Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com"=
 target=3D"_blank">ekr@rtfm.com</a>&gt;:<br>




<div>&gt; Anyway, it&#39;s not like this feature is a surprise to anyone--w=
ell at<br>
&gt; least anyone who was paying attention--it&#39;s been a feature of the<=
br>
&gt; specification since before the WG was even formed. As I said<br>
&gt; earlier, it was in the original WHATWG spec that Ian Hickson<br>
&gt; wrote.<br>
<br>
</div>I know, and I was a =A0&quot;SDP supporter&quot; at the beginning (an=
d much others<br>
like me who, after dealing with it for something more than just single<br>
audio+video phone calls, have changed their mind).=A0</blockquote><div><br>=
</div></div><div><div>This leaves me puzzled by your complaint that the cur=
rent specification</div><div>doesn&#39;t support Jingle out of the box, sin=
ce that&#39;s been obvious from the</div>


<div>very beginning and it&#39;s not like that&#39;s something that require=
d extensive</div><div>developer experience to discern. If this is important=
, why didn&#39;t you</div><div>complain then?</div></div></div></div></div>

</blockquote><div><br></div><div>Looking back at your posts from back then,=
 I see you did in fact suggest</div><div style>a set of API modifications t=
hat would have included an abstract session</div><div style>description. Th=
at would in fact have made Jingle somewhat (though not</div>

<div style>really that much) easier. So my last sentence here is unfair. My=
 apologies.</div><div style><br></div><div style>-Ekr</div><div style><br><=
/div></div></div></div>

--047d7b671fa679852904e0a31164--

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Wed, 3 Jul 2013, Eric Rescorla wrote:
> Who said anything about impossible? It's a mechanical transformation
> (see: http://xmpp.org/extensions/xep-0167.html).

That is true for simple usage and works reasonably well. See my
strophe.jingle code on github. I'd note that I want to revamp
the API such that it wraps the peerconnections SDP apis more closely.

Things like the content-add/content-remove are harder because the API 
user is required to calculate an SDP delta/diff. The appendix A.1.3 of the 
JSEP draft calls this createContentAdd, parseContentAdd, 
createContentAccept and parseContentAccept. Has anyone ever implemented 
those operations in javascript and can demonstrate running code?
--683466026-978727032-1372919747=:14960--

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2013/7/4 Eric Rescorla <ekr@rtfm.com>:
> Looking back at your posts from back then, I see you did in fact suggest
> a set of API modifications that would have included an abstract session
> description. That would in fact have made Jingle somewhat (though not
> really that much) easier. So my last sentence here is unfair. My apologie=
s.

Initially I though that SessionDescription was a powerful JS Object
with all the fields and attributes to construct a SDP, so a developer
could extract those media/transport fields and create a SDP via JS. At
the same time, the SessionDescription object would include helper
methods (defined by the WebRTC API) to build a plain SDP (i.e. toSDP)
and some others like XML-SDP (i.e. toXMLSDP).



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Thursday 04 July 2013 01:51 PM, IÃ±aki Baz Castillo wrote:
> 2013/7/4 Eric Rescorla <ekr@rtfm.com>:
>> Looking back at your posts from back then, I see you did in fact suggest
>> a set of API modifications that would have included an abstract session
>> description. That would in fact have made Jingle somewhat (though not
>> really that much) easier. So my last sentence here is unfair. My apologies.
> Initially I though that SessionDescription was a powerful JS Object
> with all the fields and attributes to construct a SDP, so a developer
> could extract those media/transport fields and create a SDP via JS. At
> the same time, the SessionDescription object would include helper
> methods (defined by the WebRTC API) to build a plain SDP (i.e. toSDP)
> and some others like XML-SDP (i.e. toXMLSDP).
I like that approach. Have an API to specify all/most of what is
in the SDP and then use a toSDP to generate actual SDP string to be sent
across the wire.

Other side should be able to reconstruct the browser side
object(s) using the received SDP string.

thanks,
Binod.
>
>
>
> --
> IÃ±aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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+1 on object model with a "toSDP / fromSDP"  (FYI - an object model 
draft is looming shortly)

I think it's best to put "toSDP / fromSDP" outside the browser binary as 
a common github JS library because I know a lot of SIP folks love custom 
SDP extensions and being able to fork/tweak the SDP without petitioning 
the browser vendors for an "update" is important.

-Robin

> Binod <mailto:binod.pg@oracle.com>
> 4 July, 2013 5:10 AM
> On Thursday 04 July 2013 01:51 PM, IÃ±aki Baz Castillo wrote:
>> 2013/7/4 Eric Rescorla <ekr@rtfm.com>:
>>> Looking back at your posts from back then, I see you did in fact 
>>> suggest
>>> a set of API modifications that would have included an abstract session
>>> description. That would in fact have made Jingle somewhat (though not
>>> really that much) easier. So my last sentence here is unfair. My 
>>> apologies.
>> Initially I though that SessionDescription was a powerful JS Object
>> with all the fields and attributes to construct a SDP, so a developer
>> could extract those media/transport fields and create a SDP via JS. At
>> the same time, the SessionDescription object would include helper
>> methods (defined by the WebRTC API) to build a plain SDP (i.e. toSDP)
>> and some others like XML-SDP (i.e. toXMLSDP).
> I like that approach. Have an API to specify all/most of what is
> in the SDP and then use a toSDP to generate actual SDP string to be sent
> across the wire.
>
> Other side should be able to reconstruct the browser side
> object(s) using the received SDP string.
>
> thanks,
> Binod.

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<html><head>
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
</head><body bgcolor="#FFFFFF" text="#000000"><span><br>
+1 on object model with a "toSDP / fromSDP"Â  (FYI - an object model 
draft is looming shortly)<br>
<br>
I think it's best to put "toSDP / fromSDP" outside the browser binary
 as a common github JS library because I know a lot of SIP folks love 
custom SDP extensions and being able to fork/tweak the SDP without 
petitioning the browser vendors for an "update" is important.<br>
<br>
-Robin<br>
  <br>
</span>
<blockquote style="border: 0px none;" 
cite="mid:51D53C07.6040402@oracle.com" type="cite">
  <div style="margin:30px 25px 10px 25px;" class="__pbConvHr"><div 
style="display:table;width:100%;border-top:1px solid 
#EDEEF0;padding-top:5px"> 	<div 
style="display:table-cell;vertical-align:middle;padding-right:6px;"><img
 photoaddress="binod.pg@oracle.com" photoname="Binod" 
src="cid:part1.02070702.06040608@hookflash.com" 
name="compose-unknown-contact.jpg" height="25px" width="25px"></div>   <div
 
style="display:table-cell;white-space:nowrap;vertical-align:middle;width:100%">
   	<a moz-do-not-send="true" href="mailto:binod.pg@oracle.com" 
style="color:#737F92 
!important;padding-right:6px;font-weight:bold;text-decoration:none 
!important;">Binod</a></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;">   
  <font color="#9FA2A5"><span style="padding-left:6px">4 July, 2013 5:10
 AM</span></font></div></div></div>
  <div style="color:#888888;margin-left:24px;margin-right:24px;" 
__pbrmquotes="true" class="__pbConvBody">On Thursday 04 July 2013 01:51 
PM, IÃ±aki Baz Castillo wrote:
<br><blockquote type="cite">2013/7/4 Eric Rescorla <a class="moz-txt-link-rfc2396E" href="mailto:ekr@rtfm.com">&lt;ekr@rtfm.com&gt;</a>:
<br><blockquote type="cite">Looking back at your posts from back then, I
 see you did in fact suggest
<br>a set of API modifications that would have included an abstract 
session
<br>description. That would in fact have made Jingle somewhat (though 
not
<br>really that much) easier. So my last sentence here is unfair. My 
apologies.
<br></blockquote>Initially I though that SessionDescription was a 
powerful JS Object
<br>with all the fields and attributes to construct a SDP, so a 
developer
<br>could extract those media/transport fields and create a SDP via JS. 
At
<br>the same time, the SessionDescription object would include helper
<br>methods (defined by the WebRTC API) to build a plain SDP (i.e. 
toSDP)
<br>and some others like XML-SDP (i.e. toXMLSDP).
<br></blockquote>I like that approach. Have an API to specify all/most 
of what is
<br>in the SDP and then use a toSDP to generate actual SDP string to be 
sent
<br>across the wire.
<br>
<br>Other side should be able to reconstruct the browser side
<br>object(s) using the received SDP string.
<br>
<br>thanks,
<br>Binod.
  </div>
</blockquote>
</body></html>

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--------------030309080505080209080703--

From matthew@matthew.at  Thu Jul  4 08:03:58 2013
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From: Matthew Kaufman <matthew@matthew.at>
Date: Thu, 4 Jul 2013 08:03:45 -0700
To: Binod <binod.pg@oracle.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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You will still be required to wrestle the browser's offer-answer state machi=
ne into submission to get the results you want.

Matthew Kaufman

(Sent from my iPhone)

On Jul 4, 2013, at 2:10 AM, Binod <binod.pg@oracle.com> wrote:

> On Thursday 04 July 2013 01:51 PM, I=C3=B1aki Baz Castillo wrote:
>> 2013/7/4 Eric Rescorla <ekr@rtfm.com>:
>>> Looking back at your posts from back then, I see you did in fact suggest=

>>> a set of API modifications that would have included an abstract session
>>> description. That would in fact have made Jingle somewhat (though not
>>> really that much) easier. So my last sentence here is unfair. My apologi=
es.
>> Initially I though that SessionDescription was a powerful JS Object
>> with all the fields and attributes to construct a SDP, so a developer
>> could extract those media/transport fields and create a SDP via JS. At
>> the same time, the SessionDescription object would include helper
>> methods (defined by the WebRTC API) to build a plain SDP (i.e. toSDP)
>> and some others like XML-SDP (i.e. toXMLSDP).
> I like that approach. Have an API to specify all/most of what is
> in the SDP and then use a toSDP to generate actual SDP string to be sent
> across the wire.
>=20
> Other side should be able to reconstruct the browser side
> object(s) using the received SDP string.
>=20
> thanks,
> Binod.
>>=20
>>=20
>>=20
>> --
>> I=C3=B1aki Baz Castillo
>> <ibc@aliax.net>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From fluffy@iii.ca  Thu Jul  4 17:47:32 2013
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As part of putting together some chair slides for Berlin, I'm trying to =
get a very rough idea of where things are. I'd like to get folks input =
on what percent done various drafts are where Done is a published RFC.=20=


This is nearly impossible to do at any level of accuracy but I do want =
to get a vague idea. I took a very rough stab below just to have a =
starting point to get the conversation going. I'm sure my numbers are =
mostly wrong but I have tried to ask a few people and sort of take the =
central cluster of the guess. If you think I am way out, please provide =
some feedback of what you think the number actually is.

I'm happy to get feedback on specific drafts or feedback of the form =
they are all too low or too high. I do encourage people to realistically =
look at the data tracker to see how long other drafts have taken for =
each stage to get a baseline instead of just going with their gut feel.=20=


Thanks, Cullen

PS - if you are an author and look at the number next to your draft and =
think it is way off, please please, say something.=20



draft-ietf-rtcweb-overview		80
draft-ietf-rtcweb-use-cases-and-requirements	70
http://dev.w3.org/2011/webrtc/editor/getusermedia.html	70
draft-burnett-rtcweb-constraints-registry		80
http://dev.w3.org/2011/webrtc/editor/webrtc.html	50
draft-nandakumar-rtcweb-stun-uri		80
draft-petithuguenin-behave-turn-uris		80
draft-ietf-rtcweb-security		80
draft-ietf-rtcweb-security-arch		80
draft-muthu-behave-consent-freshness-02 	50
draft-ietf-rtcweb-data-channel		70
draft-ietf-mmusic-sctp-sdp		70
draft-jesup-rtcweb-data-protocol 		50
draft-ietf-tsvwg-sctp-dtls-encaps		70
draft-ietf-rtcweb-rtp-usage		40
draft-ietf-avtcore-6222bis		80
draft-ietf-avtcore-avp-codecs		80
draft-ietf-avtcore-srtp-encrypted-header-ext	80
draft-ietf-avtext-multiple-clock-rates		70
draft-lennox-rtcweb-rtp-media-type-mux 		80
draft-ietf-avtcore-rtp-circuit-breakers		70
draft-ietf-avtcore-multi-media-rtp-session 	70
draft-lennox-avtcore-rtp-multi-stream		70
draft-ietf-rtcweb-jsep		50
draft-ietf-mmusic-msid		70
draft-rescorla-mmusic-ice-trickle		25
draft-ietf-mmusic-sdp-bundle-negotiation	50
draft-nandakumar-mmusic-sdp-mux-attributes	25
draft-dhesikan-tsvwg-rtcweb-qos		10 or 90
draft-ietf-rtcweb-audio		50
draft-ietf-payload-rtp-opus		90
draft-ietf-payload-vp8-08		95



From jonathan@vidyo.com  Fri Jul  5 09:29:49 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Cullen Jennings <fluffy@iii.ca>
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On Jul 4, 2013, at 8:47 PM, Cullen Jennings <fluffy@iii.ca> wrote:

> draft-ietf-avtcore-srtp-encrypted-header-ext	80

RFC 6904 - done.

(Your estimates for the other drafts I'm working on seem reasonable.)

--
Jonathan Lennox
jonathan@vidyo.com



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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Cullen Jennings <fluffy@iii.ca>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Question about the status of various drafts
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Subject: Re: [rtcweb] Question about the status of various drafts
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Hi Cullen,

|draft-muthu-behave-consent-freshness-02 	50

Reasonable assessment. We are working on the next revision.

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f Cullen Jennings
|Sent: Friday, July 05, 2013 6:17 AM
|To: rtcweb@ietf.org
|Subject: [rtcweb] Question about the status of various drafts
|
|
|As part of putting together some chair slides for Berlin, I'm trying to ge=
t a very rough idea of where
|things are. I'd like to get folks input on what percent done various draft=
s are where Done is a
|published RFC.
|
|This is nearly impossible to do at any level of accuracy but I do want to =
get a vague idea. I took a
|very rough stab below just to have a starting point to get the conversatio=
n going. I'm sure my numbers
|are mostly wrong but I have tried to ask a few people and sort of take the=
 central cluster of the
|guess. If you think I am way out, please provide some feedback of what you=
 think the number actually
|is.
|
|I'm happy to get feedback on specific drafts or feedback of the form they =
are all too low or too high.
|I do encourage people to realistically look at the data tracker to see how=
 long other drafts have
|taken for each stage to get a baseline instead of just going with their gu=
t feel.
|
|Thanks, Cullen
|
|PS - if you are an author and look at the number next to your draft and th=
ink it is way off, please
|please, say something.
|
|
|
|draft-ietf-rtcweb-overview		80
|draft-ietf-rtcweb-use-cases-and-requirements	70
|http://dev.w3.org/2011/webrtc/editor/getusermedia.html	70
|draft-burnett-rtcweb-constraints-registry		80
|http://dev.w3.org/2011/webrtc/editor/webrtc.html	50
|draft-nandakumar-rtcweb-stun-uri		80
|draft-petithuguenin-behave-turn-uris		80
|draft-ietf-rtcweb-security		80
|draft-ietf-rtcweb-security-arch		80
|draft-muthu-behave-consent-freshness-02 	50
|draft-ietf-rtcweb-data-channel		70
|draft-ietf-mmusic-sctp-sdp		70
|draft-jesup-rtcweb-data-protocol 		50
|draft-ietf-tsvwg-sctp-dtls-encaps		70
|draft-ietf-rtcweb-rtp-usage		40
|draft-ietf-avtcore-6222bis		80
|draft-ietf-avtcore-avp-codecs		80
|draft-ietf-avtcore-srtp-encrypted-header-ext	80
|draft-ietf-avtext-multiple-clock-rates		70
|draft-lennox-rtcweb-rtp-media-type-mux 		80
|draft-ietf-avtcore-rtp-circuit-breakers		70
|draft-ietf-avtcore-multi-media-rtp-session 	70
|draft-lennox-avtcore-rtp-multi-stream		70
|draft-ietf-rtcweb-jsep		50
|draft-ietf-mmusic-msid		70
|draft-rescorla-mmusic-ice-trickle		25
|draft-ietf-mmusic-sdp-bundle-negotiation	50
|draft-nandakumar-mmusic-sdp-mux-attributes	25
|draft-dhesikan-tsvwg-rtcweb-qos		10 or 90
|draft-ietf-rtcweb-audio		50
|draft-ietf-payload-rtp-opus		90
|draft-ietf-payload-vp8-08		95
|
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

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--_136c9421-69b9-4aa7-a3f6-ebcdda05f936_
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I think you need to take dependencies into account.   There are several rea=
sons for this:=20
   a. Until you've gone through the dependency graph for the items=2C the f=
ull list of documents that need to be tracked isn't known.    b. Estimates =
of completion are dependent on the progress of normative dependencies (e.g.=
 they can't be published until the dependencies are resolved).=20

As an example=2C  draft-ietf-rtcweb-overview (which is listed as 80 percent=
 complete) has normative references to the following drafts:=20
   [I-D.ietf-mmusic-sctp-sdp] 70   [I-D.ietf-rtcweb-audio] 50   [I-D.ietf-r=
tcweb-data-channel] 70   [I-D.ietf-rtcweb-jsep] 50   [I-D.ietf-rtcweb-rtp-u=
sage] 40   [I-D.ietf-rtcweb-security] 80   [I-D.ietf-rtcweb-security-arch] =
80   [I-D.nandakumar-rtcweb-stun-uri] 80   [I-D.tuexen-tsvwg-sctp-dtls-enca=
ps] 70            Since draft-ietf-rtcweb-overview is dependent on a docume=
nt (rtp usage) which only has an estimate of 40 percent done) I'd say that =
the estimate is probably closer to 40 than 80.=20

> From: fluffy@iii.ca
> Date: Thu=2C 4 Jul 2013 17:47:20 -0700
> To: rtcweb@ietf.org
> Subject: [rtcweb] Question about the status of various drafts
>=20
>=20
> As part of putting together some chair slides for Berlin=2C I'm trying to=
 get a very rough idea of where things are. I'd like to get folks input on =
what percent done various drafts are where Done is a published RFC.=20
>=20
> This is nearly impossible to do at any level of accuracy but I do want to=
 get a vague idea. I took a very rough stab below just to have a starting p=
oint to get the conversation going. I'm sure my numbers are mostly wrong bu=
t I have tried to ask a few people and sort of take the central cluster of =
the guess. If you think I am way out=2C please provide some feedback of wha=
t you think the number actually is.
>=20
> I'm happy to get feedback on specific drafts or feedback of the form they=
 are all too low or too high. I do encourage people to realistically look a=
t the data tracker to see how long other drafts have taken for each stage t=
o get a baseline instead of just going with their gut feel.=20
>=20
> Thanks=2C Cullen
>=20
> PS - if you are an author and look at the number next to your draft and t=
hink it is way off=2C please please=2C say something.=20
>=20
>=20
>=20
> draft-ietf-rtcweb-overview		80
> draft-ietf-rtcweb-use-cases-and-requirements	70
> http://dev.w3.org/2011/webrtc/editor/getusermedia.html	70
> draft-burnett-rtcweb-constraints-registry		80
> http://dev.w3.org/2011/webrtc/editor/webrtc.html	50
> draft-nandakumar-rtcweb-stun-uri		80
> draft-petithuguenin-behave-turn-uris		80
> draft-ietf-rtcweb-security		80
> draft-ietf-rtcweb-security-arch		80
> draft-muthu-behave-consent-freshness-02 	50
> draft-ietf-rtcweb-data-channel		70
> draft-ietf-mmusic-sctp-sdp		70
> draft-jesup-rtcweb-data-protocol 		50
> draft-ietf-tsvwg-sctp-dtls-encaps		70
> draft-ietf-rtcweb-rtp-usage		40
> draft-ietf-avtcore-6222bis		80
> draft-ietf-avtcore-avp-codecs		80
> draft-ietf-avtcore-srtp-encrypted-header-ext	80
> draft-ietf-avtext-multiple-clock-rates		70
> draft-lennox-rtcweb-rtp-media-type-mux 		80
> draft-ietf-avtcore-rtp-circuit-breakers		70
> draft-ietf-avtcore-multi-media-rtp-session 	70
> draft-lennox-avtcore-rtp-multi-stream		70
> draft-ietf-rtcweb-jsep		50
> draft-ietf-mmusic-msid		70
> draft-rescorla-mmusic-ice-trickle		25
> draft-ietf-mmusic-sdp-bundle-negotiation	50
> draft-nandakumar-mmusic-sdp-mux-attributes	25
> draft-dhesikan-tsvwg-rtcweb-qos		10 or 90
> draft-ietf-rtcweb-audio		50
> draft-ietf-payload-rtp-opus		90
> draft-ietf-payload-vp8-08		95
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

--_136c9421-69b9-4aa7-a3f6-ebcdda05f936_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>I think you need to take depende=
ncies into account. &nbsp=3B There are several reasons for this:&nbsp=3B<di=
v><br><div>&nbsp=3B &nbsp=3Ba. Until you've gone through the dependency gra=
ph for the items=2C the full list of documents that need to be tracked isn'=
t known.&nbsp=3B</div><div>&nbsp=3B &nbsp=3Bb. Estimates of completion are =
dependent on the progress of normative dependencies (e.g. they can't be pub=
lished until the dependencies are resolved).&nbsp=3B<br><div><br></div><div=
>As an example=2C &nbsp=3Bdraft-ietf-rtcweb-overview (which is listed as 80=
 percent complete) has normative references to the following drafts:&nbsp=
=3B<div><div><br></div><div>&nbsp=3B &nbsp=3B[I-D.ietf-mmusic-sctp-sdp] 70<=
/div><div>&nbsp=3B &nbsp=3B[I-D.ietf-rtcweb-audio] 50</div><div>&nbsp=3B &n=
bsp=3B[I-D.ietf-rtcweb-data-channel] 70</div><div>&nbsp=3B &nbsp=3B[I-D.iet=
f-rtcweb-jsep] 50</div><div>&nbsp=3B &nbsp=3B[I-D.ietf-rtcweb-rtp-usage] 40=
</div><div>&nbsp=3B &nbsp=3B[I-D.ietf-rtcweb-security] 80</div><div>&nbsp=
=3B &nbsp=3B[I-D.ietf-rtcweb-security-arch] 80</div><div>&nbsp=3B &nbsp=3B[=
I-D.nandakumar-rtcweb-stun-uri] 80</div><div>&nbsp=3B &nbsp=3B[I-D.tuexen-t=
svwg-sctp-dtls-encaps] 70</div><div>&nbsp=3B &nbsp=3B &nbsp=3B &nbsp=3B &nb=
sp=3B &nbsp=3B&nbsp=3B</div><div>Since draft-ietf-rtcweb-overview is depend=
ent on a document (rtp usage) which only has an estimate of 40 percent done=
) I'd say that the estimate is probably closer to 40 than 80.&nbsp=3B</div>=
<div><br></div><div><br><div>&gt=3B From: fluffy@iii.ca<br>&gt=3B Date: Thu=
=2C 4 Jul 2013 17:47:20 -0700<br>&gt=3B To: rtcweb@ietf.org<br>&gt=3B Subje=
ct: [rtcweb] Question about the status of various drafts<br>&gt=3B <br>&gt=
=3B <br>&gt=3B As part of putting together some chair slides for Berlin=2C =
I'm trying to get a very rough idea of where things are. I'd like to get fo=
lks input on what percent done various drafts are where Done is a published=
 RFC. <br>&gt=3B <br>&gt=3B This is nearly impossible to do at any level of=
 accuracy but I do want to get a vague idea. I took a very rough stab below=
 just to have a starting point to get the conversation going. I'm sure my n=
umbers are mostly wrong but I have tried to ask a few people and sort of ta=
ke the central cluster of the guess. If you think I am way out=2C please pr=
ovide some feedback of what you think the number actually is.<br>&gt=3B <br=
>&gt=3B I'm happy to get feedback on specific drafts or feedback of the for=
m they are all too low or too high. I do encourage people to realistically =
look at the data tracker to see how long other drafts have taken for each s=
tage to get a baseline instead of just going with their gut feel. <br>&gt=
=3B <br>&gt=3B Thanks=2C Cullen<br>&gt=3B <br>&gt=3B PS - if you are an aut=
hor and look at the number next to your draft and think it is way off=2C pl=
ease please=2C say something. <br>&gt=3B <br>&gt=3B <br>&gt=3B <br>&gt=3B d=
raft-ietf-rtcweb-overview		80<br>&gt=3B draft-ietf-rtcweb-use-cases-and-req=
uirements	70<br>&gt=3B http://dev.w3.org/2011/webrtc/editor/getusermedia.ht=
ml	70<br>&gt=3B draft-burnett-rtcweb-constraints-registry		80<br>&gt=3B htt=
p://dev.w3.org/2011/webrtc/editor/webrtc.html	50<br>&gt=3B draft-nandakumar=
-rtcweb-stun-uri		80<br>&gt=3B draft-petithuguenin-behave-turn-uris		80<br>=
&gt=3B draft-ietf-rtcweb-security		80<br>&gt=3B draft-ietf-rtcweb-security-=
arch		80<br>&gt=3B draft-muthu-behave-consent-freshness-02 	50<br>&gt=3B dr=
aft-ietf-rtcweb-data-channel		70<br>&gt=3B draft-ietf-mmusic-sctp-sdp		70<b=
r>&gt=3B draft-jesup-rtcweb-data-protocol 		50<br>&gt=3B draft-ietf-tsvwg-s=
ctp-dtls-encaps		70<br>&gt=3B draft-ietf-rtcweb-rtp-usage		40<br>&gt=3B dra=
ft-ietf-avtcore-6222bis		80<br>&gt=3B draft-ietf-avtcore-avp-codecs		80<br>=
&gt=3B draft-ietf-avtcore-srtp-encrypted-header-ext	80<br>&gt=3B draft-ietf=
-avtext-multiple-clock-rates		70<br>&gt=3B draft-lennox-rtcweb-rtp-media-ty=
pe-mux 		80<br>&gt=3B draft-ietf-avtcore-rtp-circuit-breakers		70<br>&gt=3B=
 draft-ietf-avtcore-multi-media-rtp-session 	70<br>&gt=3B draft-lennox-avtc=
ore-rtp-multi-stream		70<br>&gt=3B draft-ietf-rtcweb-jsep		50<br>&gt=3B dra=
ft-ietf-mmusic-msid		70<br>&gt=3B draft-rescorla-mmusic-ice-trickle		25<br>=
&gt=3B draft-ietf-mmusic-sdp-bundle-negotiation	50<br>&gt=3B draft-nandakum=
ar-mmusic-sdp-mux-attributes	25<br>&gt=3B draft-dhesikan-tsvwg-rtcweb-qos		=
10 or 90<br>&gt=3B draft-ietf-rtcweb-audio		50<br>&gt=3B draft-ietf-payload=
-rtp-opus		90<br>&gt=3B draft-ietf-payload-vp8-08		95<br>&gt=3B <br>&gt=3B =
<br>&gt=3B _______________________________________________<br>&gt=3B rtcweb=
 mailing list<br>&gt=3B rtcweb@ietf.org<br>&gt=3B https://www.ietf.org/mail=
man/listinfo/rtcweb<br></div></div></div></div></div></div> 		 	   		  </di=
v></body>
</html>=

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/3/13 11:37 PM, Eric Rescorla wrote:=0A=
> On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <pthatcher@google.com=0A=
> <mailto:pthatcher@google.com>> wrote:=0A=
>=0A=
>     On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <ekr@rtfm.com=0A=
>     <mailto:ekr@rtfm.com>> wrote:=0A=
>=0A=
>         On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher=0A=
>         <pthatcher@google.com <mailto:pthatcher@google.com>> wrote:=0A=
>=0A=
>             Camp #0:  I've used SDP for years and I'm very comfortable=0A=
>             with it.  Using SDP as the control surface really helps my=0A=
>             use case, which is legacy interop.  Defining an API without=
=0A=
>             SDP would be too much work, and probably fail.  Look at what=
=0A=
>             happened with SDPng!  Supporting all these advanced use=0A=
>             cases doesn't seem worth it.   If developers are doing that=
=0A=
>             much advanced stuff, they can learn to munge SDP.  It isn't=
=0A=
>             that bad.=0A=
>=0A=
>=0A=
>         Hmm... That's not my understanding of the situation at all.=0A=
>=0A=
>         Rather, I believe the expectation is that you shouldn't have to=
=0A=
>         modify the=0A=
>         SDP but rather there should be API points to cover most of the=0A=
>         use cases=0A=
>         that people want. This isn't to say that all those API points=0A=
>         exist or that=0A=
>         they work or whatever.=0A=
>=0A=
>         -Ekr=0A=
>=0A=
>=0A=
>     I'm glad to be wrong here.  Is the phrase "you shouldn't have to=0A=
>     modify the SDP but rather there should be API points to cover most=0A=
>     of the use cases"  the consensus of the working group?=0A=
>=0A=
>=0A=
> I thought it was, but I'm not the chair, so maybe you could ask Harald or=
=0A=
> Stefan.=0A=
=0A=
I definitely think this is the consensus of the working group.=0A=
=0A=
I think the exception would be when interoperating with other systems =0A=
that use another dialect of SDP - I don't think we could cover such =0A=
translations with API points.=0A=
=0A=
=0A=
>=0A=
>=0A=
>     Along with that, is "use Jingle for signalling" included in your set=
=0A=
>     of "most of the use cases"?=0A=
>=0A=
>=0A=
> No, I don't think it is, since it's not SDP.=0A=
>=0A=
> Maybe I could phrase this differently: It was my understanding that you=
=0A=
> should have=0A=
> API points to get the PC to emit SDP that would express the policies,=0A=
> preferences,=0A=
> actions, etc. that the application wants. If you want something that's=0A=
> not SDP you would=0A=
> need to gateway.=0A=
>=0A=
> This may or may not be a satisfactory state of affairs, but it was the=0A=
> rule I was using=0A=
> (based on what I thought the WG expected) for when use cases needed new A=
PI=0A=
> points.=0A=
>=0A=
> -Ekr=0A=
>=0A=
=0A=

From silviapfeiffer1@gmail.com  Sat Jul  6 04:00:10 2013
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Sat, 6 Jul 2013 20:59:49 +1000
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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An interesting read. It makes me wonder if there is a full description
available of the SDP that browsers (Chrome, Firefox) currently
support.

Looking forward to your proposal (I assume it will be on the W3C list?).

Cheers,
Silvia.

On Wed, Jun 26, 2013 at 2:37 PM, Robin Raymond <robin@hookflash.com> wrote:
>
> According to many, real adoption of WebRTC will not happen if we continue to
> force everyone to use this SDP Offer/Answer methodology.  It is clearly
> blocking our way forward and the amount of specification documentation
> remaining needed for the browser vendors to produce a compatible SDP based
> WebRTC engine in a browser is much more daunting than most are willing to
> admit.
>
>
> The draft rationale behind a JavaScript Object API model:
>
> http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00.txt
>
> Whereas:
>
> The WebRTC JavaScript Object API & Shim document will be along shortly.
>
>
> We wanted to get this initial rationale draft informational document out
> right away. Everyone that has any interest should review the reasons and
> concepts and comment before we get too far along on the details of an actual
> API, the initial API will follow in the coming days.
>
>
> Best regards,
>
> Robin Raymond
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From hadriel.kaplan@oracle.com  Sat Jul  6 07:58:44 2013
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Well in our defense, #9 wasn't our argument - that was the argument the =
browser makers themselves raised two years ago on the deciding =
conference call for why the browsers should be doing SDP rather than a =
real API.  In fact if I recall right, I think that was the main deciding =
argument for why browsers shouldn't offer a lower-level API.

-hadriel


On Jul 3, 2013, at 6:20 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/3 Hadriel Kaplan <hadriel.kaplan@oracle.com>:
>> You might also want to cite some of the reasons noted in this old =
draft:
>> http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01
>=20
> Well, section "3. Defining a WebRTC Protocol in the Browser" says:
>=20
>=20
>   9) Using the SDP offer/answer model provides a more rigid API
>     interaction model, enabling Browser vendors to perform less
>     testing and provide more robust implementations than exposing all
>     discrete components to a Javascript API would.
>=20
>   10)   Using a higher-level API model, such as would be done with an
>     SDP offer/answer model, means the cross-browser vendor-specific
>     variances would be reduced.  Exposing a lower-level API would
>     inevitably lead to some differences in different browsers due to
>     differences in their architectures/implementation.
>=20
>=20
> Time to reconsider those assertions? ;)
>=20
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>


From roman@telurix.com  Sun Jul  7 14:15:03 2013
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On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 7/3/13 11:37 PM, Eric Rescorla wrote:
> >     I'm glad to be wrong here.  Is the phrase "you shouldn't have to
> >     modify the SDP but rather there should be API points to cover most
> >     of the use cases"  the consensus of the working group?
> >
> >
> > I thought it was, but I'm not the chair, so maybe you could ask Harald =
or
> > Stefan.
>
> I definitely think this is the consensus of the working group.
>
>
Can you point out when exactly this consensus was reached? This is news to
me, but I definitely could have missed something.
_____________
Roman Shpount

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<br clear=3D"all"><div>On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson L=
K <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com"=
 target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:</=
div><div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 7/3/13 11:37 PM, Eric Rescorla wrote:<br>=
&gt; =A0 =A0 I&#39;m glad to be wrong here. =A0Is the phrase &quot;you shou=
ldn&#39;t have to<br>

&gt; =A0 =A0 modify the SDP but rather there should be API points to cover =
most<br>
&gt; =A0 =A0 of the use cases&quot; =A0the consensus of the working group?<=
br>
&gt;<br>
&gt;<br>
&gt; I thought it was, but I&#39;m not the chair, so maybe you could ask Ha=
rald or<br>
&gt; Stefan.<br>
<br>
I definitely think this is the consensus of the working group.<br>
<br></blockquote><div><br></div><div>Can you point out when exactly this co=
nsensus was reached? This is news to me, but I definitely could have missed=
 something.</div><div>_____________<br>Roman Shpount</div><br><div>=A0</div=
>
</div>

--f46d0442808e438c5404e0f26b2a--

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Jul 3, 2013, at 2:42 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> So compatibility with SIP is important but compatibility with Jingle =
is just impossible.

The mapping of SDP to jingle is in the Jingle specs =85 I'm not express =
any opinion on this one way or another other but the authors of theses =
specs have always claimed Jingle fully mapped to and from SDP.



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From: Justin Uberti <juberti@google.com>
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--f46d04427270fc208604e0f86dbc
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Just uploaded a 00 version of a spec for requesting time-limited TURN
credentials for WebRTC apps. Would like to get 10 minutes of agenda time to
present this in Berlin.

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Jul 8, 2013 at 12:15 AM
Subject: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
To: Justin Uberti <justin@uberti.name>



A new version of I-D, draft-uberti-rtcweb-turn-rest-00.txt
has been successfully submitted by Justin Uberti and posted to the
IETF repository.

Filename:        draft-uberti-rtcweb-turn-rest
Revision:        00
Title:           A REST API For Access To TURN Services
Creation date:   2013-07-08
Group:           Individual Submission
Number of pages: 7
URL:
http://www.ietf.org/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt
Status:
http://datatracker.ietf.org/doc/draft-uberti-rtcweb-turn-rest
Htmlized:        http://tools.ietf.org/html/draft-uberti-rtcweb-turn-rest-00


Abstract:
   This document describes a proposed standard REST API for obtaining
   access to TURN services via ephemeral (i.e. time-limited)
   credentials.  These credentials are vended by a web service over
   HTTP, and then supplied to and checked by a TURN server using the
   standard TURN protocol.  The usage of ephemeral credentials ensures
   that access to the TURN server can be controlled even if the
   credentials can be discovered by the user, as is the case in WebRTC
   where TURN credentials must be specified in Javascript.




The IETF Secretariat

--f46d04427270fc208604e0f86dbc
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<div dir=3D"ltr">Just uploaded a 00 version of a spec for requesting time-l=
imited TURN credentials for WebRTC apps. Would like to get 10 minutes of ag=
enda time to present this in Berlin.<br><div class=3D"gmail_quote"><div dir=
=3D"ltr">

<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</=
a>&gt;</span><br>



Date: Mon, Jul 8, 2013 at 12:15 AM<br>Subject: New Version Notification for=
 draft-uberti-rtcweb-turn-rest-00.txt<br>To: Justin Uberti &lt;<a href=3D"m=
ailto:justin@uberti.name" target=3D"_blank">justin@uberti.name</a>&gt;<br>

<br><br><br>
A new version of I-D, draft-uberti-rtcweb-turn-rest-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Filename: =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-uberti-rtcweb-turn-rest<br>
Revision: =C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 A REST API For Access To TURN Ser=
vices<br>
Creation date: =C2=A0 2013-07-08<br>
Group: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Number of pages: 7<br>
URL: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.o=
rg/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt" target=3D"_blank">=
http://www.ietf.org/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt</a=
><br>
Status: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://datatracker.iet=
f.org/doc/draft-uberti-rtcweb-turn-rest" target=3D"_blank">http://datatrack=
er.ietf.org/doc/draft-uberti-rtcweb-turn-rest</a><br>
Htmlized: =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/=
draft-uberti-rtcweb-turn-rest-00" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-uberti-rtcweb-turn-rest-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes a proposed standard REST API for obtai=
ning<br>
=C2=A0 =C2=A0access to TURN services via ephemeral (i.e. time-limited)<br>
=C2=A0 =C2=A0credentials. =C2=A0These credentials are vended by a web servi=
ce over<br>
=C2=A0 =C2=A0HTTP, and then supplied to and checked by a TURN server using =
the<br>
=C2=A0 =C2=A0standard TURN protocol. =C2=A0The usage of ephemeral credentia=
ls ensures<br>
=C2=A0 =C2=A0that access to the TURN server can be controlled even if the<b=
r>
=C2=A0 =C2=A0credentials can be discovered by the user, as is the case in W=
ebRTC<br>
=C2=A0 =C2=A0where TURN credentials must be specified in Javascript.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>
</div><br></div>

--f46d04427270fc208604e0f86dbc--

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Fwd: New Version Notification for	draft-uberti-rtcweb-turn-rest-00.txt
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 8 Jul 2013 12:38:41 +0200
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/8 Cullen Jennings <fluffy@iii.ca>:
>> So compatibility with SIP is important but compatibility with Jingle is =
just impossible.
>
> The mapping of SDP to jingle is in the Jingle specs =E2=80=A6 I'm not exp=
ress any opinion on this one way or another other but the authors of theses=
 specs have always claimed Jingle fully mapped to and from SDP.

XEP-0167 defines a XML based SDP. Mapping it from/to plain SDP
requires, at least, having all the attributes of the SDP. Instead you
are proposing that the JS receives an opaque SDP blob from the
browser, so it must be parsed via JS if the app wants to convert it
into XML SDP.

It's really ugly and the worst specification for the Web.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/7/13 11:14 PM, Roman Shpount wrote:=0A=
>=0A=
> On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     On 7/3/13 11:37 PM, Eric Rescorla wrote:=0A=
>      >     I'm glad to be wrong here.  Is the phrase "you shouldn't have =
to=0A=
>      >     modify the SDP but rather there should be API points to cover=
=0A=
>     most=0A=
>      >     of the use cases"  the consensus of the working group?=0A=
>      >=0A=
>      >=0A=
>      > I thought it was, but I'm not the chair, so maybe you could ask=0A=
>     Harald or=0A=
>      > Stefan.=0A=
>=0A=
>     I definitely think this is the consensus of the working group.=0A=
>=0A=
>=0A=
> Can you point out when exactly this consensus was reached? This is news=
=0A=
> to me, but I definitely could have missed something.=0A=
=0A=
No I can't really. To be clear, no consensus call (or similar) has been =0A=
held. But every time this is discussed, what I hear is people agreeing =0A=
and no objections. It was discussed a bit, e.g., at the f2f at TPAC last =
=0A=
year, if you read the minutes [1] you will find some traces of that =0A=
discussion.=0A=
=0A=
And, I don't understand why there would be a debate about this. There =0A=
are obviously many who want to define APIs (or constraints) that allow =0A=
most use-cases to be met without having to modify the SDP. Why would =0A=
anyone have anything against that? If there is anyone who really wants =0A=
to modify the SDP instead (I have no clue why, but anyway) they can =0A=
still do that.=0A=
=0A=
[1] http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.html=0A=
=0A=
> _____________=0A=
> Roman Shpount=0A=
>=0A=
=0A=

From simon.perreault@viagenie.ca  Mon Jul  8 05:23:41 2013
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
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Le 2013-07-08 06:24, Justin Uberti a écrit :
> Just uploaded a 00 version of a spec for requesting time-limited TURN
> credentials for WebRTC apps. Would like to get 10 minutes of agenda time
> to present this in Berlin.

Cool.

One small comment: shouldn't the response also contain the value of the
REALM parameter?

Thanks,
Simon

From pthatcher@google.com  Mon Jul  8 06:57:34 2013
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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--001a11c39d18cd80e304e1006c57
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Just in case it is missed elsewhere in the thread, I'll reply directly to
my own message.  It was pointed out to me that I made a mistake with my
addition.  The first line should have read:

61% "Good for simple Stuff"        39% "Bad for simple stuff"

Sorry about that.


On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <pthatcher@google.com> wrote:

> After about a week of input, and feedback from 19 developers doing a wide
> range of things using the WebRTC API, we have some data about their opinion
> of the current API.
>
> *Short answer*:
>
> 61% "Good for simple Stuff"    49% "Bad for simple stuff"
>  6% "Good for advanced stuff"      94% "Bad for advanced stuff"
> 46% "Good/OK/tolerable for 1.0"    54% "Horrible/intolerable for 1.0"
>  7% "Good/OK/tolerable long-term"  93% "Need a better API long-term"
>
>
> In general the attitude is *"It's OK for simple stuff but bad for
> advanced stuff",  *and *"We can tolerate it for 1.0, but we need
> something better for 2.0".  *But the majority of the feedback was against
> it even for the 1.0, so the last one is a bit of a stretch.
>
>
> *Long Answer*:
>
> I had to interpret the results a little to fit into these buckets, so if
> you don't like the interpretation, you're more than welcome to interpret
> your own results from the raw data:
> https://docs.google.com/spreadsheet/ccc?key=0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=1
>
>
> There seem to be 4 or 5 different camps:
>
> Camp #1:  Doing very simple things, doesn't touch the SDP.  Everything is
> fine, except they might have things they want to control but can't.
>
> Camp #2:  Does somewhat more advanced things, touches the SDP a little,
> has pain, anticipates more pain, would like something better, but thinks
> this is OK for 1.0.
>
> Camp #3:  Is trying advanced things, is touching the SDP a lot, and really
> feels pain.  Wants something better, and might tolerate this API for 1.0,
> but not for long.
>
> Camp #4:  Doesn't want SDP or O/A in the API at all.  Has a very strong
> dislike for it.  Won't tolerate it, even for 1.0.  The IETF and W3C should
> be embarrassed for even suggesting such a terrible API.  It's better to go
> back to the drawing board (if you think those words are strong, go read the
> spreadsheet or mailing list!)
>
> Oh, and of course, for completeness, we should describe Camp #0, which
> didn't have any input in this feedback:
>
> Camp #0:  I've used SDP for years and I'm very comfortable with it.  Using
> SDP as the control surface really helps my use case, which is legacy
> interop.  Defining an API without SDP would be too much work, and probably
> fail.  Look at what happened with SDPng!  Supporting all these advanced use
> cases doesn't seem worth it.   If developers are doing that much advanced
> stuff, they can learn to munge SDP.  It isn't that bad.
>
> (By the way, I'm not a member of Camp #0, so I may be a little off in my
> description of it.  If any member of Camp #0 would like to adjust that
> description, please feel free).
>
>
>
>
> *Suggested Next Step: Figure out what is needed for WebRTC++*
> *
> *
> As Cullen recently pointed out, we need to get a list of things developers
> want to be able to control in a future version of the API which could
> hopefully make developers happier Whether that future API is 1.0, 2.0, or
> 1.1 is an orthogonal question, but let's call it WebRTC++ for now.
> Clearly, they want a way to control things without SDP, and perhaps without
> O/A at all.  But what do they want to control?  We need a good list.
>
> I suggest we reach out more closely to the developers, and use their input
> to create not only a list of things they want to do, but a WebRTC++ API
> that fulfills those needs without requiring SDP munging.  I think we can do
> this in parallel with the current  API work without detracting from it
> (from the current work, that is).    Certainly there is plenty of energy
> from people interested in working on it (WebRTC++, that is).  Such parallel
> work would not only better capture the needs of developers, but could also
> improve the WebRTC (not ++) API  by incorporating ideas from it.
>
> So far, such activity has been frowned on, perhaps out of fear that it's a
> threat to the current work.  But I think it can be done in a way that
> doesn't threaten anything, and even learns from and builds on the valuable
> input we are receiving from developers.  The alternative is to ask all
> these developers to wait an indefinite amount of time and munge SDP like
> crazy until then, and that seems like a rather poor answer.  Therefore, I
> would support an exploratory effort at a "WebRTC++ API".
>
>
>
>
>
>
>
>

--001a11c39d18cd80e304e1006c57
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Just in case it is missed elsewhere in the thread, I&#39;l=
l reply directly to my own message. =C2=A0It was pointed out to me that I m=
ade a mistake with my addition. =C2=A0The first line should have read:<div>=
<br></div>

<div><div>61% &quot;Good for simple Stuff&quot; =C2=A0 =C2=A0 =C2=A0 =C2=A0=
39% &quot;Bad for simple stuff&quot;<br></div></div><div><br></div><div>Sor=
ry about that. =C2=A0</div></div><div class=3D"gmail_extra"><br><br><div cl=
ass=3D"gmail_quote">On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher <span di=
r=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pth=
atcher@google.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">After about a week of input=
, and feedback from 19 developers doing a wide range of things using the We=
bRTC API, we have some data about their opinion of the current API.<div>

<br></div><div><b>Short answer</b>:</div>
<div><br></div><div><div><span style=3D"font-family:&#39;courier new&#39;,m=
onospace">61%=C2=A0</span><font face=3D"courier new, monospace">&quot;Good =
for simple Stuff&quot;<span style=3D"white-space:pre-wrap">=C2=A0     </spa=
n>=C2=A0 49% &quot;Bad for simple stuff&quot;</font></div>


<div><span style=3D"font-family:&#39;courier new&#39;,monospace">=C2=A06%=
=C2=A0</span><font face=3D"courier new, monospace">&quot;Good for advanced =
stuff&quot; =C2=A0 =C2=A0 =C2=A094% &quot;Bad for advanced stuff&quot;</fon=
t></div><div><span style=3D"font-family:&#39;courier new&#39;,monospace">46=
%=C2=A0</span><font face=3D"courier new, monospace">&quot;Good/OK/tolerable=
 for 1.0&quot; =C2=A0 =C2=A054% &quot;Horrible/intolerable for 1.0&quot;</f=
ont></div>


<div><span style=3D"font-family:&#39;courier new&#39;,monospace">=C2=A07%=
=C2=A0</span><font face=3D"courier new, monospace">&quot;Good/OK/tolerable =
long-term&quot; =C2=A093</font><span style=3D"font-family:&#39;courier new&=
#39;,monospace">% &quot;Need a better API long-term&quot;</span><span style=
=3D"font-family:&#39;courier new&#39;,monospace">=C2=A0</span></div>


</div><div><br></div><div><br></div><div>In general the attitude is <b>&quo=
t;It&#39;s OK for simple stuff but bad for advanced stuff&quot;, =C2=A0</b>=
and <b>&quot;We can tolerate it for 1.0, but we need something better for 2=
.0&quot;. =C2=A0</b>But the majority of the feedback was against it even fo=
r the 1.0, so the last one is a bit of a stretch.</div>


<div><br></div><div><br></div><div><b>Long Answer</b>:</div><div><br></div>=
<div><div>I had to interpret the results a little to fit into these buckets=
, so if you don&#39;t like the interpretation, you&#39;re more than welcome=
 to interpret your own results from the raw data: =C2=A0<a href=3D"https://=
docs.google.com/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJL=
RkVPV0E#gid=3D1" target=3D"_blank">https://docs.google.com/spreadsheet/ccc?=
key=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1</a></div>


</div><div><br></div><div><br></div><div>There seem to be 4 or 5 different =
camps:</div><div><br></div><div>Camp #1: =C2=A0Doing very simple things, do=
esn&#39;t touch the SDP. =C2=A0Everything is fine, except they might have t=
hings they want to control but can&#39;t.</div>


<div><br></div><div>Camp #2: =C2=A0Does somewhat more advanced things, touc=
hes the SDP a little, has pain, anticipates more pain, would like something=
 better, but thinks this is OK for 1.0.</div><div><br></div><div>Camp #3: =
=C2=A0Is trying advanced things, is touching the SDP a lot, and really feel=
s pain. =C2=A0Wants something better, and might tolerate this API for 1.0, =
but not for long.</div>


<div><br></div><div>Camp #4: =C2=A0Doesn&#39;t want SDP or O/A in the API a=
t all. =C2=A0Has a very strong dislike for it. =C2=A0Won&#39;t tolerate it,=
 even for 1.0. =C2=A0The IETF and W3C should be embarrassed for even sugges=
ting such a terrible API. =C2=A0It&#39;s better to go back to the drawing b=
oard (if you think those words are strong, go read the spreadsheet or maili=
ng list!)</div>


<div><br></div><div>Oh, and of course, for completeness, we should describe=
 Camp #0, which didn&#39;t have any input in this feedback:</div><div><br><=
/div><div>Camp #0: =C2=A0I&#39;ve used SDP for years and I&#39;m very comfo=
rtable with it. =C2=A0Using SDP as the control surface really helps my use =
case, which is legacy interop. =C2=A0Defining an API without SDP would be t=
oo much work, and probably fail. =C2=A0Look at what happened with SDPng! =
=C2=A0Supporting all these advanced use cases doesn&#39;t seem worth it. =
=C2=A0 If developers are doing that much advanced stuff, they can learn to =
munge SDP. =C2=A0It isn&#39;t that bad.=C2=A0</div>


<div><br></div><div>(By the way, I&#39;m not a member of Camp #0, so I may =
be a little off in my description of it. =C2=A0If any member of Camp #0 wou=
ld like to adjust that description, please feel free).</div><div><br></div>

<div>
<br></div><div><br></div><div><br></div><div><b>Suggested Next Step: Figure=
 out what is needed for WebRTC++</b></div><div><b><br></b></div><div>As Cul=
len recently pointed out, we need to get a list of things developers want t=
o be able to control in a future version of the API which could hopefully m=
ake developers happier Whether that future API is 1.0, 2.0, or 1.1 is an or=
thogonal question, but let&#39;s call it WebRTC++ for now. =C2=A0 Clearly, =
they want a way to control things without SDP, and perhaps without O/A at a=
ll. =C2=A0But what do they want to control? =C2=A0We need a good list.</div=
>


<div><br></div><div>I suggest we reach out more closely to the developers, =
and use their input to create not only a list of things they want to do, bu=
t a WebRTC++ API that fulfills those needs without requiring SDP munging. =
=C2=A0I think we can do this in parallel with the current =C2=A0API work wi=
thout detracting from it (from the current work, that is). =C2=A0 =C2=A0Cer=
tainly there is plenty of energy from people interested in working on it (W=
ebRTC++, that is). =C2=A0Such parallel work would not only better capture t=
he needs of developers, but could also improve the WebRTC (not ++) API =C2=
=A0by incorporating ideas from it. =C2=A0</div>


<div><br></div><div>So far, such activity has been frowned on, perhaps out =
of fear that it&#39;s a threat to the current work. =C2=A0But I think it ca=
n be done in a way that doesn&#39;t threaten anything, and even learns from=
 and builds on the valuable input we are receiving from developers. =C2=A0T=
he alternative is to ask all these developers to wait an indefinite amount =
of time and munge SDP like crazy until then, and that seems like a rather p=
oor answer. =C2=A0Therefore, I would support an exploratory effort at a &qu=
ot;WebRTC++ API&quot;.</div>


<div><br></div><div><br></div><div><br></div><div><br></div><div><br></div>=
<div><br></div><div>=C2=A0=C2=A0</div></div>
</blockquote></div><br></div>

--001a11c39d18cd80e304e1006c57--

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 5:11 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 7/7/13 11:14 PM, Roman Shpount wrote:
> >
> > On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=C3=A5kansson LK
> > <stefan.lk.hakansson@ericsson.com
> > <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
> >
> >     On 7/3/13 11:37 PM, Eric Rescorla wrote:
> >      >     I'm glad to be wrong here.  Is the phrase "you shouldn't hav=
e
> to
> >      >     modify the SDP but rather there should be API points to cove=
r
> >     most
> >      >     of the use cases"  the consensus of the working group?
> >      >
> >      >
> >      > I thought it was, but I'm not the chair, so maybe you could ask
> >     Harald or
> >      > Stefan.
> >
> >     I definitely think this is the consensus of the working group.
> >
> >
> > Can you point out when exactly this consensus was reached? This is news
> > to me, but I definitely could have missed something.
>
> No I can't really. To be clear, no consensus call (or similar) has been
> held. But every time this is discussed, what I hear is people agreeing
> and no objections.  It was discussed a bit, e.g., at the f2f at TPAC last

year, if you read the minutes [1] you will find some traces of that
> discussion.
>

No objections at all?  That is rather surprising to me.  I asked for some
feedback the API in the spreadsheet from web developers, at there many
objections from them about this part of the API.  A majority did say it was
good enough for simple use cases, but he objections were still clearly
non-zero.

I think perhaps the web developers are not well represented at TPAC.  But I
could be wrong.  Perhaps they are but changed their mind.  Or maybe they're
objecting on the IETF mailing list when they should object on the IETF
mailing list.   Or maybe the spreadsheet has a sample bias, while TPAC is
the true sample.   I don't know, but it seems like something is going on,
and I'd like to know what it is.

Stefan, what do you think?  What could explain the apparent contradiction
between objections on the mailing list vs. no objections f2f?


Also, on discussions about the API, should those who want to propose
additions (such as the NoPlan JS API I proposed) and other comments put
them on the W3C mailing list instead of the IETF one?  Is everyone barking
up the wrong tree?

Thanks in advance for your input.





> And, I don't understand why there would be a debate about this. There
> are obviously many who want to define APIs (or constraints) that allow
> most use-cases to be met without having to modify the SDP.  Why would

anyone have anything against that?


I think your question actually has two questions:

1.  Is anyone against making additions to the API to avoid SDP munging?

2.  Why are they against additions to the API to avoid SDP munging?


My personal experience for #1 is that there are some notable people in the
WG who are very much against making additions to the API to avoid SDP
munging.  Some of them are rather upset with me for proposing a few minor
additions to the API (the NoPlan JS API) precisely because it allows the JS
to go around the SDP in many cases, and they don't like that.  So, from my
experience, it's clear that the answer to #1 is "yes, many do".

And since many object to such additions, and others want such additions,
you end up with debate.


How about #2?  Why do they object?  My experience is that they view any
additions as a threat to API as a threat to the current API, which means
it's a threat to the current WG progress, which means it's a threat to
getting done.  They are very sensitive to such threats, and so object to
any additions or changes that would allow JS to avoid using SDP.  Also, I
think they don't want to ways of doing things, so SDP must remain the one
true way.


But that's just my personal experience and perspective.  I could just be
seeing it wrong, or maybe the only person experiencing this.






> I If there is anyone who really wants
> to modify the SDP instead (I have no clue why, but anyway) they can
> still do that.
>

Well, no one is going to hit themselves in the head with a hammer if they
don't have to, but there are many uses cases that currently be done only by
SDP munging, and I don't see any API additions on the horizon that will
help those uses cases.    So, until we can agree on some API additions that
help SDP munging, many web developers are going to have to keep swinging
that hammer, no matter how much it hurts.

But clearly, if you gave them an API that didn't require such pain, they
would use it.



>
> [1] http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.html
>
> > _____________
> > Roman Shpount
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--bcaec51f99e56fabd204e10100c5
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 5:11 AM, Stefan H=C3=A5kansson LK <span dir=
=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"=
_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/7/13 11:14 PM, Roman Shpount wrote:=
<br>


&gt;<br>
&gt; On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=C3=A5kansson LK<br>
&gt; &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.haka=
nsson@ericsson.com</a><br>
</div><div class=3D"im">&gt; &lt;mailto:<a href=3D"mailto:stefan.lk.hakanss=
on@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 On 7/3/13 11:37 PM, Eric Rescorla wrote:<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt; =C2=A0 =C2=A0 I&#39;m glad to be wrong here. =
=C2=A0Is the phrase &quot;you shouldn&#39;t have to<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt; =C2=A0 =C2=A0 modify the SDP but rather there=
 should be API points to cover<br>
&gt; =C2=A0 =C2=A0 most<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt; =C2=A0 =C2=A0 of the use cases&quot; =C2=A0th=
e consensus of the working group?<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt; I thought it was, but I&#39;m not the chair, =
so maybe you could ask<br>
&gt; =C2=A0 =C2=A0 Harald or<br>
&gt; =C2=A0 =C2=A0 =C2=A0&gt; Stefan.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 I definitely think this is the consensus of the working =
group.<br>
&gt;<br>
&gt;<br>
&gt; Can you point out when exactly this consensus was reached? This is new=
s<br>
&gt; to me, but I definitely could have missed something.<br>
<br>
</div>No I can&#39;t really. To be clear, no consensus call (or similar) ha=
s been<br>
held. But every time this is discussed, what I hear is people agreeing<br>
and no objections. =C2=A0It was discussed a bit, e.g., at the f2f at TPAC l=
ast</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-le=
ft-style:solid;padding-left:1ex">


year, if you read the minutes [1] you will find some traces of that<br>
discussion.<br></blockquote><div><br></div><div>No objections at all? =C2=
=A0That is rather surprising to me. =C2=A0I asked for some feedback the API=
 in the spreadsheet from web developers, at there many objections from them=
 about this part of the API. =C2=A0A majority did say it was good enough fo=
r simple use cases, but he objections were still clearly non-zero.=C2=A0</d=
iv>

<div><br></div><div>I think perhaps the web developers are not well represe=
nted at TPAC. =C2=A0But I could be wrong. =C2=A0Perhaps they are but change=
d their mind. =C2=A0Or maybe they&#39;re objecting on the IETF mailing list=
 when they should object on the IETF mailing list. =C2=A0 Or maybe the spre=
adsheet has a sample bias, while TPAC is the true sample. =C2=A0 I don&#39;=
t know, but it seems like something is going on, and I&#39;d like to know w=
hat it is.</div>

<div><br></div><div>Stefan, what do you think? =C2=A0What could explain the=
 apparent contradiction between objections on the mailing list vs. no objec=
tions f2f?</div><div><br></div><div><br></div><div>Also, on discussions abo=
ut the API, should those who want to propose additions (such as the NoPlan =
JS API I proposed) and other comments put them on the W3C mailing list inst=
ead of the IETF one? =C2=A0Is everyone barking up the wrong tree?</div>

<div><br></div><div>Thanks in advance for your input.</div><div><br></div><=
div><br></div><div><br></div><div><br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


<br>
And, I don&#39;t understand why there would be a debate about this. There<b=
r>
are obviously many who want to define APIs (or constraints) that allow<br>
most use-cases to be met without having to modify the SDP. =C2=A0Why would<=
/blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">


anyone have anything against that?=C2=A0</blockquote><div><div><br></div><d=
iv>I think your question actually has two questions:</div><div><br></div><d=
iv>1. =C2=A0Is anyone against making additions to the API to avoid SDP mung=
ing? =C2=A0</div>

<div><br></div><div>2. =C2=A0Why are they against additions to the API to a=
void SDP munging?</div><div><br></div><div><br></div><div>My personal exper=
ience for #1 is that there are some notable people in the WG who are very m=
uch against making additions to the API to avoid SDP munging. =C2=A0Some of=
 them are rather upset with me for proposing a few minor additions to the A=
PI (the NoPlan JS API) precisely because it allows the JS to go around the =
SDP in many cases, and they don&#39;t like that. =C2=A0So, from my experien=
ce, it&#39;s clear that the answer to #1 is &quot;yes, many do&quot;.=C2=A0=
</div>

<div><br></div><div>And since many object to such additions, and others wan=
t such additions, you end up with debate. =C2=A0</div><div><br></div><div><=
br></div><div>How about #2? =C2=A0Why do they object? =C2=A0My experience i=
s that they view any additions as a threat to API as a threat to the curren=
t API, which means it&#39;s a threat to the current WG progress, which mean=
s it&#39;s a threat to getting done. =C2=A0They are very sensitive to such =
threats, and so object to any additions or changes that would allow JS to a=
void using SDP. =C2=A0Also, I think they don&#39;t want to ways of doing th=
ings, so SDP must remain the one true way.</div>

<div><br></div><div><br></div><div>But that&#39;s just my personal experien=
ce and perspective. =C2=A0I could just be seeing it wrong, or maybe the onl=
y person experiencing this.</div><div><br></div><div><br></div><div><br></d=
iv>

<div><br></div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">I If there is anyone who really=
 wants<br>


to modify the SDP instead (I have no clue why, but anyway) they can<br>
still do that.<br></blockquote><div><br></div><div>Well, no one is going to=
 hit themselves in the head with a hammer if they don&#39;t have to, but th=
ere are many uses cases that currently be done only by SDP munging, and I d=
on&#39;t see any API additions on the horizon that will help those uses cas=
es. =C2=A0 =C2=A0So, until we can agree on some API additions that help SDP=
 munging, many web developers are going to have to keep swinging that hamme=
r, no matter how much it hurts.</div>

<div><br></div><div>But clearly, if you gave them an API that didn&#39;t re=
quire such pain, they would use it.</div><div><br></div><div>=C2=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pad=
ding-left:1ex">


<br>
[1] <a href=3D"http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/00=
72.html" target=3D"_blank">http://lists.w3.org/Archives/Public/public-webrt=
c/2012Nov/0072.html</a><br>
<br>
&gt; _____________<br>
&gt; Roman Shpount<br>
<div class=3D""><div class=3D"h5">&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 07:40:04 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Sun, Jul 7, 2013 at 8:40 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jul 3, 2013, at 2:42 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote=
:
>
> > So compatibility with SIP is important but compatibility with Jingle is
> just impossible.
>
> The mapping of SDP to jingle is in the Jingle specs =E2=80=A6 I'm not exp=
ress any
> opinion on this one way or another other but the authors of theses specs
> have always claimed Jingle fully mapped to and from SDP.
>

I think he meant "impossible without SDP munging", which I think is
undeniable.



>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Sun, Jul 7, 2013 at 8:40 PM,=
 Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" tar=
get=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><div class=3D"gmail_q=
uote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:s=
olid;padding-left:1ex">

<div class=3D"im"><br>
On Jul 3, 2013, at 2:42 PM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:i=
bc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
<br>
&gt; So compatibility with SIP is important but compatibility with Jingle i=
s just impossible.<br>
<br>
</div>The mapping of SDP to jingle is in the Jingle specs =E2=80=A6 I&#39;m=
 not express any opinion on this one way or another other but the authors o=
f theses specs have always claimed Jingle fully mapped to and from SDP.<br>=
</blockquote>

<div><br></div>I think he meant &quot;impossible without SDP munging&quot;,=
 which I think is undeniable.<br><div class=3D"gmail_extra"><br></div><div>=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">


<div class=3D""><div class=3D"h5"><br>
<br>
_______________________________________________<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
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</div></div></blockquote></div><br></div></div>

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On 7/8/13 8:40 AM, Peter Thatcher wrote:
> On Sun, Jul 7, 2013 at 8:40 PM, Cullen Jennings <fluffy@iii.ca
> <mailto:fluffy@iii.ca>> wrote:
> 
> 
>     On Jul 3, 2013, at 2:42 PM, Iñaki Baz Castillo <ibc@aliax.net
>     <mailto:ibc@aliax.net>> wrote:
> 
>     > So compatibility with SIP is important but compatibility with
>     Jingle is just impossible.
> 
>     The mapping of SDP to jingle is in the Jingle specs … I'm not
>     express any opinion on this one way or another other but the authors
>     of theses specs have always claimed Jingle fully mapped to and from SDP.
> 
> 
> I think he meant "impossible without SDP munging", which I think is
> undeniable.

What do you mean by "SDP munging"?

As we know, various things are being added to SDP. The Jingle specs have
not necessarily been updated yet to track those changes, although
proposals have been made on several points. If folks want to keep Jingle
in sync, that work would need to happen at the XSF. Feel free to join
the jingle@xmpp.org list if you have proposals:

http://mail.jabber.org/mailman/listinfo/jingle

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 07:47:59 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 7:45 AM, Peter Saint-Andre <stpeter@stpeter.im>wrote=
:

> On 7/8/13 8:40 AM, Peter Thatcher wrote:
> > On Sun, Jul 7, 2013 at 8:40 PM, Cullen Jennings <fluffy@iii.ca
> > <mailto:fluffy@iii.ca>> wrote:
> >
> >
> >     On Jul 3, 2013, at 2:42 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net
> >     <mailto:ibc@aliax.net>> wrote:
> >
> >     > So compatibility with SIP is important but compatibility with
> >     Jingle is just impossible.
> >
> >     The mapping of SDP to jingle is in the Jingle specs =E2=80=A6 I'm n=
ot
> >     express any opinion on this one way or another other but the author=
s
> >     of theses specs have always claimed Jingle fully mapped to and from
> SDP.
> >
> >
> > I think he meant "impossible without SDP munging", which I think is
> > undeniable.
>
> What do you mean by "SDP munging"?
>

I mean that if the JS wants to send Jingle XML over the wire, it has to
parse the SDP.  Then, when it receives Jingle XML, it has to serialize SDP.
  That parsing and serializing of SDP I call "munging".  We could come up
with better words for more specific activities, but that seems to be the
word everyone else uses, so it's what I've used.




>
> As we know, various things are being added to SDP. The Jingle specs have
> not necessarily been updated yet to track those changes, although
> proposals have been made on several points. If folks want to keep Jingle
> in sync, that work would need to happen at the XSF. Feel free to join
> the jingle@xmpp.org list if you have proposals:
>
> http://mail.jabber.org/mailman/listinfo/jingle
>
> Peter
>
> --
> Peter Saint-Andre
> https://stpeter.im/
>
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 7:45 AM, Peter Saint-Andre <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:stpeter@stpeter.im" target=3D"_blank">stpeter@stpete=
r.im</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/8/13 8:40 AM, Peter T=
hatcher wrote:<br>
&gt; On Sun, Jul 7, 2013 at 8:40 PM, Cullen Jennings &lt;<a href=3D"mailto:=
fluffy@iii.ca">fluffy@iii.ca</a><br>
</div><div class=3D"im">&gt; &lt;mailto:<a href=3D"mailto:fluffy@iii.ca">fl=
uffy@iii.ca</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 On Jul 3, 2013, at 2:42 PM, I=C3=B1aki Baz Castillo &lt;=
<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a><br>
</div><div class=3D"im">&gt; =C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:ibc=
@aliax.net">ibc@aliax.net</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 &gt; So compatibility with SIP is important but compatib=
ility with<br>
&gt; =C2=A0 =C2=A0 Jingle is just impossible.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The mapping of SDP to jingle is in the Jingle specs =E2=
=80=A6 I&#39;m not<br>
&gt; =C2=A0 =C2=A0 express any opinion on this one way or another other but=
 the authors<br>
&gt; =C2=A0 =C2=A0 of theses specs have always claimed Jingle fully mapped =
to and from SDP.<br>
&gt;<br>
&gt;<br>
&gt; I think he meant &quot;impossible without SDP munging&quot;, which I t=
hink is<br>
&gt; undeniable.<br>
<br>
</div>What do you mean by &quot;SDP munging&quot;?<br></blockquote><div><br=
></div><div>I mean that if the JS wants to send Jingle XML over the wire, i=
t has to parse the SDP. =C2=A0Then, when it receives Jingle XML, it has to =
serialize SDP. =C2=A0 That parsing and serializing of SDP I call &quot;mung=
ing&quot;. =C2=A0We could come up with better words for more specific activ=
ities, but that seems to be the word everyone else uses, so it&#39;s what I=
&#39;ve used.</div>

<div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">
<br>
As we know, various things are being added to SDP. The Jingle specs have<br=
>
not necessarily been updated yet to track those changes, although<br>
proposals have been made on several points. If folks want to keep Jingle<br=
>
in sync, that work would need to happen at the XSF. Feel free to join<br>
the <a href=3D"mailto:jingle@xmpp.org">jingle@xmpp.org</a> list if you have=
 proposals:<br>
<br>
<a href=3D"http://mail.jabber.org/mailman/listinfo/jingle" target=3D"_blank=
">http://mail.jabber.org/mailman/listinfo/jingle</a><br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Peter<br>
<br>
--<br>
Peter Saint-Andre<br>
<a href=3D"https://stpeter.im/" target=3D"_blank">https://stpeter.im/</a><b=
r>
<br>
<br>
</font></span></blockquote></div><br></div></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 07:51:12 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Fri, Jul 5, 2013 at 11:33 PM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 7/3/13 11:37 PM, Eric Rescorla wrote:
> > On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher <pthatcher@google.com
> > <mailto:pthatcher@google.com>> wrote:
> >
> >     On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla <ekr@rtfm.com
> >     <mailto:ekr@rtfm.com>> wrote:
> >
> >         On Wed, Jul 3, 2013 at 2:06 PM, Peter Thatcher
> >         <pthatcher@google.com <mailto:pthatcher@google.com>> wrote:
> >
> >             Camp #0:  I've used SDP for years and I'm very comfortable
> >             with it.  Using SDP as the control surface really helps my
> >             use case, which is legacy interop.  Defining an API without
> >             SDP would be too much work, and probably fail.  Look at wha=
t
> >             happened with SDPng!  Supporting all these advanced use
> >             cases doesn't seem worth it.   If developers are doing that
> >             much advanced stuff, they can learn to munge SDP.  It isn't
> >             that bad.
> >
> >
> >         Hmm... That's not my understanding of the situation at all.
> >
> >         Rather, I believe the expectation is that you shouldn't have to
> >         modify the
> >         SDP but rather there should be API points to cover most of the
> >         use cases
> >         that people want. This isn't to say that all those API points
> >         exist or that
> >         they work or whatever.
> >
> >         -Ekr
> >
> >
> >     I'm glad to be wrong here.  Is the phrase "you shouldn't have to
> >     modify the SDP but rather there should be API points to cover most
> >     of the use cases"  the consensus of the working group?
> >
> >
> > I thought it was, but I'm not the chair, so maybe you could ask Harald =
or
> > Stefan.
>
> I definitely think this is the consensus of the working group.
>
> I think the exception would be when interoperating with other systems
> that use another dialect of SDP - I don't think we could cover such
> translations with API points.
>

Are you trying to say "API points couldn't cover uses cases where different
dialects of SDP go over the wire"?  Because I would have to disagree with
that.  I think we could very easily support such uses cases with rather
minor changes to the API.  In fact, the proposal I made for adding just 3
methods (the NoPlan JS API) allows for either Plan A or Plan B (or even
Jingle, mostly), which would make things a lot easier for such use cases.

Now, my design might not be the optimal.  Perhaps we could do better.  But
I think it is possible to serve all these uses cases with a pleasant API.


>
> >
> >
> >     Along with that, is "use Jingle for signalling" included in your se=
t
> >     of "most of the use cases"?
> >
> >
> > No, I don't think it is, since it's not SDP.
> >
> > Maybe I could phrase this differently: It was my understanding that you
> > should have
> > API points to get the PC to emit SDP that would express the policies,
> > preferences,
> > actions, etc. that the application wants. If you want something that's
> > not SDP you would
> > need to gateway.
> >
> > This may or may not be a satisfactory state of affairs, but it was the
> > rule I was using
> > (based on what I thought the WG expected) for when use cases needed new
> API
> > points.
> >
> > -Ekr
> >
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 5, 2013 at 11:33 PM, Stefan H=C3=A5kansson LK <span dir=
=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"=
_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/3/13 11:37 PM, Eric R=
escorla wrote:<br>
&gt; On Wed, Jul 3, 2013 at 2:25 PM, Peter Thatcher &lt;<a href=3D"mailto:p=
thatcher@google.com">pthatcher@google.com</a><br>
</div><div class=3D"im">&gt; &lt;mailto:<a href=3D"mailto:pthatcher@google.=
com">pthatcher@google.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 On Wed, Jul 3, 2013 at 2:20 PM, Eric Rescorla &lt;<a hre=
f=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a><br>
</div><div class=3D"im">&gt; =C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:ekr=
@rtfm.com">ekr@rtfm.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 On Wed, Jul 3, 2013 at 2:06 PM, Peter That=
cher<br>
</div><div class=3D"im">&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;<a href=3D"mai=
lto:pthatcher@google.com">pthatcher@google.com</a> &lt;mailto:<a href=3D"ma=
ilto:pthatcher@google.com">pthatcher@google.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Camp #0: =C2=A0I&#39;ve used=
 SDP for years and I&#39;m very comfortable<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 with it. =C2=A0Using SDP as =
the control surface really helps my<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 use case, which is legacy in=
terop. =C2=A0Defining an API without<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 SDP would be too much work, =
and probably fail. =C2=A0Look at what<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 happened with SDPng! =C2=A0S=
upporting all these advanced use<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 cases doesn&#39;t seem worth=
 it. =C2=A0 If developers are doing that<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 much advanced stuff, they ca=
n learn to munge SDP. =C2=A0It isn&#39;t<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 that bad.<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Hmm... That&#39;s not my understanding of =
the situation at all.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Rather, I believe the expectation is that =
you shouldn&#39;t have to<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 modify the<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 SDP but rather there should be API points =
to cover most of the<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 use cases<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 that people want. This isn&#39;t to say th=
at all those API points<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 exist or that<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 they work or whatever.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 -Ekr<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 I&#39;m glad to be wrong here. =C2=A0Is the phrase &quot=
;you shouldn&#39;t have to<br>
&gt; =C2=A0 =C2=A0 modify the SDP but rather there should be API points to =
cover most<br>
&gt; =C2=A0 =C2=A0 of the use cases&quot; =C2=A0the consensus of the workin=
g group?<br>
&gt;<br>
&gt;<br>
&gt; I thought it was, but I&#39;m not the chair, so maybe you could ask Ha=
rald or<br>
&gt; Stefan.<br>
<br>
</div>I definitely think this is the consensus of the working group.<br>
<br>
I think the exception would be when interoperating with other systems<br>
that use another dialect of SDP - I don&#39;t think we could cover such<br>
translations with API points.<br></blockquote><div><br></div><div>Are you t=
rying to say &quot;API points couldn&#39;t cover uses cases where different=
 dialects of SDP go over the wire&quot;? =C2=A0Because I would have to disa=
gree with that. =C2=A0I think we could very easily support such uses cases =
with rather minor changes to the API. =C2=A0In fact, the proposal I made fo=
r adding just 3 methods (the NoPlan JS API) allows for either Plan A or Pla=
n B (or even Jingle, mostly), which would make things a lot easier for such=
 use cases. =C2=A0</div>

<div><br></div><div>Now, my design might not be the optimal. =C2=A0Perhaps =
we could do better. =C2=A0But I think it is possible to serve all these use=
s cases with a pleasant API.</div><div><br></div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">


<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Along with that, is &quot;use Jingle for signalling&quot=
; included in your set<br>
&gt; =C2=A0 =C2=A0 of &quot;most of the use cases&quot;?<br>
&gt;<br>
&gt;<br>
&gt; No, I don&#39;t think it is, since it&#39;s not SDP.<br>
&gt;<br>
&gt; Maybe I could phrase this differently: It was my understanding that yo=
u<br>
&gt; should have<br>
&gt; API points to get the PC to emit SDP that would express the policies,<=
br>
&gt; preferences,<br>
&gt; actions, etc. that the application wants. If you want something that&#=
39;s<br>
&gt; not SDP you would<br>
&gt; need to gateway.<br>
&gt;<br>
&gt; This may or may not be a satisfactory state of affairs, but it was the=
<br>
&gt; rule I was using<br>
&gt; (based on what I thought the WG expected) for when use cases needed ne=
w API<br>
&gt; points.<br>
&gt;<br>
&gt; -Ekr<br>
&gt;<br>
<br>
</div></div></blockquote></div><br></div></div>

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, 8 Jul 2013, Peter Thatcher wrote:
> > Â  Â  > So compatibility with SIP is important but compatibility with
> > Â  Â  Jingle is just impossible.
> >
> > Â  Â  The mapping of SDP to jingle is in the Jingle specs ? I'm not
> > Â  Â  express any opinion on this one way or another other but the authors
> > Â  Â  of theses specs have always claimed Jingle fully mapped to and from SDP.
> >
> >
> > I think he meant "impossible without SDP munging", which I think is
> > undeniable.
> 
> What do you mean by "SDP munging"?
> 
> 
> I mean that if the JS wants to send Jingle XML over the wire, it has to parse the SDP. Â Then, when it receives Jingle XML, it has to serialize SDP. Â  That parsing and serializing of SDP I call "munging". Â We could come up with better words for more specific
> activities, but that seems to be the word everyone else uses, so it's what I've used.

I think "mangle" is a better term here. People who want to do jingle are 
aware of the fact that this is more difficult than running their own 
proprietary stuff over xml. Or use SoX :-)
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On Wed, Jul 3, 2013 at 11:35 PM, Philipp Hancke
<fippo@goodadvice.pages.de>wrote:

> On Wed, 3 Jul 2013, Eric Rescorla wrote:
>
>> Who said anything about impossible? It's a mechanical transformation
>> (see: http://xmpp.org/**extensions/xep-0167.html<http://xmpp.org/extensions/xep-0167.html>
>> ).
>>
>
> That is true for simple usage and works reasonably well. See my
> strophe.jingle code on github. I'd note that I want to revamp
> the API such that it wraps the peerconnections SDP apis more closely.
>
> Things like the content-add/content-remove are harder because the API user
> is required to calculate an SDP delta/diff. The appendix A.1.3 of the JSEP
> draft calls this createContentAdd, parseContentAdd, createContentAccept and
> parseContentAccept. Has anyone ever implemented those operations in
> javascript and can demonstrate running code?
>

I think those were added about a week ago.  Both Chrome and Firefox try to
update quickly, but that would be amazing!


> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 11:35 PM, Philipp Hancke <span dir=3D"ltr">&=
lt;<a href=3D"mailto:fippo@goodadvice.pages.de" target=3D"_blank">fippo@goo=
dadvice.pages.de</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On Wed, 3 Jul 2013, Eric R=
escorla wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Who said anything about impossible? It&#39;s a mechanical transformation<br=
>
(see:=C2=A0<a href=3D"http://xmpp.org/extensions/xep-0167.html" target=3D"_=
blank">http://xmpp.org/<u></u>extensions/xep-0167.html</a>).<br>
</blockquote>
<br></div>
That is true for simple usage and works reasonably well. See my<br>
strophe.jingle code on github. I&#39;d note that I want to revamp<br>
the API such that it wraps the peerconnections SDP apis more closely.<br>
<br>
Things like the content-add/content-remove are harder because the API user =
is required to calculate an SDP delta/diff. The appendix A.1.3 of the JSEP =
draft calls this createContentAdd, parseContentAdd, createContentAccept and=
 parseContentAccept. Has anyone ever implemented those operations in javasc=
ript and can demonstrate running code?<br>

</blockquote><div><br></div><div>I think those were added about a week ago.=
 =C2=A0Both Chrome and Firefox try to update quickly, but that would be ama=
zing!</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 07:56:13 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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I'm fine with "mangle".


On Mon, Jul 8, 2013 at 7:54 AM, Philipp Hancke <fippo@goodadvice.pages.de>wrote:

> On Mon, 8 Jul 2013, Peter Thatcher wrote:
>
>> >     > So compatibility with SIP is important but compatibility with
>> >     Jingle is just impossible.
>> >
>> >     The mapping of SDP to jingle is in the Jingle specs ? I'm not
>> >     express any opinion on this one way or another other but the authors
>> >     of theses specs have always claimed Jingle fully mapped to and from
>> SDP.
>> >
>> >
>> > I think he meant "impossible without SDP munging", which I think is
>> > undeniable.
>>
>> What do you mean by "SDP munging"?
>>
>>
>> I mean that if the JS wants to send Jingle XML over the wire, it has to
>> parse the SDP.  Then, when it receives Jingle XML, it has to serialize SDP.
>>   That parsing and serializing of SDP I call "munging".  We could come up
>> with better words for more specific
>> activities, but that seems to be the word everyone else uses, so it's
>> what I've used.
>>
>
> I think "mangle" is a better term here. People who want to do jingle are
> aware of the fact that this is more difficult than running their own
> proprietary stuff over xml. Or use SoX :-)

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<div dir=3D"ltr">I&#39;m fine with &quot;mangle&quot;.=C2=A0</div><div clas=
s=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Jul 8, 2013 at=
 7:54 AM, Philipp Hancke <span dir=3D"ltr">&lt;<a href=3D"mailto:fippo@good=
advice.pages.de" target=3D"_blank">fippo@goodadvice.pages.de</a>&gt;</span>=
 wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On Mon, 8 Jul 2013, Peter =
Thatcher wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
&gt; =C2=A0 =C2=A0 &gt; So compatibility with SIP is important but compatib=
ility with<br>
&gt; =C2=A0 =C2=A0 Jingle is just impossible.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The mapping of SDP to jingle is in the Jingle specs ? I&=
#39;m not<br>
&gt; =C2=A0 =C2=A0 express any opinion on this one way or another other but=
 the authors<br>
&gt; =C2=A0 =C2=A0 of theses specs have always claimed Jingle fully mapped =
to and from SDP.<br>
&gt;<br>
&gt;<br>
&gt; I think he meant &quot;impossible without SDP munging&quot;, which I t=
hink is<br>
&gt; undeniable.<br>
<br>
What do you mean by &quot;SDP munging&quot;?<br>
<br>
<br>
I mean that if the JS wants to send Jingle XML over the wire, it has to par=
se the SDP. =C2=A0Then, when it receives Jingle XML, it has to serialize SD=
P. =C2=A0 That parsing and serializing of SDP I call &quot;munging&quot;. =
=C2=A0We could come up with better words for more specific<br>


activities, but that seems to be the word everyone else uses, so it&#39;s w=
hat I&#39;ve used.<br>
</blockquote>
<br></div>
I think &quot;mangle&quot; is a better term here. People who want to do jin=
gle are aware of the fact that this is more difficult than running their ow=
n proprietary stuff over xml. Or use SoX :-)</blockquote></div><br></div>


--bcaec520eacd45a49a04e101410c--

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 08:00:41 -0700
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On Sat, Jul 6, 2013 at 3:59 AM, Silvia Pfeiffer
<silviapfeiffer1@gmail.com>wrote:

> An interesting read. It makes me wonder if there is a full description
> available of the SDP that browsers (Chrome, Firefox) currently
> support.
>

Here's the description for Chrome:

https://code.google.com/p/libjingle/source/browse/trunk/talk/app/webrtc/webrtcsdp.cc

:)


>
> Looking forward to your proposal (I assume it will be on the W3C list?).
>
> Cheers,
> Silvia.
>
> On Wed, Jun 26, 2013 at 2:37 PM, Robin Raymond <robin@hookflash.com>
> wrote:
> >
> > According to many, real adoption of WebRTC will not happen if we
> continue to
> > force everyone to use this SDP Offer/Answer methodology.  It is clearly
> > blocking our way forward and the amount of specification documentation
> > remaining needed for the browser vendors to produce a compatible SDP
> based
> > WebRTC engine in a browser is much more daunting than most are willing to
> > admit.
> >
> >
> > The draft rationale behind a JavaScript Object API model:
> >
> >
> http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00.txt
> >
> > Whereas:
> >
> > The WebRTC JavaScript Object API & Shim document will be along shortly.
> >
> >
> > We wanted to get this initial rationale draft informational document out
> > right away. Everyone that has any interest should review the reasons and
> > concepts and comment before we get too far along on the details of an
> actual
> > API, the initial API will follow in the coming days.
> >
> >
> > Best regards,
> >
> > Robin Raymond
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sat, Jul 6, 2013 at 3:59 AM, Silvia Pfeiffer <span dir=3D"ltr">&=
lt;<a href=3D"mailto:silviapfeiffer1@gmail.com" target=3D"_blank">silviapfe=
iffer1@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">An interesting read. It makes me wonder if there is a full=
 description<br>


available of the SDP that browsers (Chrome, Firefox) currently<br>
support.<br></blockquote><div><br></div><div>Here&#39;s the description for=
 Chrome:</div><div><br></div><div><a href=3D"https://code.google.com/p/libj=
ingle/source/browse/trunk/talk/app/webrtc/webrtcsdp.cc">https://code.google=
.com/p/libjingle/source/browse/trunk/talk/app/webrtc/webrtcsdp.cc</a><br>

</div><div><br></div><div>:)</div><div>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-le=
ft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<br>
Looking forward to your proposal (I assume it will be on the W3C list?).<br=
>
<br>
Cheers,<br>
Silvia.<br>
<div class=3D""><div class=3D"h5"><br>
On Wed, Jun 26, 2013 at 2:37 PM, Robin Raymond &lt;<a href=3D"mailto:robin@=
hookflash.com">robin@hookflash.com</a>&gt; wrote:<br>
&gt;<br>
&gt; According to many, real adoption of WebRTC will not happen if we conti=
nue to<br>
&gt; force everyone to use this SDP Offer/Answer methodology. =C2=A0It is c=
learly<br>
&gt; blocking our way forward and the amount of specification documentation=
<br>
&gt; remaining needed for the browser vendors to produce a compatible SDP b=
ased<br>
&gt; WebRTC engine in a browser is much more daunting than most are willing=
 to<br>
&gt; admit.<br>
&gt;<br>
&gt;<br>
&gt; The draft rationale behind a JavaScript Object API model:<br>
&gt;<br>
&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-we=
brtc-js-obj-api-rationale-00.txt" target=3D"_blank">http://www.ietf.org/int=
ernet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00.txt</a><br=
>


&gt;<br>
&gt; Whereas:<br>
&gt;<br>
&gt; The WebRTC JavaScript Object API &amp; Shim document will be along sho=
rtly.<br>
&gt;<br>
&gt;<br>
&gt; We wanted to get this initial rationale draft informational document o=
ut<br>
&gt; right away. Everyone that has any interest should review the reasons a=
nd<br>
&gt; concepts and comment before we get too far along on the details of an =
actual<br>
&gt; API, the initial API will follow in the coming days.<br>
&gt;<br>
&gt;<br>
&gt; Best regards,<br>
&gt;<br>
&gt; Robin Raymond<br>
&gt;<br>
&gt;<br>
</div></div><div class=3D""><div class=3D"h5">&gt; ________________________=
_______________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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From: Jim Barnett <Jim.Barnett@genesyslab.com>
To: Peter Thatcher <pthatcher@google.com>, Philipp Hancke <fippo@goodadvice.pages.de>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Date: Mon, 8 Jul 2013 15:02:24 +0000
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Date: Mon, 8 Jul 2013 11:22:38 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 8:11 AM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> > On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson LK
> > <stefan.lk.hakansson@ericsson.com
> > <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
> >
> >     On 7/3/13 11:37 PM, Eric Rescorla wrote:
> >      >     I'm glad to be wrong here.  Is the phrase "you shouldn't hav=
e
> to
> >      >     modify the SDP but rather there should be API points to cove=
r
> >     most
> >      >     of the use cases"  the consensus of the working group?
> >      >
> >      >
> >      > I thought it was, but I'm not the chair, so maybe you could ask
> >     Harald or
> >      > Stefan.
> >
> >     I definitely think this is the consensus of the working group.
> >
> >
> > Can you point out when exactly this consensus was reached? This is news
> > to me, but I definitely could have missed something.
>
> No I can't really. To be clear, no consensus call (or similar) has been
> held. But every time this is discussed, what I hear is people agreeing
> and no objections. It was discussed a bit, e.g., at the f2f at TPAC last
> year, if you read the minutes [1] you will find some traces of that
> discussion.
>

> And, I don't understand why there would be a debate about this. There
> are obviously many who want to define APIs (or constraints) that allow
> most use-cases to be met without having to modify the SDP. Why would
> anyone have anything against that? If there is anyone who really wants
> to modify the SDP instead (I have no clue why, but anyway) they can
> still do that.
>
> [1] http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.html
>
>
I would not disagree that there should not be any debate that things that
can be done through API should be done through API. My question is, was
there ever consensus that SDP should be a opaque non modifiable blob which
should not be used within the application to control things. Or is SDP just
a more advanced API surface that is supposed to be used to implement more
advanced scenarios? Is this group even suppose to specify what can be
modified in by the application SDP or is it supposed to be an empty set?
So, was there ever a consensus that "developers MUST NOT touch SDP"?

_____________
Roman Shpount

--001a11c1879c5e4e6c04e1019da9
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<div>On Mon, Jul 8, 2013 at 8:11 AM, Stefan H=E5kansson LK <span dir=3D"ltr=
">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank"=
>stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:</div><div class=3D"=
gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">&gt; On Sat, Jul 6, 2013 a=
t 2:33 AM, Stefan H=E5kansson LK<br>
&gt; &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.haka=
nsson@ericsson.com</a><br>
</div><div class=3D"im">&gt; &lt;mailto:<a href=3D"mailto:stefan.lk.hakanss=
on@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 On 7/3/13 11:37 PM, Eric Rescorla wrote:<br>
&gt; =A0 =A0 =A0&gt; =A0 =A0 I&#39;m glad to be wrong here. =A0Is the phras=
e &quot;you shouldn&#39;t have to<br>
&gt; =A0 =A0 =A0&gt; =A0 =A0 modify the SDP but rather there should be API =
points to cover<br>
&gt; =A0 =A0 most<br>
&gt; =A0 =A0 =A0&gt; =A0 =A0 of the use cases&quot; =A0the consensus of the=
 working group?<br>
&gt; =A0 =A0 =A0&gt;<br>
&gt; =A0 =A0 =A0&gt;<br>
&gt; =A0 =A0 =A0&gt; I thought it was, but I&#39;m not the chair, so maybe =
you could ask<br>
&gt; =A0 =A0 Harald or<br>
&gt; =A0 =A0 =A0&gt; Stefan.<br>
&gt;<br>
&gt; =A0 =A0 I definitely think this is the consensus of the working group.=
<br>
&gt;<br>
&gt;<br>
&gt; Can you point out when exactly this consensus was reached? This is new=
s<br>
&gt; to me, but I definitely could have missed something.<br>
<br>
</div>No I can&#39;t really. To be clear, no consensus call (or similar) ha=
s been<br>
held. But every time this is discussed, what I hear is people agreeing<br>
and no objections. It was discussed a bit, e.g., at the f2f at TPAC last<br=
>
year, if you read the minutes [1] you will find some traces of that<br>
discussion.<br></blockquote><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
And, I don&#39;t understand why there would be a debate about this. There<b=
r>
are obviously many who want to define APIs (or constraints) that allow<br>
most use-cases to be met without having to modify the SDP. Why would<br>
anyone have anything against that? If there is anyone who really wants<br>
to modify the SDP instead (I have no clue why, but anyway) they can<br>
still do that.<br>
<br>
[1] <a href=3D"http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/00=
72.html" target=3D"_blank">http://lists.w3.org/Archives/Public/public-webrt=
c/2012Nov/0072.html</a><br>
<br></blockquote><div><br></div><div>I would not disagree that there should=
 not be any debate that things that can be done through API should be done =
through API. My question is, was there ever consensus that SDP should be a =
opaque non modifiable blob which should not be used within the application =
to control things. Or is SDP just a more advanced API surface that is suppo=
sed to be used to implement more advanced scenarios? Is this group even sup=
pose to specify what can be modified in by the application SDP or is it sup=
posed to be an empty set? So, was there ever a consensus that &quot;develop=
ers MUST NOT touch SDP&quot;?</div>
<div><br></div><div>_____________<br>Roman Shpount</div><div>=A0</div></div=
>

--001a11c1879c5e4e6c04e1019da9--

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Whoops, nevermind.  It's been there since
http://tools.ietf.org/html/draft-uberti-rtcweb-jsep-00, and it's not what I
thought it was at first glance.  So forget what I said :).


On Mon, Jul 8, 2013 at 7:54 AM, Peter Thatcher <pthatcher@google.com> wrote:

>
>
>
> On Wed, Jul 3, 2013 at 11:35 PM, Philipp Hancke <fippo@goodadvice.pages.de
> > wrote:
>
>> On Wed, 3 Jul 2013, Eric Rescorla wrote:
>>
>>> Who said anything about impossible? It's a mechanical transformation
>>> (see: http://xmpp.org/**extensions/xep-0167.html<http://xmpp.org/extensions/xep-0167.html>
>>> ).
>>>
>>
>> That is true for simple usage and works reasonably well. See my
>> strophe.jingle code on github. I'd note that I want to revamp
>> the API such that it wraps the peerconnections SDP apis more closely.
>>
>> Things like the content-add/content-remove are harder because the API
>> user is required to calculate an SDP delta/diff. The appendix A.1.3 of the
>> JSEP draft calls this createContentAdd, parseContentAdd,
>> createContentAccept and parseContentAccept. Has anyone ever implemented
>> those operations in javascript and can demonstrate running code?
>>
>
> I think those were added about a week ago.  Both Chrome and Firefox try to
> update quickly, but that would be amazing!
>
>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

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<div dir=3D"ltr">Whoops, nevermind. =C2=A0It&#39;s been there since=C2=A0<a=
 href=3D"http://tools.ietf.org/html/draft-uberti-rtcweb-jsep-00">http://too=
ls.ietf.org/html/draft-uberti-rtcweb-jsep-00</a>, and it&#39;s not what I t=
hought it was at first glance. =C2=A0So forget what I said :).</div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Jul 8=
, 2013 at 7:54 AM, Peter Thatcher <span dir=3D"ltr">&lt;<a href=3D"mailto:p=
thatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt;</span> =
wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Wed, Jul 3=
, 2013 at 11:35 PM, Philipp Hancke <span dir=3D"ltr">&lt;<a href=3D"mailto:=
fippo@goodadvice.pages.de" target=3D"_blank">fippo@goodadvice.pages.de</a>&=
gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>On Wed, 3 Jul 2013, Eric Rescorla wrote=
:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Who said anything about impossible? It&#39;s a mechanical transformation<br=
>
(see:=C2=A0<a href=3D"http://xmpp.org/extensions/xep-0167.html" target=3D"_=
blank">http://xmpp.org/<u></u>extensions/xep-0167.html</a>).<br>
</blockquote>
<br></div>
That is true for simple usage and works reasonably well. See my<br>
strophe.jingle code on github. I&#39;d note that I want to revamp<br>
the API such that it wraps the peerconnections SDP apis more closely.<br>
<br>
Things like the content-add/content-remove are harder because the API user =
is required to calculate an SDP delta/diff. The appendix A.1.3 of the JSEP =
draft calls this createContentAdd, parseContentAdd, createContentAccept and=
 parseContentAccept. Has anyone ever implemented those operations in javasc=
ript and can demonstrate running code?<br>


</blockquote><div><br></div></div></div><div>I think those were added about=
 a week ago. =C2=A0Both Chrome and Firefox try to update quickly, but that =
would be amazing!</div><div class=3D"im"><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">


_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div></div><br></div></div>
</blockquote></div><br></div>

--bcaec51f99e5b2318704e101ac8d--

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To: Roman Shpount <roman@telurix.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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As I recall the consensus, developers are supposed to be able to modify the=
 SDP, but we needed to specify what parts of the SDP could be modified.  In=
 other words, there might be some limits to what could be modified, but we =
haven't yet determined/agreed on what they are.


-          Jim

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: Monday, July 08, 2013 11:23 AM
To: Stefan H=E5kansson LK
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the cu=
rrent WebRTC API and SDP as a control surface

On Mon, Jul 8, 2013 at 8:11 AM, Stefan H=E5kansson LK <stefan.lk.hakansson@=
ericsson.com<mailto:stefan.lk.hakansson@ericsson.com>> wrote:
> On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson LK
> <stefan.lk.hakansson@ericsson.com<mailto:stefan.lk.hakansson@ericsson.com=
>
> <mailto:stefan.lk.hakansson@ericsson.com<mailto:stefan.lk.hakansson@erics=
son.com>>> wrote:
>
>     On 7/3/13 11:37 PM, Eric Rescorla wrote:
>      >     I'm glad to be wrong here.  Is the phrase "you shouldn't have =
to
>      >     modify the SDP but rather there should be API points to cover
>     most
>      >     of the use cases"  the consensus of the working group?
>      >
>      >
>      > I thought it was, but I'm not the chair, so maybe you could ask
>     Harald or
>      > Stefan.
>
>     I definitely think this is the consensus of the working group.
>
>
> Can you point out when exactly this consensus was reached? This is news
> to me, but I definitely could have missed something.
No I can't really. To be clear, no consensus call (or similar) has been
held. But every time this is discussed, what I hear is people agreeing
and no objections. It was discussed a bit, e.g., at the f2f at TPAC last
year, if you read the minutes [1] you will find some traces of that
discussion.

And, I don't understand why there would be a debate about this. There
are obviously many who want to define APIs (or constraints) that allow
most use-cases to be met without having to modify the SDP. Why would
anyone have anything against that? If there is anyone who really wants
to modify the SDP instead (I have no clue why, but anyway) they can
still do that.

[1] http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.html

I would not disagree that there should not be any debate that things that c=
an be done through API should be done through API. My question is, was ther=
e ever consensus that SDP should be a opaque non modifiable blob which shou=
ld not be used within the application to control things. Or is SDP just a m=
ore advanced API surface that is supposed to be used to implement more adva=
nced scenarios? Is this group even suppose to specify what can be modified =
in by the application SDP or is it supposed to be an empty set? So, was the=
re ever a consensus that "developers MUST NOT touch SDP"?

_____________
Roman Shpount

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">As I recall the consensus=
, developers are supposed to be able to modify the SDP, but we needed to sp=
ecify what parts of the SDP could be modified. &nbsp;In other
 words, there might be some limits to what could be modified, but we haven&=
#8217;t yet determined/agreed on what they are.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
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list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Jim<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> Monday, July 08, 2013 11:23 AM<br>
<b>To:</b> Stefan H=E5kansson LK<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] Summary of Application Developers' opinions of=
 the current WebRTC API and SDP as a control surface<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Mon, Jul 8, 2013 at 8:11 AM, Stefan H=E5kansson L=
K &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank"=
>stefan.lk.hakansson@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<p class=3D"MsoNormal">&gt; On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kans=
son LK<br>
&gt; &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.haka=
nsson@ericsson.com</a><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&gt; &lt;mailto:<a hr=
ef=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@ericsson=
.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; &nbsp; &nbsp; On 7/3/13 11:37 PM, Eric Rescorla wrote:<br>
&gt; &nbsp; &nbsp; &nbsp;&gt; &nbsp; &nbsp; I'm glad to be wrong here. &nbs=
p;Is the phrase &quot;you shouldn't have to<br>
&gt; &nbsp; &nbsp; &nbsp;&gt; &nbsp; &nbsp; modify the SDP but rather there=
 should be API points to cover<br>
&gt; &nbsp; &nbsp; most<br>
&gt; &nbsp; &nbsp; &nbsp;&gt; &nbsp; &nbsp; of the use cases&quot; &nbsp;th=
e consensus of the working group?<br>
&gt; &nbsp; &nbsp; &nbsp;&gt;<br>
&gt; &nbsp; &nbsp; &nbsp;&gt;<br>
&gt; &nbsp; &nbsp; &nbsp;&gt; I thought it was, but I'm not the chair, so m=
aybe you could ask<br>
&gt; &nbsp; &nbsp; Harald or<br>
&gt; &nbsp; &nbsp; &nbsp;&gt; Stefan.<br>
&gt;<br>
&gt; &nbsp; &nbsp; I definitely think this is the consensus of the working =
group.<br>
&gt;<br>
&gt;<br>
&gt; Can you point out when exactly this consensus was reached? This is new=
s<br>
&gt; to me, but I definitely could have missed something.<o:p></o:p></p>
</div>
<p class=3D"MsoNormal">No I can't really. To be clear, no consensus call (o=
r similar) has been<br>
held. But every time this is discussed, what I hear is people agreeing<br>
and no objections. It was discussed a bit, e.g., at the f2f at TPAC last<br=
>
year, if you read the minutes [1] you will find some traces of that<br>
discussion.<o:p></o:p></p>
</blockquote>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
And, I don't understand why there would be a debate about this. There<br>
are obviously many who want to define APIs (or constraints) that allow<br>
most use-cases to be met without having to modify the SDP. Why would<br>
anyone have anything against that? If there is anyone who really wants<br>
to modify the SDP instead (I have no clue why, but anyway) they can<br>
still do that.<br>
<br>
[1] <a href=3D"http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/00=
72.html" target=3D"_blank">
http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.html</a><o:p=
></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I would not disagree that there should not be any de=
bate that things that can be done through API should be done through API. M=
y question is, was there ever consensus that SDP should be a opaque non mod=
ifiable blob which should not be used
 within the application to control things. Or is SDP just a more advanced A=
PI surface that is supposed to be used to implement more advanced scenarios=
? Is this group even suppose to specify what can be modified in by the appl=
ication SDP or is it supposed to
 be an empty set? So, was there ever a consensus that &quot;developers MUST=
 NOT touch SDP&quot;?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2013 08:30:16 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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On Wed, Jul 3, 2013 at 3:20 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> 2013/7/3 Hadriel Kaplan <hadriel.kaplan@oracle.com>:
> > You might also want to cite some of the reasons noted in this old draft=
:
> > http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01
>
> Well, section "3. Defining a WebRTC Protocol in the Browser" says:
>
>
>    9) Using the SDP offer/answer model provides a more rigid API
>      interaction model, enabling Browser vendors to perform less
>      testing and provide more robust implementations than exposing all
>      discrete components to a Javascript API would.
>
>    10)   Using a higher-level API model, such as would be done with an
>      SDP offer/answer model, means the cross-browser vendor-specific
>      variances would be reduced.  Exposing a lower-level API would
>      inevitably lead to some differences in different browsers due to
>      differences in their architectures/implementation.
>
>
> Time to reconsider those assertions? ;)
>
>
>
These assertions assume we have only two choices:  1.  A high level
SDP-based API or 2.   A low-level API that "[exposes] all discrete
components".

Was a "middle ground" API ever considered?  One which  doesn't require
everything go through SDP, and also doesn't expose "all discrete
components"?   A sort of "best of both worlds" approach?   I'd be
interested to know if such an ideas was proposed and rejected or simply
never proposed.



> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 3, 2013 at 3:20 PM, I=C3=B1aki Baz Castillo <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.n=
et</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">2013/7/3 Hadriel Kaplan &lt;<a href=3D"mailto:hadriel.kapl=
an@oracle.com" target=3D"_blank">hadriel.kaplan@oracle.com</a>&gt;:<br>



<div>&gt; You might also want to cite some of the reasons noted in this old=
 draft:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-01"=
 target=3D"_blank">http://tools.ietf.org/html/draft-kaplan-rtcweb-api-reqs-=
01</a><br>
<br>
</div>Well, section &quot;3. Defining a WebRTC Protocol in the Browser&quot=
; says:<br>
<br>
<br>
=C2=A0 =C2=A09) Using the SDP offer/answer model provides a more rigid API<=
br>
=C2=A0 =C2=A0 =C2=A0interaction model, enabling Browser vendors to perform =
less<br>
=C2=A0 =C2=A0 =C2=A0testing and provide more robust implementations than ex=
posing all<br>
=C2=A0 =C2=A0 =C2=A0discrete components to a Javascript API would.<br>
<br>
=C2=A0 =C2=A010) =C2=A0 Using a higher-level API model, such as would be do=
ne with an<br>
=C2=A0 =C2=A0 =C2=A0SDP offer/answer model, means the cross-browser vendor-=
specific<br>
=C2=A0 =C2=A0 =C2=A0variances would be reduced. =C2=A0Exposing a lower-leve=
l API would<br>
=C2=A0 =C2=A0 =C2=A0inevitably lead to some differences in different browse=
rs due to<br>
=C2=A0 =C2=A0 =C2=A0differences in their architectures/implementation.<br>
<br>
<br>
Time to reconsider those assertions? ;)<br>
<div><br>
<br></div></blockquote><div><br></div><div>These assertions assume we have =
only two choices: =C2=A01. =C2=A0A high level SDP-based API or 2. =C2=A0 A =
low-level API that &quot;[exposes] all discrete components&quot;. =C2=A0=C2=
=A0</div><div><br></div>


<div>Was a &quot;middle ground&quot; API ever considered? =C2=A0One which =
=C2=A0doesn&#39;t require everything go through SDP, and also doesn&#39;t e=
xpose &quot;all discrete components&quot;? =C2=A0 A sort of &quot;best of b=
oth worlds&quot; approach? =C2=A0 I&#39;d be interested to know if such an =
ideas was proposed and rejected or simply never proposed.</div>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex"><div>

--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
</div><div><div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a11330f2a0e7bee04e101bbff--

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 11:29 AM, Jim Barnett <Jim.Barnett@genesyslab.com>wr=
ote:

>  As I recall the consensus, developers are supposed to be able to modify
> the SDP, but we needed to specify what parts of the SDP could be modified=
.
>  In other words, there might be some limits to what could be modified, bu=
t
> we haven=92t yet determined/agreed on what they are.****
>
>
>
This is exactly what I thought.
_____________
Roman Shpount

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<br><div class=3D"gmail_quote">On Mon, Jul 8, 2013 at 11:29 AM, Jim Barnett=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:Jim.Barnett@genesyslab.com" target=
=3D"_blank">Jim.Barnett@genesyslab.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As I recall the consensus=
, developers are supposed to be able to modify the SDP, but we needed to sp=
ecify what parts of the SDP could be modified. =A0In other
 words, there might be some limits to what could be modified, but we haven=
=92t yet determined/agreed on what they are.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>This is exactly what I thought.</div><div>_____________<br>Roman Shpount</=
div><div>=A0</div></div>

--089e012292e2264e8504e101d3d1--

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 8 Jul 2013 09:18:07 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Y


On Mon, Jul 8, 2013 at 8:37 AM, Roman Shpount <roman@telurix.com> wrote:

>
> On Mon, Jul 8, 2013 at 11:29 AM, Jim Barnett <Jim.Barnett@genesyslab.com>=
wrote:
>
>>  As I recall the consensus, developers are supposed to be able to modify
>> the SDP, but we needed to specify what parts of the SDP could be modifie=
d.
>>  In other words, there might be some limits to what could be modified, b=
ut
>> we haven=92t yet determined/agreed on what they are.****
>>
>>
>>
> This is exactly what I thought.
>
> Yes, this matches my understanding as well. However, I believe there was
also
general agreement that most things one would want to do should be
doable without modifying SDP.

-Ekr

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<div dir=3D"ltr">Y<br><div class=3D"gmail_extra"><br><br><div class=3D"gmai=
l_quote">On Mon, Jul 8, 2013 at 8:37 AM, Roman Shpount <span dir=3D"ltr">&l=
t;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com<=
/a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br><div class=3D"gmail_quote"><div class=3D=
"im">On Mon, Jul 8, 2013 at 11:29 AM, Jim Barnett <span dir=3D"ltr">&lt;<a =
href=3D"mailto:Jim.Barnett@genesyslab.com" target=3D"_blank">Jim.Barnett@ge=
nesyslab.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As I recall the consensus=
, developers are supposed to be able to modify the SDP, but we needed to sp=
ecify what parts of the SDP could be modified. =A0In other
 words, there might be some limits to what could be modified, but we haven=
=92t yet determined/agreed on what they are.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div></di=
v><div>This is exactly what I thought.</div><div><br></div></div></blockquo=
te><div style>Yes, this matches my understanding as well. However, I believ=
e there was also</div>

<div style>general agreement that most things one would want to do should b=
e</div><div style>doable without modifying SDP.</div><div style><br></div><=
div style>-Ekr</div><div style>=A0<br></div></div></div></div>

--047d7bdc95142a570b04e102661c--

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 12:18 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> Yes, this matches my understanding as well. However, I believe there was
> also
> general agreement that most things one would want to do should be
> doable without modifying SDP.
>


"Should be doable" being the operative word. We are not there yet. We need
both new API types (like capabilites) and more settings for existing API
methods(more constraints).This also means either a lot of API extensions in
the future (with new API extension for each new SDP based feature) or
always having SDP as a fallback API surface.  If this is done right we can
just keep constraints and get rid of SDP altogether.

______________
Roman Shpount

--089e0112cf9afb6a9704e1029637
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<div class=3D"gmail_quote">On Mon, Jul 8, 2013 at 12:18 PM, Eric Rescorla <=
span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@=
rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>Yes, this matches my understanding as well. However, I believe there was a=
lso</div>

<div>general agreement that most things one would want to do should be</div=
><div>doable without modifying SDP.</div></div></div></div></blockquote><di=
v>=A0</div><div><br></div><div>&quot;Should be doable&quot; being the opera=
tive word. We are not there yet. We need both new API types (like capabilit=
es) and more settings for existing API methods(more constraints).This also =
means either a lot of API extensions in the future (with new API extension =
for each new SDP based feature) or always having SDP as a fallback API surf=
ace. =A0If this is done right we can just keep constraints and get rid of S=
DP altogether.</div>
<div><br></div><div>______________</div><div>Roman Shpount</div></div>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 8 Jul 2013 09:41:17 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 9:32 AM, Roman Shpount <roman@telurix.com> wrote:

> On Mon, Jul 8, 2013 at 12:18 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> Yes, this matches my understanding as well. However, I believe there was
>> also
>> general agreement that most things one would want to do should be
>> doable without modifying SDP.
>>
>
>
> "Should be doable" being the operative word. We are not there yet. We need
> both new API types (like capabilites) and more settings for existing API
> methods(more constraints).
>

Yes, I agree with that. What would be useful would be for people to
make a list of these use cases and the extensions which would
enable them.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 9:32 AM, Roman Shpount <span dir=3D"ltr">&lt=
;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</=
a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"gmail_quote"><div class=3D"im"=
>On Mon, Jul 8, 2013 at 12:18 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wr=
ote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>Yes, this matches my understanding as well. However, I believe there was a=
lso</div>

<div>general agreement that most things one would want to do should be</div=
><div>doable without modifying SDP.</div></div></div></div></blockquote><di=
v>=A0</div><div><br></div></div><div>&quot;Should be doable&quot; being the=
 operative word. We are not there yet. We need both new API types (like cap=
abilites) and more settings for existing API methods(more constraints).</di=
v>

</div></blockquote><div><br></div><div style>Yes, I agree with that. What w=
ould be useful would be for people to</div><div style>make a list of these =
use cases and the extensions which would</div><div style>enable them.</div>

<div style><br></div><div style>-Ekr</div><div style><br></div><div style><=
br></div><div style>=A0</div></div></div></div>

--047d7b6da78210e1b204e102b9af--

From martin.thomson@gmail.com  Mon Jul  8 09:48:05 2013
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Two separate replies...

On 8 July 2013 05:23, Simon Perreault <simon.perreault@viagenie.ca> wrote:
> Le 2013-07-08 06:24, Justin Uberti a =C3=A9crit :
>> Just uploaded a 00 version of a spec for requesting time-limited TURN
>> credentials for WebRTC apps. Would like to get 10 minutes of agenda time
>> to present this in Berlin.

Justin, is this the right working group for this draft?  Presumably,
you want to talk to whoever owns TURN, which is (I believe) behave.

> One small comment: shouldn't the response also contain the value of the
> REALM parameter?

I assume that this is intended to use the short term credential
mechanism.  No need to use long term credentials in this case.

From roman@telurix.com  Mon Jul  8 10:02:22 2013
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Mon, Jul 8, 2013 at 12:41 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
> What would be useful would be for people to
> make a list of these use cases and the extensions which would
> enable them.
>
>
This would be ability to modify all the information present in SDP. So the
rule of thumb should be that if you can remove SDP from the API call, then
you have done it right.

For me personally I would like to see:
1. Specify codec list including relative priorities, framerate, number of
channels, and per codec parameters (like useinbandfec for OPUS).
2. Specify ptimes and maxptime per media stream and per codec
3. Specify bandwidth per media stream
4. If we support SDES SRTP, specify encryption profiles
_____________
Roman Shpount

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<div class=3D"gmail_quote">On Mon, Jul 8, 2013 at 12:41 PM, Eric Rescorla <=
span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@=
rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<div>What would be useful would be for people to</div><div>make a list of t=
hese use cases and the extensions which would</div><div>enable them.</div>

<div><br></div></div></div></div></blockquote><div><br></div><div>This woul=
d be ability to modify all the information present in SDP. So the rule of t=
humb should be that if you can remove SDP from the API call, then you have =
done it right.</div>
<div><br></div><div>For me personally I would like to see:</div><div>1. Spe=
cify codec list including relative priorities, framerate, number of channel=
s, and per codec parameters (like=A0<span style=3D"font-size:1em">useinband=
fec for OPUS).</span></div>
<div>2. Specify ptimes and maxptime per media stream and per codec</div><di=
v>3. Specify bandwidth per media stream</div><div>4. If we support SDES SRT=
P, specify encryption profiles</div><div>_____________<br>Roman Shpount</di=
v>
<div>=A0</div></div>

--001a11c3669e611cc204e1030124--

From ibc@aliax.net  Mon Jul  8 10:07:19 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 8 Jul 2013 19:06:52 +0200
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/8 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> And, I don't understand why there would be a debate about this. There
> are obviously many who want to define APIs (or constraints) that allow
> most use-cases to be met without having to modify the SDP. Why would
> anyone have anything against that? If there is anyone who really wants
> to modify the SDP instead (I have no clue why, but anyway) they can
> still do that.

This sounds to me like "SDP will be here forever because it was talked
about two years ago, but let us improve it by adding some more API
methods so there is no need to parse/mangle it".

Just to clarify:

- Some of us don't want a better API for the currently proposed SDP-based s=
pec.

- SDP must be parsed/mangled anyway if you want a JS WebRTC app that
talks Jingle XEP 0167, so there is no "API improvement" that would
avoid such a pain.

- It's not just SDP but also the mandate of SDP O/A. What I think is
that it is not well understood that SDP O/A is an artificial constrain
introduced in WebRTC just because SIP phones do that ***at application
level*** (so you are mandating that *any* WebRTC application in the
future must play like-SIP rules and SDP O/A model, which is insane). A
better and more complete API for the current spec DOES NOT avoid this
artificial limitation.


In short: some of us DO NOT want a better JS API for the current
opaque-unmanageable-SDP-blob based "API". Period. We strongly consider
it can be done really better by throwing out SDP and SDP O/A and by
providing a *real* JS Objects based API.




--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/8 Jim Barnett <Jim.Barnett@genesyslab.com>:
> As I recall the consensus, developers are supposed to be able to modify t=
he
> SDP, but we needed to specify what parts of the SDP could be modified.  I=
n
> other words, there might be some limits to what could be modified, but we
> haven=E2=80=99t yet determined/agreed on what they are.

At this rate, wasting so long time on extending SDP to fullfil WebRTC
requirements, it would be done within the next 3-4 years. SDP is the
main issue delaying WebRTC 1.0.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2013/5/9 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> Whatever signaling is used to indicate bundling usage in SDP in RTCWEB
> should be consistent with what is used in clue. It would be unpleasant if=
 we
> had to have a translator in the web server that must translate one SDP
> notation to another.

Hi,

Give a powerfiul real JavaScript Object based API (instead of an
opaque SDP blob) and let the web developer decide whether to construct
**via JavaScript** a CLUE compatible SDP or not (just if he needs
that), and we all will win.

In contrast, by mandating plain SDP (just because SIP uses it) we have
to have a translator in the web server that must translate plain-SDP
to XML-SDP (in Jingle XEP 0167 deployments), or build the best SDP
parser at JS level to parse the SDP blob created by the browser and
convert it into XML, and then convert it back into plain SDP.

Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 8 Jul 2013 14:02:10 -0400
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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
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On Mon, Jul 8, 2013 at 12:48 PM, Martin Thomson <martin.thomson@gmail.com>w=
rote:

> Two separate replies...
>
> On 8 July 2013 05:23, Simon Perreault <simon.perreault@viagenie.ca> wrote=
:
> > Le 2013-07-08 06:24, Justin Uberti a =C3=A9crit :
> >> Just uploaded a 00 version of a spec for requesting time-limited TURN
> >> credentials for WebRTC apps. Would like to get 10 minutes of agenda ti=
me
> >> to present this in Berlin.
>
> Justin, is this the right working group for this draft?  Presumably,
> you want to talk to whoever owns TURN, which is (I believe) behave.
>

WebRTC applications are the primary beneficiaries of this proposal, which
is why I submitted it here. I can certainly send it to BEHAVE if that's a
better fit. (Depending on how closely this references
draft-reddy-behave-turn-auth, that might be a good indication of where this
should end up.)

>
> > One small comment: shouldn't the response also contain the value of the
> > REALM parameter?
>
> I assume that this is intended to use the short term credential
> mechanism.  No need to use long term credentials in this case.
>

RFC 5766 mandates the use of the long-term credential mechanism. One of the
goals of this proposal is to work with existing TURN servers, so it also
uses the long-term credential mechanism, the key point being that the
vended credentials have finite lifetimes.


> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 12:48 PM, Martin Thomson <span dir=3D"ltr">&=
lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.tho=
mson@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Two separate replies...<br>
<div class=3D"im"><br>
On 8 July 2013 05:23, Simon Perreault &lt;<a href=3D"mailto:simon.perreault=
@viagenie.ca">simon.perreault@viagenie.ca</a>&gt; wrote:<br>
&gt; Le 2013-07-08 06:24, Justin Uberti a =C3=A9crit :<br>
&gt;&gt; Just uploaded a 00 version of a spec for requesting time-limited T=
URN<br>
&gt;&gt; credentials for WebRTC apps. Would like to get 10 minutes of agend=
a time<br>
&gt;&gt; to present this in Berlin.<br>
<br>
</div>Justin, is this the right working group for this draft? =C2=A0Presuma=
bly,<br>
you want to talk to whoever owns TURN, which is (I believe) behave.<br></bl=
ockquote><div><br></div><div>WebRTC applications are the primary beneficiar=
ies of this proposal, which is why I submitted it here. I can certainly sen=
d it to BEHAVE if that&#39;s a better fit. (Depending on how closely this r=
eferences draft-reddy-behave-turn-auth, that might be a good indication of =
where this should end up.)</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div class=3D"im"><br>
&gt; One small comment: shouldn&#39;t the response also contain the value o=
f the<br>
&gt; REALM parameter?<br>
<br>
</div>I assume that this is intended to use the short term credential<br>
mechanism. =C2=A0No need to use long term credentials in this case.<br></bl=
ockquote><div><br></div><div>RFC 5766 mandates the use of the long-term cre=
dential mechanism. One of the goals of this proposal is to work with existi=
ng TURN servers, so it also uses the long-term credential mechanism, the ke=
y point being that the vended credentials have finite lifetimes.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">
<div class=3D""><div class=3D"h5">_________________________________________=
______<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 8 Jul 2013 14:12:21 -0400
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On Mon, Jul 8, 2013 at 8:23 AM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Le 2013-07-08 06:24, Justin Uberti a =C3=A9crit :
> > Just uploaded a 00 version of a spec for requesting time-limited TURN
> > credentials for WebRTC apps. Would like to get 10 minutes of agenda tim=
e
> > to present this in Berlin.
>
> Cool.
>
> One small comment: shouldn't the response also contain the value of the
> REALM parameter?
>

It could carry the REALM parameter, but if so, it would only be used to
verify the REALM that was supplied by the server as part of the ALLOCATE
challenge. Did you have something else in mind?

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 8:23 AM, Simon Perreault <span dir=3D"ltr">&=
lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.p=
erreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2013-07-08 06:24, Justin Uberti a =C3=A9c=
rit :<br>
<div class=3D"im">&gt; Just uploaded a 00 version of a spec for requesting =
time-limited TURN<br>
&gt; credentials for WebRTC apps. Would like to get 10 minutes of agenda ti=
me<br>
&gt; to present this in Berlin.<br>
<br>
</div>Cool.<br>
<br>
One small comment: shouldn&#39;t the response also contain the value of the=
<br>
REALM parameter?<br></blockquote><div><br></div><div>It could carry the REA=
LM parameter, but if so, it would only be used to verify the REALM that was=
 supplied by the server as part of the ALLOCATE challenge. Did you have som=
ething else in mind?</div>

</div></div></div>

--047d7ba97b9485f1a704e103fd2d--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 8 Jul 2013 14:15:55 -0400
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On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi Perumal (mperumal) <
mperumal@cisco.com> wrote:

>  Hi Justin,****
>
> ** **
>
> A few quick comments:****
>
> 1) The primary advantage of the proposed mechanism seems not requiring any
> interaction between the web service and the TURN service in order for the
> TURN service to grant TURN credentials in the HTTP response -- this absence
> of interaction isn't evident on a first read. A diagram showing the client,
> web service, TURN service and the messages exchanged would be helpful.
>

This is mentioned in the introduction, but agree a diagram would be
helpful.

> ****
>
> ** **
>
> 2) ****
>
> |If desired, the TURN server can optionally verify that the parsed****
>
> |user id value corresponds to a currently valid user of an external****
>
> |service (e.g. is currently logged in to the web app that is making****
>
> |use of TURN).  This requires proprietary communication between the****
>
> |TURN server and external service on each ALLOCATE request, so this****
>
> |usage is not recommended for typical applications.  If this external****
>
> |verification fails, it SHOULD reject the request with a 401****
>
> |(Unauthorized) error.****
>
> ** **
>
> Was the intention of putting "not recommended" having a normative
> statement? If not, it would be better to change it to "no needed".
>

I was waffling on this - I think I will just make it "not needed", and
leave this decision up to the implementor.

> ****
>
> ** **
>
> 3) There is no text describing how the timestamp encoded in the UNSERNAME
> attribute of the ALLOCAE requested could be protected.
>

In the HTTP Interactions section, I mention that the password used for the
MESSAGE-INTEGRITY is a digest of the username, but I can make this more
explicit.

> ****
>
> ** **
>
> 4) draft-reddy-behave-turn-auth describes the issues with TURN
> authentication and draft-uberti-rtcweb-turn-rest looks like one possible
> solution. Looks both could reference each other.
>

Agreed - hadn't seen that draft before since it wasn't in rtcweb.

> ****
>
> ** **
>
> Muthu****
>
> ** **
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Justin Uberti
> *Sent:* Monday, July 08, 2013 9:55 AM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Fwd: New Version Notification for
> draft-uberti-rtcweb-turn-rest-00.txt****
>
> ** **
>
> Just uploaded a 00 version of a spec for requesting time-limited TURN
> credentials for WebRTC apps. Would like to get 10 minutes of agenda time to
> present this in Berlin.****
>
> ** **
>
> ---------- Forwarded message ----------
> From: <internet-drafts@ietf.org>
> Date: Mon, Jul 8, 2013 at 12:15 AM
> Subject: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
> To: Justin Uberti <justin@uberti.name>
>
>
>
> A new version of I-D, draft-uberti-rtcweb-turn-rest-00.txt
> has been successfully submitted by Justin Uberti and posted to the
> IETF repository.
>
> Filename:        draft-uberti-rtcweb-turn-rest
> Revision:        00
> Title:           A REST API For Access To TURN Services
> Creation date:   2013-07-08
> Group:           Individual Submission
> Number of pages: 7
> URL:
> http://www.ietf.org/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt
> Status:
> http://datatracker.ietf.org/doc/draft-uberti-rtcweb-turn-rest
> Htmlized:
> http://tools.ietf.org/html/draft-uberti-rtcweb-turn-rest-00
>
>
> Abstract:
>    This document describes a proposed standard REST API for obtaining
>    access to TURN services via ephemeral (i.e. time-limited)
>    credentials.  These credentials are vended by a web service over
>    HTTP, and then supplied to and checked by a TURN server using the
>    standard TURN protocol.  The usage of ephemeral credentials ensures
>    that access to the TURN server can be controlled even if the
>    credentials can be discovered by the user, as is the case in WebRTC
>    where TURN credentials must be specified in Javascript.
>
>
>
>
> The IETF Secretariat****
>
> ** **
>
> ** **
>

--f46d0442727042a0ed04e1040afa
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi Perumal (mperumal)=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_bla=
nk">mperumal@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">Hi Justin,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">A few quick comments:<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">1) The primary advantage of the proposed=
 mechanism seems not requiring any interaction between the web service and =
the TURN service in order for the TURN service to grant
 TURN credentials in the HTTP response -- this absence of interaction isn&#=
39;t evident on a first read. A diagram showing the client, web service, TU=
RN service and the messages exchanged would be helpful.</span></p></div>

</div></blockquote><div><br></div><div>This is mentioned in the introductio=
n, but agree a diagram would be helpful.=C2=A0</div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&qu=
ot;serif&quot;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">2)
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|If desired, the TURN server can optiona=
lly verify that the parsed<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|user id value corresponds to a currentl=
y valid user of an external<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|service (e.g. is currently logged in to=
 the web app that is making<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|use of TURN).=C2=A0 This requires propr=
ietary communication between the<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|TURN server and external service on eac=
h ALLOCATE request, so this<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|usage is not recommended for typical ap=
plications.=C2=A0 If this external<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|verification fails, it SHOULD reject th=
e request with a 401<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|(Unauthorized) error.<u></u><u></u></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">Was the intention of putting &quot;not r=
ecommended&quot; having a normative statement? If not, it would be better t=
o change it to &quot;no needed&quot;.</span></p>

</div></div></blockquote><div><br></div><div>I was waffling on this - I thi=
nk I will just make it &quot;not needed&quot;, and leave this decision up t=
o the implementor.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&qu=
ot;serif&quot;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">3) There is no text describing how the t=
imestamp encoded in the UNSERNAME attribute of the ALLOCAE requested could =
be protected.</span></p>

</div></div></blockquote><div>=C2=A0<br></div><div>In the HTTP Interactions=
 section, I mention that the password used for the MESSAGE-INTEGRITY is a d=
igest of the username, but I can make this more explicit.</div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&qu=
ot;serif&quot;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">4) draft-reddy-behave-turn-auth describe=
s the issues with TURN authentication and draft-uberti-rtcweb-turn-rest loo=
ks like one possible solution. Looks both could reference
 each other.</span></p></div></div></blockquote><div><br></div><div>Agreed =
- hadn&#39;t seen that draft before since it wasn&#39;t in rtcweb.=C2=A0</d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&qu=
ot;serif&quot;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">Muthu<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><u></u>=C2=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href=
=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank"=
>rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Justin Uberti<br>
<b>Sent:</b> Monday, July 08, 2013 9:55 AM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> [rtcweb] Fwd: New Version Notification for draft-uberti-rtc=
web-turn-rest-00.txt<u></u><u></u></span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Just uploaded a 00 version of a spec for requesting =
time-limited TURN credentials for WebRTC apps. Would like to get 10 minutes=
 of agenda time to present this in Berlin.<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">---------- Forwarded =
message ----------<br>
From: &lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">int=
ernet-drafts@ietf.org</a>&gt;<br>
Date: Mon, Jul 8, 2013 at 12:15 AM<br>
Subject: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt<=
br>
To: Justin Uberti &lt;<a href=3D"mailto:justin@uberti.name" target=3D"_blan=
k">justin@uberti.name</a>&gt;<br>
<br>
<br>
<br>
A new version of I-D, draft-uberti-rtcweb-turn-rest-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Filename: =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-uberti-rtcweb-turn-rest<br>
Revision: =C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 A REST API For Access To TURN Ser=
vices<br>
Creation date: =C2=A0 2013-07-08<br>
Group: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Number of pages: 7<br>
URL: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.o=
rg/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt" target=3D"_blank">
http://www.ietf.org/internet-drafts/draft-uberti-rtcweb-turn-rest-00.txt</a=
><br>
Status: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://datatracker.iet=
f.org/doc/draft-uberti-rtcweb-turn-rest" target=3D"_blank">http://datatrack=
er.ietf.org/doc/draft-uberti-rtcweb-turn-rest</a><br>
Htmlized: =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/=
draft-uberti-rtcweb-turn-rest-00" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-uberti-rtcweb-turn-rest-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes a proposed standard REST API for obtai=
ning<br>
=C2=A0 =C2=A0access to TURN services via ephemeral (i.e. time-limited)<br>
=C2=A0 =C2=A0credentials. =C2=A0These credentials are vended by a web servi=
ce over<br>
=C2=A0 =C2=A0HTTP, and then supplied to and checked by a TURN server using =
the<br>
=C2=A0 =C2=A0standard TURN protocol. =C2=A0The usage of ephemeral credentia=
ls ensures<br>
=C2=A0 =C2=A0that access to the TURN server can be controlled even if the<b=
r>
=C2=A0 =C2=A0credentials can be discovered by the user, as is the case in W=
ebRTC<br>
=C2=A0 =C2=A0where TURN credentials must be specified in Javascript.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br></div></div>

--f46d0442727042a0ed04e1040afa--

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On 8 July 2013 11:02, Justin Uberti <juberti@google.com> wrote:
> RFC 5766 mandates the use of the long-term credential mechanism. One of the
> goals of this proposal is to work with existing TURN servers, so it also
> uses the long-term credential mechanism, the key point being that the vended
> credentials have finite lifetimes.

You could update 5766 to remove this constraint (I forgot about that
bit)...and the extra round trip required for the challenge.

Or you could provide the client with a realm and nonce in the
response, but that seems like a little too much.

From simon.perreault@viagenie.ca  Mon Jul  8 11:41:25 2013
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
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Le 2013-07-08 20:12, Justin Uberti a Ã©crit :
>     One small comment: shouldn't the response also contain the value of the
>     REALM parameter?
>
>
> It could carry the REALM parameter, but if so, it would only be used to
> verify the REALM that was supplied by the server as part of the ALLOCATE
> challenge. Did you have something else in mind?

Either that or the text needs to say that the REALM sent by the server 
is to be ignored by the client.

I have no opinion at this point, really, other than the text needs to 
say something about REALM.

Simon
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Date: Mon, 8 Jul 2013 16:09:20 -0400
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From: Justin Uberti <juberti@google.com>
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On Jul 8, 2013 2:16 PM, "Martin Thomson" <martin.thomson@gmail.com> wrote:
>
> On 8 July 2013 11:02, Justin Uberti <juberti@google.com> wrote:
> > RFC 5766 mandates the use of the long-term credential mechanism. One of
the
> > goals of this proposal is to work with existing TURN servers, so it also
> > uses the long-term credential mechanism, the key point being that the
vended
> > credentials have finite lifetimes.
>
> You could update 5766 to remove this constraint (I forgot about that
> bit)...and the extra round trip required for the challenge.
>
> Or you could provide the client with a realm and nonce in the
> response, but that seems like a little too much.

The issue with using short term credentials, without a nonce, is the
possibility of replay attacks by an eavesdropper.

Passing realm and nonce solves this but is problematic because nonces need
to be per-allocation, so you'd need some sort of master nonce that could be
used for nonce generation. Plus the Rtciceserver API would need to be
expanded.

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<p dir="ltr"><br>
On Jul 8, 2013 2:16 PM, &quot;Martin Thomson&quot; &lt;<a href="mailto:martin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On 8 July 2013 11:02, Justin Uberti &lt;<a href="mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt; &gt; RFC 5766 mandates the use of the long-term credential mechanism. One of the<br>
&gt; &gt; goals of this proposal is to work with existing TURN servers, so it also<br>
&gt; &gt; uses the long-term credential mechanism, the key point being that the vended<br>
&gt; &gt; credentials have finite lifetimes.<br>
&gt;<br>
&gt; You could update 5766 to remove this constraint (I forgot about that<br>
&gt; bit)...and the extra round trip required for the challenge.<br>
&gt;<br>
&gt; Or you could provide the client with a realm and nonce in the<br>
&gt; response, but that seems like a little too much.</p>
<p dir="ltr">The issue with using short term credentials, without a nonce, is the possibility of replay attacks by an eavesdropper.</p>
<p dir="ltr">Passing realm and nonce solves this but is problematic because nonces need to be per-allocation, so you&#39;d need some sort of master nonce that could be used for nonce generation. Plus the Rtciceserver API would need to be expanded.<br>
</p>

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On 7/8/13 4:38 PM, Peter Thatcher wrote:=0A=
>=0A=
>=0A=
>=0A=
> On Mon, Jul 8, 2013 at 5:11 AM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     On 7/7/13 11:14 PM, Roman Shpount wrote:=0A=
>      >=0A=
>      > On Sat, Jul 6, 2013 at 2:33 AM, Stefan H=E5kansson LK=0A=
>      > <stefan.lk.hakansson@ericsson.com=0A=
>     <mailto:stefan.lk.hakansson@ericsson.com>=0A=
>      > <mailto:stefan.lk.hakansson@ericsson.com=0A=
>     <mailto:stefan.lk.hakansson@ericsson.com>>> wrote:=0A=
>      >=0A=
>      >     On 7/3/13 11:37 PM, Eric Rescorla wrote:=0A=
>      >      >     I'm glad to be wrong here.  Is the phrase "you=0A=
>     shouldn't have to=0A=
>      >      >     modify the SDP but rather there should be API points=0A=
>     to cover=0A=
>      >     most=0A=
>      >      >     of the use cases"  the consensus of the working group?=
=0A=
>      >      >=0A=
>      >      >=0A=
>      >      > I thought it was, but I'm not the chair, so maybe you=0A=
>     could ask=0A=
>      >     Harald or=0A=
>      >      > Stefan.=0A=
>      >=0A=
>      >     I definitely think this is the consensus of the working group.=
=0A=
>      >=0A=
>      >=0A=
>      > Can you point out when exactly this consensus was reached? This=0A=
>     is news=0A=
>      > to me, but I definitely could have missed something.=0A=
>=0A=
>     No I can't really. To be clear, no consensus call (or similar) has be=
en=0A=
>     held. But every time this is discussed, what I hear is people agreein=
g=0A=
>     and no objections.  It was discussed a bit, e.g., at the f2f at TPAC=
=0A=
>     last=0A=
>=0A=
>     year, if you read the minutes [1] you will find some traces of that=
=0A=
>     discussion.=0A=
>=0A=
>=0A=
> No objections at all?  That is rather surprising to me.  I asked for=0A=
> some feedback the API in the spreadsheet from web developers, at there=0A=
> many objections from them about this part of the API.  A majority did=0A=
> say it was good enough for simple use cases, but he objections were=0A=
> still clearly non-zero.=0A=
=0A=
I think we're discussing different things. I'm referring to adding API =0A=
methods that would allow meeting most use-cases without SDP mangling - =0A=
under the presumption that SDP O/A would be used as part of the design. =0A=
And that is what we have been assuming, and the WebRTC WG confirmed this =
=0A=
last year when it was up for discussion.=0A=
=0A=
What I see being most discussed now is the role of SDP O/A - not that it =
=0A=
would not be helpful to have API methods that allowed meeting use-cases =0A=
without SDP mangling if SDP O/A stays.=0A=
=0A=
>=0A=
> I think perhaps the web developers are not well represented at TPAC.=0A=
>   But I could be wrong.  Perhaps they are but changed their mind.  Or=0A=
> maybe they're objecting on the IETF mailing list when they should object=
=0A=
> on the IETF mailing list.   Or maybe the spreadsheet has a sample bias,=
=0A=
> while TPAC is the true sample.   I don't know, but it seems like=0A=
> something is going on, and I'd like to know what it is.=0A=
>=0A=
> Stefan, what do you think?  What could explain the apparent=0A=
> contradiction between objections on the mailing list vs. no objections f2=
f?=0A=
=0A=
Again, I think that we are to some extent discussing different things =0A=
(see above).=0A=
=0A=
But I can only speculate. There is not a 100% overlap between people who =
=0A=
have raised concerns on the rtcweb list and the people usually active in =
=0A=
the W3C WebRTC WG, this can be a reason.=0A=
=0A=
>=0A=
>=0A=
> Also, on discussions about the API, should those who want to propose=0A=
> additions (such as the NoPlan JS API I proposed) and other comments put=
=0A=
> them on the W3C mailing list instead of the IETF one?  Is everyone=0A=
> barking up the wrong tree?=0A=
=0A=
I think API proposals should be taken to the W3C list.=0A=
=0A=
>=0A=
> Thanks in advance for your input.=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>     And, I don't understand why there would be a debate about this. There=
=0A=
>     are obviously many who want to define APIs (or constraints) that allo=
w=0A=
>     most use-cases to be met without having to modify the SDP.  Why would=
=0A=
>=0A=
>     anyone have anything against that?=0A=
>=0A=
>=0A=
> I think your question actually has two questions:=0A=
>=0A=
> 1.  Is anyone against making additions to the API to avoid SDP munging?=
=0A=
>=0A=
> 2.  Why are they against additions to the API to avoid SDP munging?=0A=
>=0A=
>=0A=
> My personal experience for #1 is that there are some notable people in=0A=
> the WG who are very much against making additions to the API to avoid=0A=
> SDP munging.  Some of them are rather upset with me for proposing a few=
=0A=
> minor additions to the API (the NoPlan JS API) precisely because it=0A=
> allows the JS to go around the SDP in many cases, and they don't like=0A=
> that.  So, from my experience, it's clear that the answer to #1 is "yes,=
=0A=
> many do".=0A=
=0A=
Just to be clear, what I meant in my comment was APIs that made the =0A=
browser create an SDP that is suitable for the use-case at hand, not an =0A=
API to go around SDP.=0A=
=0A=
>=0A=
> And since many object to such additions, and others want such additions,=
=0A=
> you end up with debate.=0A=
>=0A=
>=0A=
> How about #2?  Why do they object?  My experience is that they view any=
=0A=
> additions as a threat to API as a threat to the current API, which means=
=0A=
> it's a threat to the current WG progress, which means it's a threat to=0A=
> getting done.  They are very sensitive to such threats, and so object to=
=0A=
> any additions or changes that would allow JS to avoid using SDP.=0A=
=0A=
I must admit I am personally also a bit afraid of WG progress if we =0A=
change too much. If your proposed API addition would be "too much" is =0A=
another question.=0A=
=0A=
> Also,=0A=
> I think they don't want to ways of doing things, so SDP must remain the=
=0A=
> one true way.=0A=
>=0A=
>=0A=
> But that's just my personal experience and perspective.  I could just be=
=0A=
> seeing it wrong, or maybe the only person experiencing this.=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>     I If there is anyone who really wants=0A=
>     to modify the SDP instead (I have no clue why, but anyway) they can=
=0A=
>     still do that.=0A=
>=0A=
>=0A=
> Well, no one is going to hit themselves in the head with a hammer if=0A=
> they don't have to, but there are many uses cases that currently be done=
=0A=
> only by SDP munging, and I don't see any API additions on the horizon=0A=
> that will help those uses cases.    So, until we can agree on some API=0A=
> additions that help SDP munging, many web developers are going to have=0A=
> to keep swinging that hammer, no matter how much it hurts.=0A=
=0A=
I think I need to do some reading up because the info must all be there =0A=
among all the mails, but it would really be helpful to have a list of =0A=
things people would like to do, and that currently require SDP mangling.=0A=
=0A=
>=0A=
> But clearly, if you gave them an API that didn't require such pain, they=
=0A=
> would use it.=0A=
>=0A=
>=0A=
>     [1] http://lists.w3.org/Archives/Public/public-webrtc/2012Nov/0072.ht=
ml=0A=
>=0A=
>      > _____________=0A=
>      > Roman Shpount=0A=
>      >=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
=0A=

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/8/13 5:22 PM, Roman Shpount wrote:=0A=
=0A=
> So, was there ever a consensus that "developers MUST NOT=0A=
> touch SDP"?=0A=
=0A=
No, developers are free to touch the SDP (and probably must in certain =0A=
cases).=0A=
=0A=
>=0A=
> _____________=0A=
> Roman Shpount=0A=
=0A=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/8 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
>> So, was there ever a consensus that "developers MUST NOT
>> touch SDP"?
>
> No, developers are free to touch the SDP (and probably must in certain
> cases).


Do you feel how much painful is "touching a browser generated SDP"? To be c=
lear:

- JS requests "A" to browser.
- Browser returns "A" (in the form of unmanageable blob).
- JS modifies "A" to send "A-bis" in the wire.
- Remote peer receives "A-bis" and replies "B-bis".
- Both JS app are lying to their respective  browsers to get the
desired behavior.

This is really painful, really.

In the other side, as I explained in other thread (may be in this
one?) adding some methods to the current API (to avoid playing with
the SDP) is not the way to go. A specification in which the JS app
does not know what he is sending in-the-wire (a blob generated by the
browser) is doomed to failure.

Please remember that mandating SDP (plain SDP) means that no other
media signaling protocol can be implemented in future WebRTC apps.

SIP phones have *fixed* code and *fixed* logic. This is no longer true
in the Web.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From martin.thomson@gmail.com  Mon Jul  8 14:43:38 2013
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
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On 8 July 2013 13:09, Justin Uberti <juberti@google.com> wrote:
> The issue with using short term credentials, without a nonce, is the
> possibility of replay attacks by an eavesdropper.

It is no less vulnerable than having the long term credential set
(username, nonce, realm, and password) overheard.  Assuming that the
lifetime of the password is the same in both cases.  In either case,
the link that the eavesdropper is required to attack is the HTTP link.

> Passing realm and nonce solves this [...]

I was suggesting that since you have spent some very expensive
round-trips getting this information, there are no advantages in
spending yet another round-trip on a challenge.  I don't think that
passing realm and nonce is a good idea in practice - it creates a
tighter coupling between this new thing and the TURN server.

In practice, a master nonce is not quite what you need, you need a
nonce-generator function, or a line to the TURN server whereby you
query for every request you get.  The former imposes too-strong
constraints on implementations, the latter renders much of the
advantages of something like this moot.

From adam@nostrum.com  Mon Jul  8 14:51:10 2013
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On 7/7/13 23:24, Justin Uberti wrote:
> Just uploaded a 00 version of a spec for requesting time-limited TURN 
> credentials for WebRTC apps. Would like to get 10 minutes of agenda 
> time to present this in Berlin.

Thanks for getting ball rolling on this, Justin. I agree that this is an 
important mechanism to have defined.

I only find one issue in my initial read through the document: why are 
we proposing the use of HTTP POST for stateless, idempotent information 
retrieval? The key benefit of this approach is that it doesn't actually 
create state in the network. That's a much better fit for GET.

Compare RFC 2616, sections 9.3 and 9.5, and see if you don't agree.

/a

From martin.thomson@gmail.com  Mon Jul  8 14:57:15 2013
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On 8 July 2013 14:50, Adam Roach <adam@nostrum.com> wrote:
> I only find one issue in my initial read through the document: why are we
> proposing the use of HTTP POST for stateless, idempotent information
> retrieval? The key benefit of this approach is that it doesn't actually
> create state in the network. That's a much better fit for GET.

That's an interesting question.  From one perspective, this is not
idempotent, nor is it safe.  It creates a new resource: a new
username/password pair with a defined validity.  You could even
identify this "resource" with a URI.  Imagine what a DELETE on that
resource would do...

But it's equally valid to try to pretend otherwise and use GET.  After
all, most servers won't actually be allocating any state for these
things.  And I can't think of an actual, practical use for DELETE, so
that's just a silly feature.

--Martin

p.s., I was going to let this slide, but this design is not RESTful,
no sense in pretending like it is.

From adam@nostrum.com  Mon Jul  8 15:06:17 2013
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On 7/8/13 16:57, Martin Thomson wrote:
> On 8 July 2013 14:50, Adam Roach <adam@nostrum.com> wrote:
>> I only find one issue in my initial read through the document: why are we
>> proposing the use of HTTP POST for stateless, idempotent information
>> retrieval? The key benefit of this approach is that it doesn't actually
>> create state in the network. That's a much better fit for GET.
> That's an interesting question.  From one perspective, this is not
> idempotent, nor is it safe.  It creates a new resource: a new
> username/password pair with a defined validity.

That's like claiming that every time I hit cnn.com, it creates a new 
resource, since the organization of the articles and ads changes as a 
function of time (new articles are added) and local cookies (encoding my 
section preferences). I think both are kind of a tortured interpretation 
of resource creation.


> But it's equally valid to try to pretend otherwise and use GET.  After
> all, most servers won't actually be allocating any state for these
> things.

I hope none of them do. It's kind of the main point of the whole draft.

> p.s., I was going to let this slide, but this design is not RESTful,
> no sense in pretending like it is.

I'm not sure I agree with your assessment. Which of these six 
constraints do you believe are violated?

http://en.wikipedia.org/wiki/Representational_state_transfer#Constraints

/a

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On 8 July 2013 15:05, Adam Roach <adam@nostrum.com> wrote:
>> But it's equally valid to try to pretend otherwise and use GET.  After
>> all, most servers won't actually be allocating any state for these
>> things.
>
> I hope none of them do. It's kind of the main point of the whole draft.

I expect all of them to do so, but not hold clients responsible for
that, so GET is still OK.

>> p.s., I was going to let this slide, but this design is not RESTful,
>> no sense in pretending like it is.
>
> I'm not sure I agree with your assessment. Which of these six constraints do
> you believe are violated?
>
> http://en.wikipedia.org/wiki/Representational_state_transfer#Constraints

Will you think worse of me if I avoid opening this particular can of
worms?  I regret even mentioning it.

I've learned not to use the term RESTful, I should have known better.

From juberti@google.com  Mon Jul  8 17:51:20 2013
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From: Justin Uberti <juberti@google.com>
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On Jul 8, 2013 5:43 PM, "Martin Thomson" <martin.thomson@gmail.com> wrote:
>
> On 8 July 2013 13:09, Justin Uberti <juberti@google.com> wrote:
> > The issue with using short term credentials, without a nonce, is the
> > possibility of replay attacks by an eavesdropper.
>
> It is no less vulnerable than having the long term credential set
> (username, nonce, realm, and password) overheard.  Assuming that the
> lifetime of the password is the same in both cases.  In either case,
> the link that the eavesdropper is required to attack is the HTTP link.

I don't think this is true. In the short term case, with no nonce, the
packet can be replayed verbatim.

> > Passing realm and nonce solves this [...]
>
> I was suggesting that since you have spent some very expensive
> round-trips getting this information, there are no advantages in
> spending yet another round-trip on a challenge.  I don't think that
> passing realm and nonce is a good idea in practice - it creates a
> tighter coupling between this new thing and the TURN server.
>
> In practice, a master nonce is not quite what you need, you need a
> nonce-generator function, or a line to the TURN server whereby you
> query for every request you get.  The former imposes too-strong
> constraints on implementations, the latter renders much of the
> advantages of something like this moot.

Right, the master nonce I suggested would be used to generate regular
nonces. But it's not a good idea anyway.

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<p dir=3D"ltr"><br>
On Jul 8, 2013 5:43 PM, &quot;Martin Thomson&quot; &lt;<a href=3D"mailto:ma=
rtin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On 8 July 2013 13:09, Justin Uberti &lt;<a href=3D"mailto:juberti@goog=
le.com">juberti@google.com</a>&gt; wrote:<br>
&gt; &gt; The issue with using short term credentials, without a nonce, is =
the<br>
&gt; &gt; possibility of replay attacks by an eavesdropper.<br>
&gt;<br>
&gt; It is no less vulnerable than having the long term credential set<br>
&gt; (username, nonce, realm, and password) overheard. =C2=A0Assuming that =
the<br>
&gt; lifetime of the password is the same in both cases. =C2=A0In either ca=
se,<br>
&gt; the link that the eavesdropper is required to attack is the HTTP link.=
</p>
<p dir=3D"ltr">I don&#39;t think this is true. In the short term case, with=
 no nonce, the packet can be replayed verbatim.</p>
<p dir=3D"ltr">&gt; &gt; Passing realm and nonce solves this [...]<br>
&gt;<br>
&gt; I was suggesting that since you have spent some very expensive<br>
&gt; round-trips getting this information, there are no advantages in<br>
&gt; spending yet another round-trip on a challenge. =C2=A0I don&#39;t thin=
k that<br>
&gt; passing realm and nonce is a good idea in practice - it creates a<br>
&gt; tighter coupling between this new thing and the TURN server.<br>
&gt;<br>
&gt; In practice, a master nonce is not quite what you need, you need a<br>
&gt; nonce-generator function, or a line to the TURN server whereby you<br>
&gt; query for every request you get. =C2=A0The former imposes too-strong<b=
r>
&gt; constraints on implementations, the latter renders much of the<br>
&gt; advantages of something like this moot.</p>
<p dir=3D"ltr">Right, the master nonce I suggested would be used to generat=
e regular nonces. But it&#39;s not a good idea anyway.<br>
</p>

--089e01493e00e35a5304e1098e57--

From silviapfeiffer1@gmail.com  Mon Jul  8 20:03:35 2013
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Tue, 9 Jul 2013 13:03:12 +1000
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Tue, Jul 9, 2013 at 12:38 AM, Peter Thatcher <pthatcher@google.com> wrote:
>
> No objections at all?  That is rather surprising to me.  I asked for some
> feedback the API in the spreadsheet from web developers, at there many
> objections from them about this part of the API.  A majority did say it was
> good enough for simple use cases, but he objections were still clearly
> non-zero.
>
> I think perhaps the web developers are not well represented at TPAC.  But I
> could be wrong.  Perhaps they are but changed their mind.  Or maybe they're
> objecting on the IETF mailing list when they should object on the IETF
> mailing list.   Or maybe the spreadsheet has a sample bias, while TPAC is
> the true sample.   I don't know, but it seems like something is going on,
> and I'd like to know what it is.
>
> Stefan, what do you think?  What could explain the apparent contradiction
> between objections on the mailing list vs. no objections f2f?

I'm not Stefan, but I have a theory.

I think there are two groups objecting and there are two things to
object against.

Here's what I'm reading...

Web Developers really just want to be given functionality. They can
build their own nice API on top of what browsers implement (jquery did
a marvellous job of that). But they want functionality. Seeing as some
things are not currently possible with SDP and O/A, they either want
to extend those to allow their functionality, or - in particular where
SDP & O/A are overkill - want an option to get out of using SDP & O/A
and use something else.

Telco engineers want their existing devices to be interoperable with
browsers as well as the ability to develop new devices in the future
with extended functionality. So, on behalf of the first goal, they
object to including functionality into SDP that other devices don't
already support, and on behalf of the second goal, they are torn
between extending SDP & O/A and starting new.

As to why you haven't seen anyone object at the TPAC (did you really
mean TPAC, btw? The next TPAC meeting is only in November) - maybe
none of the people of these two groups were present at last year's
TPAC? Or they didn't understand the consequences yet.... Also, it's
easy to object on a mailing list, while in a meeting room, you'd have
to provide an alternative and it's really hard to come up with such an
alternative IMHO.

I certainly want a simple API, but I am not sure whether what we
currently have is already the simplest possible (given current
circumstances), nor am I sure whether it wouldn't just be possible to
build a simple one on top the jquery way, which would be sufficient
for now.

HTH,
Silvia.

From martin.thomson@gmail.com  Mon Jul  8 20:44:06 2013
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On 8 July 2013 20:03, Silvia Pfeiffer <silviapfeiffer1@gmail.com> wrote:
> Or they didn't understand the consequences yet.... Also, it's
> easy to object on a mailing list, while in a meeting room, you'd have
> to provide an alternative and it's really hard to come up with such an
> alternative IMHO.

Hey Silvia,

We were there, in person, in Lyon, last November, but there were just
two of us.  We even had a viable alternative.

It's easy to dismiss two people.  Almost as easy, if not even easier,
to dismiss an absent mass.

From silviapfeiffer1@gmail.com  Mon Jul  8 20:49:58 2013
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Date: Tue, 9 Jul 2013 13:49:37 +1000
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Tue, Jul 9, 2013 at 1:44 PM, Martin Thomson <martin.thomson@gmail.com> wrote:
> On 8 July 2013 20:03, Silvia Pfeiffer <silviapfeiffer1@gmail.com> wrote:
>> Or they didn't understand the consequences yet.... Also, it's
>> easy to object on a mailing list, while in a meeting room, you'd have
>> to provide an alternative and it's really hard to come up with such an
>> alternative IMHO.
>
> Hey Silvia,
>
> We were there, in person, in Lyon, last November, but there were just
> two of us.  We even had a viable alternative.

Right!

> It's easy to dismiss two people.  Almost as easy, if not even easier,
> to dismiss an absent mass.

Agreed. But I think things are different now that there is so much
more experience so maybe next TPAC will look different.

Cheers,
Silvia.

From stefan.lk.hakansson@ericsson.com  Tue Jul  9 00:40:38 2013
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Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/9/13 5:44 AM, Martin Thomson wrote:=0A=
> On 8 July 2013 20:03, Silvia Pfeiffer <silviapfeiffer1@gmail.com> wrote:=
=0A=
>> Or they didn't understand the consequences yet.... Also, it's=0A=
>> easy to object on a mailing list, while in a meeting room, you'd have=0A=
>> to provide an alternative and it's really hard to come up with such an=
=0A=
>> alternative IMHO.=0A=
>=0A=
> Hey Silvia,=0A=
>=0A=
> We were there, in person, in Lyon, last November, but there were just=0A=
> two of us.  We even had a viable alternative.=0A=
=0A=
I want to be very clear and careful on what I say. So I am repeating:=0A=
=0A=
* My comment that I think Eric is right in that there is consensus on =0A=
providing APIs that allow most use-cases to be met without SDP mangling =0A=
is meant only in that context: SDP O/A is kept (and PeerConnection more =0A=
or less as is and so on).=0A=
=0A=
I don't think Martin or anyone else objected to this in Lyon or at any =0A=
other discussion.=0A=
=0A=
It is also true that an alternative API proposal, CU-rtcweb, was brought =
=0A=
forward to the WebRTC WG earlier during 2012. The WG took this under =0A=
consideration, and (at that time) decided to continue developing based =0A=
on the PeerConnection API including SDP O/A.=0A=
=0A=
I think the discussion in Lyon only evolved around PeerConnection (wiht =0A=
SDP O/A), we did not discuss other API alternatives.=0A=
=0A=
>=0A=
> It's easy to dismiss two people.  Almost as easy, if not even easier,=0A=
> to dismiss an absent mass.=0A=
> _______________________________________________=0A=
> rtcweb mailing list=0A=
> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2013/7/9 Silvia Pfeiffer <silviapfeiffer1@gmail.com>:
> I think there are two groups objecting and there are two things to
> object against.
>
> Here's what I'm reading...
>
> Web Developers really just want to be given functionality. They can
> build their own nice API on top of what browsers implement (jquery did
> a marvellous job of that). But they want functionality. Seeing as some
> things are not currently possible with SDP and O/A, they either want
> to extend those to allow their functionality, or - in particular where
> SDP & O/A are overkill - want an option to get out of using SDP & O/A
> and use something else.
>
> Telco engineers want their existing devices to be interoperable with
> browsers as well as the ability to develop new devices in the future
> with extended functionality. So, on behalf of the first goal, they
> object to including functionality into SDP that other devices don't
> already support, and on behalf of the second goal, they are torn
> between extending SDP & O/A and starting new.

Hi Silvia,

Another important point is that many of those telco engineers have
clearly recognized that they don't know Web, nor JS, nor designing JS
APIs for the Web, and that they just want WebRTC to be compatible with
their current business (literally: "I want browsers to be compatible
with SIP").

For example, two years ago the initial proposal by many of those telco
engineers was "allow plain RTP and mandate SIP as in-the-wire protocol
for WebRTC" (this is true). I also remember that somebody proposed
that the browser should include a "call history" panel, which clearly
shows a limited and wrong WWW vision.

They strongly think that by mandating SDP O/A they are done and have
accomplished their job in this WG on behalf of their big companies
(those that have never succeeded in the WWW). And of course, they are
much more used to IETF processes than Web guys (is Facebook
represented in this WG?), thus the only argument they expose
*nowadays* is: "SDP mandate was talked about and agreed two years ago
in a meeting full of telco engineers".

And then, the current spec is basically an opaque and unmanageable SDP
blob generated by the browser that must be sent by the JS to the peer.
The same "logic" within a phone. But here is the problem: a web
application is all but **fixed** code and **fixed** logic. Don't kill
the Web.




> I certainly want a simple API, but I am not sure whether what we
> currently have is already the simplest possible (given current
> circumstances),

What we have now is not a real API.


> nor am I sure whether it wouldn't just be possible to
> build a simple one on top the jquery way, which would be sufficient
> for now.

An API should be good enough to not require an extra layer on top of
it to make it feasible for developers.


As developer in JsSIP project I must say that we have properly
implemented SIP pure protocol on top of JS and WebSocket, no problems
at all (regardless how complex SIP becomes sometimes). The only real
problem we have if related to WebRTC "API". Tons of hours dealing with
unexpected/wrong specifications and behaviors (and yes, SDP blob
hackery). We would be very happy if we could generate our *own* SDP
with JS (which means we need to retrieve *real* JS Objects from the
WebRTC stack rather than an opaque SDP blob). And telcos, ey!!!, we
are doing SIP even much more than you !!! believe us !!!


Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 9 Jul 2013 12:18:00 +0200
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/9 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> I want to be very clear and careful on what I say. So I am repeating:
>
> * My comment that I think Eric is right in that there is consensus on
> providing APIs that allow most use-cases to be met without SDP mangling
> is meant only in that context: SDP O/A is kept (and PeerConnection more
> or less as is and so on).

Hi Stefan,

You insist that "there is consensus on keeping SDP O/A but providing a
better API". Given current discussions IMHO it is clear there is not
such a consensus (not at all). Or may be you are just talking about
two years ago in some IETF meeting (if so I'm sorry).

Please review the results in
https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZ=
dV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1

  Bad for advanced stuff: 94%
  OK for 1.0: 40%

IMHO such a result indicates all but "let's keep SDP O/A and improve a
bit the API" (I mean now, in July 2013).


Best regards.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 9 Jul 2013 12:40:27 +0200
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Subject: Re: [rtcweb] Reducing glare
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2013/5/10 Dale R. Worley <worley@ariadne.com>:
> Two possible solutions for glare problems come to mind:
>
> One is to require the designated "follower" to accept and give a 200
> response to an incoming re-INVITE even if it has an outstanding
> re-INVITE.  The difficulty that follows is that the follower doesn't
> know whether the "leader" has accepted or rejected its re-INVITE until
> the response arrives, so there is an interval (between sending the 200
> to the leader's re-INVITE and receiving the leaders response to its
> re-INVITE) during which the follower doesn't know its outgoing media
> configuration.  We could probably fix that by annotating the leader's
> offers with the o=3D value of the follower's SDP that the leader
> currently is using.
>
> A second one would be to simply drastically shorten the glare backoff
> timer.  Given that we're talking about video systems, the RFC 3261
> timer ranges of 0 to 2 secs and 2.1 to 4 secs are very slow.  If we
> cut those times by a factor of 10, wouldn't the user-visible problems
> nearly vanish?


A third and better solution would be to stop assuming that SIP
problems are also WebRTC problems. The "glare" problem occurs due to
the artificial mandate of SDP O/A in WebRTC.

Want to avoid the "glare" problem and "SDP v=3D" number problem? Remove
SDP O/A from WebRTC and you are done. Then if somebody wants to play
with SDP at JS it can do it **at JS level**.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 2013-07-09 12:18, I=F1aki Baz Castillo wrote:=0A=
> 2013/7/9 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com>:=0A=
>> I want to be very clear and careful on what I say. So I am repeating:=0A=
>>=0A=
>> * My comment that I think Eric is right in that there is consensus on=0A=
>> providing APIs that allow most use-cases to be met without SDP mangling=
=0A=
>> is meant only in that context: SDP O/A is kept (and PeerConnection more=
=0A=
>> or less as is and so on).=0A=
>=0A=
> Hi Stefan,=0A=
>=0A=
> You insist that "there is consensus on keeping SDP O/A but providing a=0A=
> better API".=0A=
=0A=
I must be expressing myself quite badly, because the message seems not =0A=
to get through.=0A=
=0A=
* I think we have consensus on adding API that allows most use-cases to =0A=
be met without SDP mangling (given that SDP O/A is kept naturally)=0A=
=0A=
* We discussed last year an alternative API that did not use SDP at all =0A=
(and did not use PeerConnection). The decision at that time was to =0A=
continue developing the API based on PeerConnection and SDP O/A.=0A=
=0A=
Have not stated anything more than that.=0A=
=0A=
Stefan=0A=
=0A=
> Given current discussions IMHO it is clear there is not=0A=
> such a consensus (not at all). Or may be you are just talking about=0A=
> two years ago in some IETF meeting (if so I'm sorry).=0A=
>=0A=
> Please review the results in=0A=
> https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdH=
lZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1=0A=
>=0A=
>    Bad for advanced stuff: 94%=0A=
>    OK for 1.0: 40%=0A=
>=0A=
> IMHO such a result indicates all but "let's keep SDP O/A and improve a=0A=
> bit the API" (I mean now, in July 2013).=0A=
>=0A=
>=0A=
> Best regards.=0A=
>=0A=
>=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

From ted.ietf@gmail.com  Tue Jul  9 08:33:09 2013
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--20cf3036361fc983b104e115e0b4
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Howdy,

The recent set of API discussions has been spread across both the rtcweb
and public-webrtc mailing lists.  That's making it both harder to follow
and harder for folks to work out who is saying what under which rules.  The
chairs of both groups believe that the right place for the discussion to
continue should be public-webrtc.  Please direct follow-ups on this topic
to that list.

regards,

Ted Hardie

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Howdy,<br><br>The recent set of API discussions has been spread across both=
 the rtcweb and public-webrtc mailing lists.=A0 That&#39;s making it both h=
arder to follow and harder for folks to work out who is saying what under w=
hich rules.=A0 The chairs of both groups believe that the right place for t=
he discussion to continue should be public-webrtc.=A0 Please direct follow-=
ups on this topic to that list.<br>
<br>regards,<br><br>Ted Hardie<br>

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From derhoermi@gmx.net  Tue Jul  9 10:26:33 2013
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From: Bjoern Hoehrmann <derhoermi@gmx.net>
To: cowwoc <cowwoc@bbs.darktech.org>
Date: Tue, 09 Jul 2013 19:26:08 +0200
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Subject: Re: [rtcweb] Locus of API discussion
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* cowwoc wrote:
>     I agree, with one caveat: virtually none of the high-level 
>stakeholders (spec editors, browser vendors, etc) bother to engage the 
>community on public-webrtc.

W3C's Web Real-Time Communications Working Group is required to address
all feedback it receives in a timely and technically sound manner. Take
http://lists.w3.org/Archives/Public/public-script-coord/2013AprJun/0081
up Anne van Kesteren's offer to help if that is not working out for you.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
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Subject: Re: [rtcweb] Locus of API discussion
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Could you explain the reasoning behind moving the API discussion to the W3C=
 list while leaving the actual API specification documents as Internet Draf=
ts created and edited by the IETF WG?

I'm all for moving the API work (back) to W3C, but we should move all of it=
, don't you think?

Matthew Kaufman

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ted Hardie
Sent: Tuesday, July 9, 2013 8:33 AM
To: rtcweb@ietf.org; public-webrtc@w3.org
Subject: [rtcweb] Locus of API discussion

Howdy,

The recent set of API discussions has been spread across both the rtcweb an=
d public-webrtc mailing lists.  That's making it both harder to follow and =
harder for folks to work out who is saying what under which rules.  The cha=
irs of both groups believe that the right place for the discussion to conti=
nue should be public-webrtc.  Please direct follow-ups on this topic to tha=
t list.

regards,

Ted Hardie

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<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D">Could you explain the reasoning behind moving the API discussion to the =
W3C list while leaving the actual API specification documents
 as Internet Drafts created and edited by the IETF WG?<o:p></o:p></span></a=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I&#8217;m all for moving =
the API work (back) to W3C, but we should move all of it, don&#8217;t you t=
hink?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Matthew Kaufman<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
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<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"> rtcweb=
-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> Tuesday, July 9, 2013 8:33 AM<br>
<b>To:</b> rtcweb@ietf.org; public-webrtc@w3.org<br>
<b>Subject:</b> [rtcweb] Locus of API discussion<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Howdy,<br>
<br>
The recent set of API discussions has been spread across both the rtcweb an=
d public-webrtc mailing lists.&nbsp; That's making it both harder to follow=
 and harder for folks to work out who is saying what under which rules.&nbs=
p; The chairs of both groups believe that
 the right place for the discussion to continue should be public-webrtc.&nb=
sp; Please direct follow-ups on this topic to that list.<br>
<br>
regards,<br>
<br>
Ted Hardie<o:p></o:p></p>
</div>
</div>
</body>
</html>

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Subject: Re: [rtcweb] Locus of API discussion
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Agreed, and in addition, is SDP with offer / answer a mandated surface 
API by the RTCWEB WG or the W3C? (i.e. a mandate that some use cases are 
only solvable with touching the SDP, and web developer must know about 
offer/answer in the process)

If so, which group is mandating that requirement be part of the web 
developer's API?

This helps define which group is appropriate to address those kinds of 
concerns.

-Robin


> Matthew Kaufman (SKYPE) <mailto:matthew.kaufman@skype.net>
> 9 July, 2013 2:04 PM
>
> Could you explain the reasoning behind moving the API discussion to 
> the W3C list while leaving the actual API specification documents as 
> Internet Drafts created and edited by the IETF WG?
>
> I'm all for moving the API work (back) to W3C, but we should move all 
> of it, don't you think?
>
> Matthew Kaufman
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Ted Hardie
> *Sent:* Tuesday, July 9, 2013 8:33 AM
> *To:* rtcweb@ietf.org; public-webrtc@w3.org
> *Subject:* [rtcweb] Locus of API discussion
>
> Howdy,
>
> The recent set of API discussions has been spread across both the 
> rtcweb and public-webrtc mailing lists.  That's making it both harder 
> to follow and harder for folks to work out who is saying what under 
> which rules.  The chairs of both groups believe that the right place 
> for the discussion to continue should be public-webrtc.  Please direct 
> follow-ups on this topic to that list.
>
> regards,
>
> Ted Hardie
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> Ted Hardie <mailto:ted.ietf@gmail.com>
> 9 July, 2013 11:33 AM
> Howdy,
>
> The recent set of API discussions has been spread across both the 
> rtcweb and public-webrtc mailing lists.  That's making it both harder 
> to follow and harder for folks to work out who is saying what under 
> which rules.  The chairs of both groups believe that the right place 
> for the discussion to continue should be public-webrtc.  Please direct 
> follow-ups on this topic to that list.
>
> regards,
>
> Ted Hardie
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head>
<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
</head><body bgcolor="#FFFFFF" text="#000000"><br>
Agreed, and in addition, is SDP with offer / answer a mandated surface 
API by the RTCWEB WG or the W3C? (i.e. a mandate that some use cases are
 only solvable with touching the SDP, and web developer must know about 
offer/answer in the process)<br>
<br>
If so, which group is mandating that requirement be part of the web 
developer's API?<br>
<br>
This helps define which group is appropriate to address those kinds of 
concerns.<br>
<br>
-Robin<br>
<br>
<br>
<blockquote style="border: 0px none;" 
cite="mid:AE1A6B5FD507DC4FB3C5166F3A05A4841D143E48@TK5EX14MBXC274.redmond.corp.microsoft.com"
 type="cite">
  <div style="margin:30px 25px 10px 25px;" class="__pbConvHr"><div 
style="display:table;width:100%;border-top:1px solid 
#EDEEF0;padding-top:5px"> 	<div 
style="display:table-cell;vertical-align:middle;padding-right:6px;"><img
 photoaddress="matthew.kaufman@skype.net" photoname="Matthew Kaufman 
(SKYPE)" src="cid:part1.05060700.05070807@hookflash.com" 
name="compose-unknown-contact.jpg" height="25px" width="25px"></div>   <div
 
style="display:table-cell;white-space:nowrap;vertical-align:middle;width:100%">
   	<a moz-do-not-send="true" href="mailto:matthew.kaufman@skype.net" 
style="color:#737F92 
!important;padding-right:6px;font-weight:bold;text-decoration:none 
!important;">Matthew Kaufman (SKYPE)</a></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;">   
  <font color="#9FA2A5"><span style="padding-left:6px">9 July, 2013 2:04
 PM</span></font></div></div></div>
  <div style="color:#888888;margin-left:24px;margin-right:24px;" 
__pbrmquotes="true" class="__pbConvBody">

<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
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<div class="WordSection1">
<p class="MsoNormal"><a moz-do-not-send="true" name="_MailEndCompose"><span
 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Could
 you explain the reasoning behind moving the API discussion to the W3C 
list while leaving the actual API specification documents
 as Internet Drafts created and edited by the IETF WG?<o:p></o:p></span></a></p>
<p class="MsoNormal"><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I&#8217;m
 all for moving the API work (back) to W3C, but we should move all of 
it, don&#8217;t you think?<o:p></o:p></span></p>
<p class="MsoNormal"><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Matthew
 Kaufman<o:p></o:p></span></p>
<p class="MsoNormal"><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div style="border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in
 4.0pt">
<div>
<div style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in
 0in 0in">
<p class="MsoNormal"><b><span 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span
 
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">
 <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> Tuesday, July 9, 2013 8:33 AM<br>
<b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a><br>
<b>Subject:</b> [rtcweb] Locus of API discussion<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
<p class="MsoNormal">Howdy,<br>
<br>
The recent set of API discussions has been spread across both the rtcweb
 and public-webrtc mailing lists.&nbsp; That's making it both harder to 
follow and harder for folks to work out who is saying what under which 
rules.&nbsp; The chairs of both groups believe that
 the right place for the discussion to continue should be 
public-webrtc.&nbsp; Please direct follow-ups on this topic to that list.<br>
<br>
regards,<br>
<br>
Ted Hardie<o:p></o:p></p>
</div>
</div>
<div>_______________________________________________<br>rtcweb mailing 
list<br><a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></div></div>
  <div style="margin:30px 25px 10px 25px;" class="__pbConvHr"><div 
style="display:table;width:100%;border-top:1px solid 
#EDEEF0;padding-top:5px"> 	<div 
style="display:table-cell;vertical-align:middle;padding-right:6px;"><img
 photoaddress="ted.ietf@gmail.com" photoname="Ted Hardie" 
src="cid:part1.05060700.05070807@hookflash.com" 
name="compose-unknown-contact.jpg" height="25px" width="25px"></div>   <div
 
style="display:table-cell;white-space:nowrap;vertical-align:middle;width:100%">
   	<a moz-do-not-send="true" href="mailto:ted.ietf@gmail.com" 
style="color:#737F92 
!important;padding-right:6px;font-weight:bold;text-decoration:none 
!important;">Ted Hardie</a></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;">   
  <font color="#9FA2A5"><span style="padding-left:6px">9 July, 2013 
11:33 AM</span></font></div></div></div>
  <div style="color:#888888;margin-left:24px;margin-right:24px;" 
__pbrmquotes="true" class="__pbConvBody">Howdy,<br><br>The recent set of
 API discussions has been spread across both the rtcweb and 
public-webrtc mailing lists.&nbsp; That's making it both harder to follow and
 harder for folks to work out who is saying what under which rules.&nbsp; The
 chairs of both groups believe that the right place for the discussion 
to continue should be public-webrtc.&nbsp; Please direct follow-ups on this 
topic to that list.<br>
<br>regards,<br><br>Ted Hardie<br>

<div>_______________________________________________<br>rtcweb mailing 
list<br><a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></div></div>
</blockquote>
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On Tue, Jul 9, 2013 at 9:11 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:

> On 09/07/2013 11:33 AM, Ted Hardie wrote:
>
>> Howdy,
>>
>> The recent set of API discussions has been spread across both the rtcweb
>> and public-webrtc mailing lists.  That's making it both harder to follow
>> and harder for folks to work out who is saying what under which rules.  The
>> chairs of both groups believe that the right place for the discussion to
>> continue should be public-webrtc.  Please direct follow-ups on this topic
>> to that list.
>>
>> regards,
>>
>> Ted Hardie
>>
> Ted,
>
>     I agree, with one caveat: virtually none of the high-level
> stakeholders (spec editors, browser vendors, etc) bother to engage the
> community on public-webrtc.
>

I'm not sure I understand this complaint. Is it that the aforementioned
"high-level stakeholders"
aren't engaging or merely that they are only engaging on RTCWEB? If it's
the former, than
I don't think that's actually true, since in the past week, you've had
responses from (at least)
the following people who fall into those categories:

Cullen Jennings (spec editor)
Adam Bergqvist (spec editor)
Peter Thatcher (works on Chrome)
Me (works on Firefox and Chrome; spec editor)
Christer Holmberg (spec editor)
Several people from Microsoft.

Who, exactly, are you expecting to engage that hasn't engaged?


If your complaint is just that they're engaging on the wrong mailing list,
well
that seems to reinforce Ted's point above.

-Ekr




    What's the point of discussing the API on this mailing list if our
> opinion goes unnoticed? We shouldn't be moving the discussion to
> public-webrtc as a nice way to filter us out. This discussion requires
> their attention, be it on one mailing list or the other. I don't mind where
> we discuss it, so long as they get involved.
>
>     Is it their intention to get involved on public-webrtc and summarize
> the results on rtcweb?
>
> Thanks,
> Gili
>
>

--001a11c2e4d4f4ef2e04e11a2df6
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Jul 9, 2013 at 9:11 AM, cowwoc <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:cowwoc@bbs.darktech.org" target=3D"_blank">cowwoc@bbs.darktech.=
org</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div>On 09/07/2013 11:33 AM, Ted Hardie=
 wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Howdy,<br>
<br>
The recent set of API discussions has been spread across both the rtcweb an=
d public-webrtc mailing lists. =A0That&#39;s making it both harder to follo=
w and harder for folks to work out who is saying what under which rules. =
=A0The chairs of both groups believe that the right place for the discussio=
n to continue should be public-webrtc. =A0Please direct follow-ups on this =
topic to that list.<br>



<br>
regards,<br>
<br>
Ted Hardie<br>
</blockquote></div></div>
Ted,<br>
<br>
=A0 =A0 I agree, with one caveat: virtually none of the high-level stakehol=
ders (spec editors, browser vendors, etc) bother to engage the community on=
 public-webrtc.<br></blockquote><div><br></div><div>I&#39;m not sure I unde=
rstand this complaint. Is it that the aforementioned &quot;high-level stake=
holders&quot;</div>


<div>aren&#39;t engaging or merely that they are only engaging on RTCWEB? I=
f it&#39;s the former, than</div><div>I don&#39;t think that&#39;s actually=
 true, since in the past week, you&#39;ve had responses from (at least)</di=
v>

<div>the following people who fall into those categories:</div><div><br></d=
iv><div>Cullen Jennings (spec editor)</div><div>Adam Bergqvist (spec editor=
)</div><div>Peter Thatcher (works on Chrome)</div><div style>Me (works on F=
irefox and Chrome; spec editor)</div>

<div style>Christer Holmberg (spec editor)</div><div style>Several people f=
rom Microsoft.</div><div style><br></div><div style>Who, exactly, are you e=
xpecting to engage that hasn&#39;t engaged?=A0</div><div style><br></div>

<div style><br></div><div style>If your complaint is just that they&#39;re =
engaging on the wrong mailing list, well</div><div style>that seems to rein=
force Ted&#39;s point above.</div><div style><br></div><div style>-Ekr</div=
>

<div style><br></div><div><br></div><div><br></div><div><br></div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">
=A0 =A0 What&#39;s the point of discussing the API on this mailing list if =
our opinion goes unnoticed? We shouldn&#39;t be moving the discussion to pu=
blic-webrtc as a nice way to filter us out. This discussion requires their =
attention, be it on one mailing list or the other. I don&#39;t mind where w=
e discuss it, so long as they get involved.<br>



<br>
=A0 =A0 Is it their intention to get involved on public-webrtc and summariz=
e the results on rtcweb?<br>
<br>
Thanks,<br>
Gili<br>
<br>
</blockquote></div><br></div></div>

--001a11c2e4d4f4ef2e04e11a2df6--

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On Mon, Jul 8, 2013 at 1:15 PM, Justin Uberti <juberti@google.com> wrote:

> On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi Perumal (mperumal) <
> mperumal@cisco.com> wrote:
>>
>> 3) There is no text describing how the timestamp encoded in the UNSERNAME
>> attribute of the ALLOCAE requested could be protected.
>>
>
> In the HTTP Interactions section, I mention that the password used for the
> MESSAGE-INTEGRITY is a digest of the username, but I can make this more
> explicit.
>

Muthu,

What situations did you have in mind with regards to timestamp protection
that potentially would be vulnerable (beyond any protection the the digest
gives)?

Matthew Fredrickson

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<div dir=3D"ltr">On Mon, Jul 8, 2013 at 1:15 PM, Justin Uberti <span dir=3D=
"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@g=
oogle.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=3D=
"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"gmail_quote">On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi=
 Perumal (mperumal) <span dir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.=
com" target=3D"_blank">mperumal@cisco.com</a>&gt;</span> wrote:<blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">3) There is no text describing how the t=
imestamp encoded in the UNSERNAME attribute of the ALLOCAE requested could =
be protected.</span></p>


</div></blockquote><div>=A0<br></div><div>In the HTTP Interactions section,=
 I mention that the password used for the MESSAGE-INTEGRITY is a digest of =
the username, but I can make this more explicit.</div></div></div></div>
</blockquote><div><br></div><div>Muthu,</div><div><br></div><div>What situa=
tions did you have in mind with regards to timestamp protection that potent=
ially would be vulnerable (beyond any protection the the digest gives)?</di=
v>
<div><br></div><div>Matthew Fredrickson=A0</div></div></div></div>

--001a11c266bebe5fec04e11b442b--

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On Mon, Jul 8, 2013 at 5:50 PM, Adam Roach <adam@nostrum.com> wrote:

> On 7/7/13 23:24, Justin Uberti wrote:
>
>> Just uploaded a 00 version of a spec for requesting time-limited TURN
>> credentials for WebRTC apps. Would like to get 10 minutes of agenda time to
>> present this in Berlin.
>>
>
> Thanks for getting ball rolling on this, Justin. I agree that this is an
> important mechanism to have defined.
>
> I only find one issue in my initial read through the document: why are we
> proposing the use of HTTP POST for stateless, idempotent information
> retrieval? The key benefit of this approach is that it doesn't actually
> create state in the network. That's a much better fit for GET.
>
> Compare RFC 2616, sections 9.3 and 9.5, and see if you don't agree.


I originally had this as a GET and switched it to a POST for basically the
same reasons that Martin outlined - it wasn't completely stateless, and two
different requests would return different usernames (because of new
timestamps). But if these concerns are overall insignificant, I would be
more than happy to switch back to GET.

--001a11c338d035177904e120df12
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 8, 2013 at 5:50 PM, Adam Roach <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/7/13 23:24, Justin Ub=
erti wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Just uploaded a 00 version of a spec for requesting time-limited TURN crede=
ntials for WebRTC apps. Would like to get 10 minutes of agenda time to pres=
ent this in Berlin.<br>
</blockquote>
<br></div>
Thanks for getting ball rolling on this, Justin. I agree that this is an im=
portant mechanism to have defined.<br>
<br>
I only find one issue in my initial read through the document: why are we p=
roposing the use of HTTP POST for stateless, idempotent information retriev=
al? The key benefit of this approach is that it doesn&#39;t actually create=
 state in the network. That&#39;s a much better fit for GET.<br>


<br>
Compare RFC 2616, sections 9.3 and 9.5, and see if you don&#39;t agree.</bl=
ockquote><div><br></div><div>I originally had this as a GET and switched it=
 to a POST for basically the same reasons that Martin outlined - it wasn&#3=
9;t completely stateless, and two different requests would return different=
 usernames (because of new timestamps). But if these concerns are overall i=
nsignificant, I would be more than happy to switch back to GET.</div>

</div><br></div></div>

--001a11c338d035177904e120df12--

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Hi all,

We submitted a new draft proposing to make use of existing codec APIs on =
mobile terminals to provide a more cost-effective way of achieving =
better interoperability without extra costs for the browser vendors to =
re-implement relevant audio/video codecs. It also proposes to add =
notifications from the browser to web apps whether or not a codec is =
provided by the local platform or not, so that the apps can make an =
informed decision on which codec to use in its specific communication =
scenario.

http://www.ietf.org/internet-drafts/draft-deng-rtcweb-codecapi-00.txt

Your comments are welcome.

BR
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: internet-drafts@ietf.org =
[mailto:internet-drafts@ietf.org]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2013=E5=B9=B47=E6=9C=888=E6=97=A5 =
12:57
=E6=94=B6=E4=BB=B6=E4=BA=BA: Deng Lingli; Yu Qing; Lingfei Ni; Qing Yu; =
Lingli Deng
=E4=B8=BB=E9=A2=98: New Version Notification for =
draft-deng-rtcweb-codecapi-00.txt


A new version of I-D, draft-deng-rtcweb-codecapi-00.txt
has been successfully submitted by Lingli Deng and posted to the
IETF repository.

Filename:	 draft-deng-rtcweb-codecapi
Revision:	 00
Title:		 Proposal for reusing local codec APIs on mobile devices
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 5
URL:             =
http://www.ietf.org/internet-drafts/draft-deng-rtcweb-codecapi-00.txt
Status:          =
http://datatracker.ietf.org/doc/draft-deng-rtcweb-codecapi
Htmlized:        =
http://tools.ietf.org/html/draft-deng-rtcweb-codecapi-00


Abstract:
   This document proposes the browser to make use of local codec APIs,
   if available on a mobile device, and notifies RTCWEB applications to
   allow for better tuning between device capability and media quality.

                                                                         =
        =20


The IETF Secretariat





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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Matt Fredrickson <creslin@digium.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb-turn-rest-00.txt
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|What situations did you have in mind with regards to timestamp protection
|that potentially would be vulnerable (beyond any protection the the digest
|gives)?

I think the digest protection should suffice for all practical purposes. On=
 the other hand, is the web and TURN server clocks expected to be in sync? =
I think the draft should discuss it.

Muthu

From: Matt Fredrickson [mailto:creslin@digium.com]
Sent: Wednesday, July 10, 2013 3:29 AM
To: Justin Uberti
Cc: Muthu Arul Mozhi Perumal (mperumal); rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-rtcweb=
-turn-rest-00.txt

On Mon, Jul 8, 2013 at 1:15 PM, Justin Uberti <juberti@google.com<mailto:ju=
berti@google.com>> wrote:
On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi Perumal (mperumal) <mperum=
al@cisco.com<mailto:mperumal@cisco.com>> wrote:
3) There is no text describing how the timestamp encoded in the UNSERNAME a=
ttribute of the ALLOCAE requested could be protected.

In the HTTP Interactions section, I mention that the password used for the =
MESSAGE-INTEGRITY is a digest of the username, but I can make this more exp=
licit.

Muthu,

What situations did you have in mind with regards to timestamp protection t=
hat potentially would be vulnerable (beyond any protection the the digest g=
ives)?

Matthew Fredrickson

--_000_E721D8C6A2E1544DB2DEBC313AF54DE224189868xmbrcdx02ciscoc_
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<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|What situations did you have in mind wi=
th regards to timestamp protection
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|that potentially would be vulnerable (b=
eyond any protection the the digest
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">|gives)?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">I think the digest protection should suf=
fice for all practical purposes. On the other hand, is the web and TURN ser=
ver clocks expected to be in sync? I think the draft should
 discuss it.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;">Muthu<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Matt Fre=
drickson [mailto:creslin@digium.com]
<br>
<b>Sent:</b> Wednesday, July 10, 2013 3:29 AM<br>
<b>To:</b> Justin Uberti<br>
<b>Cc:</b> Muthu Arul Mozhi Perumal (mperumal); rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Fwd: New Version Notification for draft-uberti=
-rtcweb-turn-rest-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Mon, Jul 8, 2013 at 1:15 PM, Justin Uberti &lt;<a=
 href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a=
>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<div>
<p class=3D"MsoNormal">On Mon, Jul 8, 2013 at 1:52 AM, Muthu Arul Mozhi Per=
umal (mperumal) &lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_blank"=
>mperumal@cisco.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot=
;,&quot;serif&quot;">3) There is no text describing how the timestamp encod=
ed in the UNSERNAME attribute of the ALLOCAE requested could
 be protected.</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">In the HTTP Interactions section, I mention that the=
 password used for the MESSAGE-INTEGRITY is a digest of the username, but I=
 can make this more explicit.<o:p></o:p></p>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Muthu,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">What situations did you have in mind with regards to=
 timestamp protection that potentially would be vulnerable (beyond any prot=
ection the the digest gives)?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Matthew Fredrickson&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_E721D8C6A2E1544DB2DEBC313AF54DE224189868xmbrcdx02ciscoc_--

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Subject: Re: [rtcweb] Locus of API discussion
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On 09/07/2013 11:33 AM, Ted Hardie wrote:
> Howdy,
>
> The recent set of API discussions has been spread across both the 
> rtcweb and public-webrtc mailing lists.  That's making it both harder 
> to follow and harder for folks to work out who is saying what under 
> which rules.  The chairs of both groups believe that the right place 
> for the discussion to continue should be public-webrtc.  Please direct 
> follow-ups on this topic to that list.
>
> regards,
>
> Ted Hardie
Ted,

     I agree, with one caveat: virtually none of the high-level 
stakeholders (spec editors, browser vendors, etc) bother to engage the 
community on public-webrtc.

     What's the point of discussing the API on this mailing list if our 
opinion goes unnoticed? We shouldn't be moving the discussion to 
public-webrtc as a nice way to filter us out. This discussion requires 
their attention, be it on one mailing list or the other. I don't mind 
where we discuss it, so long as they get involved.

     Is it their intention to get involved on public-webrtc and 
summarize the results on rtcweb?

Thanks,
Gili

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On 09/07/2013 1:26 PM, Bjoern Hoehrmann wrote:
> * cowwoc wrote:
>>      I agree, with one caveat: virtually none of the high-level
>> stakeholders (spec editors, browser vendors, etc) bother to engage the
>> community on public-webrtc.
> W3C's Web Real-Time Communications Working Group is required to address
> all feedback it receives in a timely and technically sound manner. Take
> http://lists.w3.org/Archives/Public/public-script-coord/2013AprJun/0081
> up Anne van Kesteren's offer to help if that is not working out for you.

     Glad to hear it. Thanks!

     I look forward to hearing their feedback.

Gili

From cowwoc@bbs.darktech.org  Tue Jul  9 14:45:46 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Locus of API discussion
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On 09/07/2013 4:40 PM, Eric Rescorla wrote:
> I'm not sure I understand this complaint. Is it that the 
> aforementioned "high-level stakeholders"
> aren't engaging or merely that they are only engaging on RTCWEB? If 
> it's the former, than
> I don't think that's actually true, since in the past week, you've had 
> responses from (at least)
> the following people who fall into those categories:
>
> Cullen Jennings (spec editor)
> Adam Bergqvist (spec editor)
> Peter Thatcher (works on Chrome)
> Me (works on Firefox and Chrome; spec editor)
> Christer Holmberg (spec editor)
> Several people from Microsoft.
>
> Who, exactly, are you expecting to engage that hasn't engaged?
>
Hi Eric,

     It's my understanding that public-webrtc is for discussing the 
WebRTC API, and RTCWeb is for discussing the WebRTC wire protocol.

     Three weeks ago I posted a summary of discussion points that came 
up in the WebRTC World conference (most of which had to do with the 
WebRTC API). To date, I have not received a reply from any of the people 
you have listed. I am unable to gather the necessary momentum to turn 
these points into action items without your help. I was/am frustrated 
that the spec editors and vendors are responsible to engage the 
community on these matters, but did not. I hope this clarifies what I meant.

     On a side-note, I do not view the SDP discussion as a response to 
my post. This discussion was well under-way beforehand and only made up 
one of the seven points I brought up. I would appreciate it if multiple 
stakeholders would address each of the points I brought up.

> If your complaint is just that they're engaging on the wrong mailing 
> list, well
> that seems to reinforce Ted's point above.

     That's fine. I don't mind moving all API discussion to this list, 
so long as they actually reply this time.

Thank you,
Gili

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From andrew.hutton@siemens-enterprise.com  Wed Jul 10 07:00:21 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "rt >> \"rtcweb@ietf.org\"" <rtcweb@ietf.org>
Thread-Topic: draft-ietf-rtcweb-use-cases-and-requirements - HTTP Only Firewall.
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Hi,

Regarding the use case in 3.2.3 covering a HTTP only FW the text states:

"This use-case is almost identical to the Simple Video Communication Servic=
e use-case (Section 3.2.1).  The difference is that one of the users is beh=
ind a FW that only allows http traffic".

This needs to be changed to allow for the common case when a HTTP Proxy is =
deployed so I suggest changing the last sentence to

"The difference is that one of the users is behind a FW that only allows tr=
affic via a HTTP Proxy".

There also needs to be a corresponding change to requirement F37.

I believe we have previously discussed changing this but have to admit I co=
uld not fine the e-mail chain so maybe it was during a meeting.

Regards
Andy




> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stefan H=E5kansson LK
> Sent: 27 June 2013 09:50
> To: rt >> "rtcweb@ietf.org"
> Subject: [rtcweb] Fwd: New Version Notification for draft-ietf-rtcweb-
> use-cases-and-requirements-11.txt
>=20
>  From the change log:
>=20
>     o  Described that the API requirements are really from a W3C
>        perspective and are supplied as an appendix in the introduction.
>        Moved API requirements to an Appendix.
>=20
>     o  Removed the "Conventions" section with the key-words and
> reference
>        to RFC2119.  Also changed uppercase MUST's/SHOULD's to
> lowercase.
>=20
>     o  Added a note on the proposed use of the document to the
>        introduction.
>=20
>     o  Removed the note talking about WS from the "FW that only allows
>        http" use-case.
>=20
>     o  Removed the word "Skype" that was used as example in one of the
>        use-cases.
>=20
>     o  Clarified F3 (the req saying the everything the browser sends
> must
>        be rate controlled).
>=20
>     o  Removed the TBD saying we need to define reasonable levels from
>        the requirement saying that quality must be good even in
> presence
>        of packet losses (F5), and changed "must" to "should" (Based on
> a
>        list discussion involving Bernard).
>=20
>     o  Removed F6 ("The browser must be able to handle high loss and
>        jitter levels in a graceful way."), also after a list
> discussion.
>=20
>     o  Clarified F7 (used to say that the browser must support fast
>        stream switches, now says that reference frames must be inserted
>        when requested).
>     o  Removed the questions from F9 (echo cancellation), F10
>        (syncronization), F21 (telephony codec).
>=20
>     o  Exchanged "restrictive firewalls" for "limited middleboxes" in
> F19
>        (as proposed by Martin).
>=20
>     o  Expanded DTMF and IVR in F22 (proposed by Martin)
>=20
>     o  Added ref to RFC5405 in F23 (proposed by Lars Eggert).
>=20
>     o  Exchanged "service provided" for "web application" in F32.
>=20
>     o  Changed the text in 3.2.1 that motivates F36 (new text "It is
>        essential that media and data be encrypted, authenticated ...
>        bound to the user identity."); and rewrote F36, included a ref
> to
>        RFC5479.
>=20
>     o  Changed "quality of service" to "quality of experience" in F38.
>=20
>     o  Added F39.
>=20
>     o  Used new formulation of A17 (proposed by Martin).
>=20
>     o  Updated A20.
>=20
>     o  Updated A25.
>=20
> Things that have not been done:
>=20
> - No use-case on emergency services added (as said already in
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07253.html)
>=20
> - No use-case on real-time text added (as said already in
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07254.html)
>=20
> - No clarification on what solution(s) related to multiple resolutions
> of the same content added (discussed in
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07256.html but
> no
> input received).
>=20
> - The order of the requirements (Fn) is still a mess - but I kept it as
> is for this version to make diffing simpler. To be fixed in an upcoming
> version.
>=20
> Stefan
>=20
>=20
> -------- Original Message --------
> Subject: New Version Notification for
> draft-ietf-rtcweb-use-cases-and-requirements-11.txt
> Date: 10:40
> From: internet-drafts@ietf.org <internet-drafts@ietf.org>
> To: Christer Holmberg <christer.holmberg@ericsson.com>,    G=F6ran
> Eriksson AP <goran.ap.eriksson@ericsson.com>,    Stefan H=E5kansson LK
> <stefan.lk.hakansson@ericsson.com>,    G=F6ran Eriksson AP
> <goran.ap.eriksson@ericsson.com>
>=20
>=20
> A new version of I-D, draft-ietf-rtcweb-use-cases-and-requirements-
> 11.txt
> has been successfully submitted by Christer Holmberg and posted to the
> IETF repository.
>=20
> Filename:	 draft-ietf-rtcweb-use-cases-and-requirements
> Revision:	 11
> Title:		 Web Real-Time Communication Use-cases and
> Requirements
> Creation date:	 2013-06-27
> Group:		 rtcweb
> Number of pages: 30
> URL:
> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-
> requirements-11.txt
> Status:
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-
> requirements
> Htmlized:
> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-
> requirements-11
> Diff:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-use-cases-and-
> requirements-11
>=20
> Abstract:
>     This document describes web based real-time communication use-
> cases.
>     Requirements on the browser functionality are derived from use-
> cases.
>=20
>=20
>=20
>=20
>=20
> The IETF Secretariat
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Matt Fredrickson <creslin@digium.com>
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On Wed, Jul 10, 2013 at 1:40 AM, Muthu Arul Mozhi Perumal (mperumal) <
mperumal@cisco.com> wrote:

> I think the digest protection should suffice for all practical purposes.
> On the other hand, is the web and TURN server clocks expected to be in
> sync? I think the draft should discuss it.
>

I tend to agree as well.  Just from an implementation perspective, it seems
like that a presumed current timestamp on the credentials could be useful
for TTLs on the credentials.  It seems that this also could be a potential
security vulnerability if the time was out of sync between the two servers.

Matthew Fredrickson

--089e01493b1649263f04e129b2b5
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<div dir=3D"ltr">On Wed, Jul 10, 2013 at 1:40 AM, Muthu Arul Mozhi Perumal =
(mperumal) <span dir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.com" targ=
et=3D"_blank">mperumal@cisco.com</a>&gt;</span> wrote:<div class=3D"gmail_e=
xtra"><div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&=
quot;Courier New&quot;,&quot;serif&quot;">I think the digest protection sho=
uld suffice for all practical purposes. On the other hand, is the web and T=
URN server clocks expected to be in sync? I think the draft should
 discuss it.</span></p></div></blockquote><div><br></div><div>I tend to agr=
ee as well. =A0Just from an implementation perspective, it seems like that =
a presumed current timestamp on the credentials could be useful for TTLs on=
 the credentials. =A0It seems that this also could be a potential security =
vulnerability if the time was out of sync between the two servers.</div>
<div><br></div><div>Matthew Fredrickson</div></div></div></div>

--089e01493b1649263f04e129b2b5--

From eckert@cisco.com  Wed Jul 10 11:02:34 2013
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Date: Wed, 10 Jul 2013 11:01:47 -0700
From: "Toerless Eckert (eckert)" <eckert@cisco.com>
To: tsvwg@ietf.org, rtcweb@ietf.org, rmcat@ietf.org
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Subject: [rtcweb] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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When dealing with congestion for real-time datagram (UDP/RTP) based video flows, one idea
that we think has shown to provide good value is to intelligently drop in the network
preferentially the least important packets within a video flow, for example non-referenced
P-frames, because their loss creates the lowest impact to video quality.

One of the main concerns raised against such schemes was one of fairness. If flows 
would start to indicate different priorities for packets in them, how would we
be able to avoid that flows would indicate all their packets to be of high priority
to shift packet drops to other flows. Or even more importantly: how could we motivate
flows to indicate that some of their packets are low priority without them having to
fear that they would experience more loss than than flows that don't do this.

So, there is one draft that we would like to present (presumably in TSVWG)  explaining
how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.txt

(Actually, this is how we did solve the problem in our implementatin of intelligent dropping,
 so this is not theoretical).

Now granted, the problem of fairness is but one element of building a complete ssytem
around the idea of intelligent dropping, so if folks here on the lists would like to
see for example 

  a) how well can it work in routers to shift packet drops to low-prio packets
  b) whats the evidence that it's good for video quality
  c) How should we do the marking best going forward

Then please let us know. We'd be very interested to provide insight into the parts of
those pieces that we have also worked out or discuss the ones we have not (primarily c ;-)

Thanks!
    Toerless

In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>

On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> Hi all,
> 
> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> 
> Best regards,
> CJ
> 
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
> Sent: Friday, February 08, 2013 6:25 PM
> To: Cheng-Jia Lai (chelai)
> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> 
> 
> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> has been successfully submitted by Cheng-Jia Lai and posted to the
> IETF repository.
> 
> Filename:	 draft-lai-tsvwg-normalizer
> Revision:	 00
> Title:		 Normalization Marker for AF PHB Group in DiffServ
> Creation date:	 2013-02-08
> WG ID:		 Individual Submission
> Number of pages: 15
> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> 
> 
> Abstract:
>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>    useful for traffic conditioning with Active Queue Management (AQM)
>    when the AF PHB group is deployed for multimedia service classes that
>    carry video packets with advanced codec technologies.  Using NM can
>    offer an incentive that is however missing in today's ecosystem of
>    practical deployment, i.e., when congestion occurs in the AF PHB
>    group, the network should allow a video codec to benefit from its
>    effort of generating finer layers of intra-flow precedence (IFP) with
>    discardable packets in a more balanced distribution.
> 
>                                                                                   
> 
> 
> The IETF Secretariat
> 

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


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On Wed, Jul 10, 2013 at 11:01 AM, Toerless Eckert (eckert)
<eckert@cisco.com> wrote:
> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.

I've begun to look at webrtc of late in the context of fq_codel.

One thing that might work would be ECN marking important frames and
not ECN marking
others. Problem overall is that ecn can be gamed and is largely not
enabled, and so on...

> One of the main concerns raised against such schemes was one of fairness.=
 If flows
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.

same gaming problem...

> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
>  so this is not theoretical).

In the case of the fq_codel algorithm I can see a "delayed drop"
mechanism happening,
where upon deciding to drop, it could defer a drop up to a few packets
based on the classification
of the packet, say 3 packets at most. Since flows are already
identified, this is kind of easy
by adding another state variable...

At the moment what I have deployed and tested most is a simple 3 tier
fq_codel based shaper,
which has a priority, best effort, and background queue. The priority
queue is generally only DNS traffic,
the background is only CS1, and fq_codel does such a good job with
sparse streams in
the general case that for VOIP, etc on the networks I fiddle with,
there is no point in having
much more than the BE queue. And presently all other markings are ignored.

video is another story. (or could be, I don't know enough about
webrtc's behaviors to
"see" what will happen)

So I can see trying to utilize the AF markings as you describe in your
draft in codel's
drop decision. While this is also subject to being gamed, the
disincentives are slight.

This is a little different from using "drop probability" as the
decider, however, but I
would hope that this mapping of the old AF style random drop ideas
into the new style delay
ideas could work...

And what codel depends on is a definition of "TCP friendly". That if a
stream is not
going to behave in a tcp friendly manner, then an entirely different
strategy for such
flows seems necessary, and glomming on some additional state isn't going to
be the right thing.

> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example
>
>   a) how well can it work in routers to shift packet drops to low-prio pa=
ckets

I guess where I fall apart here is how to shift the encoders to send thinne=
r
streams at  any level of packet drop.

>   b) whats the evidence that it's good for video quality
>   c) How should we do the marking best going forward

> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)

I'd be interested in specific codecs, example video streams, and test
scripts to try
and fiddle with this, as well as viable metrics.

>
> Thanks!
>     Toerless
>
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>
>> Best regards,
>> CJ
>>
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>
>>
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>
>> Filename:      draft-lai-tsvwg-normalizer
>> Revision:      00
>> Title:                 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:         2013-02-08
>> WG ID:                 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>
>>
>> Abstract:
>>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>    useful for traffic conditioning with Active Queue Management (AQM)
>>    when the AF PHB group is deployed for multimedia service classes that
>>    carry video packets with advanced codec technologies.  Using NM can
>>    offer an incentive that is however missing in today's ecosystem of
>>    practical deployment, i.e., when congestion occurs in the AF PHB
>>    group, the network should allow a video codec to benefit from its
>>    effort of generating finer layers of intra-flow precedence (IFP) with
>>    discardable packets in a more balanced distribution.
>>
>>
>>
>>
>> The IETF Secretariat
>>
>
> --
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From chelai@cisco.com  Wed Jul 10 16:43:00 2013
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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: Dave Taht <dave.taht@gmail.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
Thread-Index: AQHOfaBDTO2Q2/ChiUO6fKWGKHceJZleig/Q
Date: Wed, 10 Jul 2013 23:42:52 +0000
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References: <20130710180147.GA614@cisco.com> <CAA93jw68E5DgtzvE5N=2-QnGzZQzYQkevFwqWmiSGxiBZk0h4Q@mail.gmail.com>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "aqm@ietf.org" <aqm@ietf.org>
Subject: Re: [rtcweb] [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Yes, NM could also work with ECN (and it should be made so). In fact, we di=
d implementation using WRED on routers that support ECN markings. If RTP en=
dpoints support ECN markings, WRED will mark ECN instead of dropping the pa=
ckets as avg Q size is between min & max threshold. But it still suffers fr=
om unfairness at congestion, where the avg Q size starts to exceed the max =
threshold for the lowest priority packet class...

We did experience a hard time trying to properly configure WRED parameters =
for preferential dropping of video packets with our NM implementation... It=
 is a good insight you said here: maybe it is worth exploring how our work =
can perhaps use other AQM algorithms like CoDel, PIE, etc.

In fact, NM doesn't need specific AQM to drop by probabilities... it will b=
e interesting to see how it can work with delay-based dropping. However, NM=
 does require some "weighted" version of those AQMs. Hopefully a weighted v=
ersion (if any) won't introduce new parameters (or too many) to configure t=
he AQM.

Regards,
CJ


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of D=
ave Taht
Sent: Wednesday, July 10, 2013 12:02 PM
To: Toerless Eckert (eckert)
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

On Wed, Jul 10, 2013 at 11:01 AM, Toerless Eckert (eckert)
<eckert@cisco.com> wrote:
> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.

I've begun to look at webrtc of late in the context of fq_codel.

One thing that might work would be ECN marking important frames and
not ECN marking
others. Problem overall is that ecn can be gamed and is largely not
enabled, and so on...

> One of the main concerns raised against such schemes was one of fairness.=
 If flows
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.

same gaming problem...

> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
>  so this is not theoretical).

In the case of the fq_codel algorithm I can see a "delayed drop"
mechanism happening,
where upon deciding to drop, it could defer a drop up to a few packets
based on the classification
of the packet, say 3 packets at most. Since flows are already
identified, this is kind of easy
by adding another state variable...

At the moment what I have deployed and tested most is a simple 3 tier
fq_codel based shaper,
which has a priority, best effort, and background queue. The priority
queue is generally only DNS traffic,
the background is only CS1, and fq_codel does such a good job with
sparse streams in
the general case that for VOIP, etc on the networks I fiddle with,
there is no point in having
much more than the BE queue. And presently all other markings are ignored.

video is another story. (or could be, I don't know enough about
webrtc's behaviors to
"see" what will happen)

So I can see trying to utilize the AF markings as you describe in your
draft in codel's
drop decision. While this is also subject to being gamed, the
disincentives are slight.

This is a little different from using "drop probability" as the
decider, however, but I
would hope that this mapping of the old AF style random drop ideas
into the new style delay
ideas could work...

And what codel depends on is a definition of "TCP friendly". That if a
stream is not
going to behave in a tcp friendly manner, then an entirely different
strategy for such
flows seems necessary, and glomming on some additional state isn't going to
be the right thing.

> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example
>
>   a) how well can it work in routers to shift packet drops to low-prio pa=
ckets

I guess where I fall apart here is how to shift the encoders to send thinne=
r
streams at  any level of packet drop.

>   b) whats the evidence that it's good for video quality
>   c) How should we do the marking best going forward

> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)

I'd be interested in specific codecs, example video streams, and test
scripts to try
and fiddle with this, as well as viable metrics.

>
> Thanks!
>     Toerless
>
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>
>> Best regards,
>> CJ
>>
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>
>>
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>
>> Filename:      draft-lai-tsvwg-normalizer
>> Revision:      00
>> Title:                 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:         2013-02-08
>> WG ID:                 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>
>>
>> Abstract:
>>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>    useful for traffic conditioning with Active Queue Management (AQM)
>>    when the AF PHB group is deployed for multimedia service classes that
>>    carry video packets with advanced codec technologies.  Using NM can
>>    offer an incentive that is however missing in today's ecosystem of
>>    practical deployment, i.e., when congestion occurs in the AF PHB
>>    group, the network should allow a video codec to benefit from its
>>    effort of generating finer layers of intra-flow precedence (IFP) with
>>    discardable packets in a more balanced distribution.
>>
>>
>>
>>
>> The IETF Secretariat
>>
>
> --
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From stefan.lk.hakansson@ericsson.com  Wed Jul 10 23:43:58 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-use-cases-and-requirements - HTTP Only Firewall.
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Hi Andrew,=0A=
=0A=
thanks for catching this!=0A=
=0A=
F37 currently reads:=0A=
=0A=
"The browser must be able to send streams and data to a peer in the =0A=
presence of FWs that only allows http(s) traffic."=0A=
=0A=
Would a change to:=0A=
=0A=
"The browser must be able to send streams and data to a peer in the =0A=
presence of FWs that only allows traffic via a HTTP Proxy."=0A=
=0A=
be correct?=0A=
=0A=
Br,=0A=
Stefan=0A=
=0A=
=0A=
On 7/10/13 4:00 PM, Hutton, Andrew wrote:=0A=
> Hi,=0A=
>=0A=
> Regarding the use case in 3.2.3 covering a HTTP only FW the text=0A=
> states:=0A=
>=0A=
> "This use-case is almost identical to the Simple Video Communication=0A=
> Service use-case (Section 3.2.1).  The difference is that one of the=0A=
> users is behind a FW that only allows http traffic".=0A=
>=0A=
> This needs to be changed to allow for the common case when a HTTP=0A=
> Proxy is deployed so I suggest changing the last sentence to=0A=
>=0A=
> "The difference is that one of the users is behind a FW that only=0A=
> allows traffic via a HTTP Proxy".=0A=
>=0A=
> There also needs to be a corresponding change to requirement F37.=0A=
>=0A=
> I believe we have previously discussed changing this but have to=0A=
> admit I could not fine the e-mail chain so maybe it was during a=0A=
> meeting.=0A=
>=0A=
> Regards Andy=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>> -----Original Message----- From: rtcweb-bounces@ietf.org=0A=
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan H=E5kansson LK=0A=
>> Sent: 27 June 2013 09:50 To: rt >> "rtcweb@ietf.org" Subject:=0A=
>> [rtcweb] Fwd: New Version Notification for draft-ietf-rtcweb-=0A=
>> use-cases-and-requirements-11.txt=0A=
>>=0A=
>> From the change log:=0A=
>>=0A=
>> o  Described that the API requirements are really from a W3C=0A=
>> perspective and are supplied as an appendix in the introduction.=0A=
>> Moved API requirements to an Appendix.=0A=
>>=0A=
>> o  Removed the "Conventions" section with the key-words and=0A=
>> reference to RFC2119.  Also changed uppercase MUST's/SHOULD's to=0A=
>> lowercase.=0A=
>>=0A=
>> o  Added a note on the proposed use of the document to the=0A=
>> introduction.=0A=
>>=0A=
>> o  Removed the note talking about WS from the "FW that only allows=0A=
>> http" use-case.=0A=
>>=0A=
>> o  Removed the word "Skype" that was used as example in one of the=0A=
>> use-cases.=0A=
>>=0A=
>> o  Clarified F3 (the req saying the everything the browser sends=0A=
>> must be rate controlled).=0A=
>>=0A=
>> o  Removed the TBD saying we need to define reasonable levels from=0A=
>> the requirement saying that quality must be good even in presence=0A=
>> of packet losses (F5), and changed "must" to "should" (Based on a=0A=
>> list discussion involving Bernard).=0A=
>>=0A=
>> o  Removed F6 ("The browser must be able to handle high loss and=0A=
>> jitter levels in a graceful way."), also after a list discussion.=0A=
>>=0A=
>> o  Clarified F7 (used to say that the browser must support fast=0A=
>> stream switches, now says that reference frames must be inserted=0A=
>> when requested). o  Removed the questions from F9 (echo=0A=
>> cancellation), F10 (syncronization), F21 (telephony codec).=0A=
>>=0A=
>> o  Exchanged "restrictive firewalls" for "limited middleboxes" in=0A=
>> F19 (as proposed by Martin).=0A=
>>=0A=
>> o  Expanded DTMF and IVR in F22 (proposed by Martin)=0A=
>>=0A=
>> o  Added ref to RFC5405 in F23 (proposed by Lars Eggert).=0A=
>>=0A=
>> o  Exchanged "service provided" for "web application" in F32.=0A=
>>=0A=
>> o  Changed the text in 3.2.1 that motivates F36 (new text "It is=0A=
>> essential that media and data be encrypted, authenticated ... bound=0A=
>> to the user identity."); and rewrote F36, included a ref to=0A=
>> RFC5479.=0A=
>>=0A=
>> o  Changed "quality of service" to "quality of experience" in F38.=0A=
>>=0A=
>> o  Added F39.=0A=
>>=0A=
>> o  Used new formulation of A17 (proposed by Martin).=0A=
>>=0A=
>> o  Updated A20.=0A=
>>=0A=
>> o  Updated A25.=0A=
>>=0A=
>> Things that have not been done:=0A=
>>=0A=
>> - No use-case on emergency services added (as said already in=0A=
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07253.html)=0A=
>>=0A=
>> - No use-case on real-time text added (as said already in=0A=
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07254.html)=0A=
>>=0A=
>> - No clarification on what solution(s) related to multiple=0A=
>> resolutions of the same content added (discussed in=0A=
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07256.html=0A=
>> but no input received).=0A=
>>=0A=
>> - The order of the requirements (Fn) is still a mess - but I kept=0A=
>> it as is for this version to make diffing simpler. To be fixed in=0A=
>> an upcoming version.=0A=
>>=0A=
>> Stefan=0A=
>>=0A=
>>=0A=
>> -------- Original Message -------- Subject: New Version=0A=
>> Notification for=0A=
>> draft-ietf-rtcweb-use-cases-and-requirements-11.txt Date: 10:40=0A=
>> From: internet-drafts@ietf.org <internet-drafts@ietf.org> To:=0A=
>> Christer Holmberg <christer.holmberg@ericsson.com>,    G=F6ran=0A=
>> Eriksson AP <goran.ap.eriksson@ericsson.com>,    Stefan H=E5kansson=0A=
>> LK <stefan.lk.hakansson@ericsson.com>,    G=F6ran Eriksson AP=0A=
>> <goran.ap.eriksson@ericsson.com>=0A=
>>=0A=
>>=0A=
>> A new version of I-D,=0A=
>> draft-ietf-rtcweb-use-cases-and-requirements- 11.txt has been=0A=
>> successfully submitted by Christer Holmberg and posted to the IETF=0A=
>> repository.=0A=
>>=0A=
>> Filename:	 draft-ietf-rtcweb-use-cases-and-requirements Revision:=0A=
>> 11 Title:		 Web Real-Time Communication Use-cases and Requirements=0A=
>> Creation date:	 2013-06-27 Group:		 rtcweb Number of pages: 30=0A=
>> URL:=0A=
>> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-=0A=
>>=0A=
>>=0A=
requirements-11.txt=0A=
>> Status:=0A=
>> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-=0A=
>> requirements Htmlized:=0A=
>> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-=0A=
>> requirements-11 Diff:=0A=
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-use-cases-and-=0A=
>> requirements-11=0A=
>>=0A=
>> Abstract: This document describes web based real-time communication=0A=
>> use- cases. Requirements on the browser functionality are derived=0A=
>> from use- cases.=0A=
>>=0A=
>>=0A=
>>=0A=
>>=0A=
>>=0A=
>> The IETF Secretariat=0A=
>>=0A=
>>=0A=
>>=0A=
>>=0A=
>> _______________________________________________ rtcweb mailing=0A=
>> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

From magnus.westerlund@ericsson.com  Thu Jul 11 01:28:08 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Question about the status of various drafts
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On 2013-07-05 02:47, Cullen Jennings wrote:
> draft-ietf-rtcweb-rtp-usage		40

I disagree that that this is as low as 40%. There are some few thorny
issues that might take some discussion to resolve. However, most things
are from an authors perspective quite solid. I would say 60-70%. My main
caveat is that I haven't seen that many reviews.

You can expect an update prior to the deadline resolving some of the
stated open issues and dealing with issues from Bernard's review.

> draft-ietf-avtcore-6222bis		80

I expect this to be approved in a week or two, thus 90

> draft-ietf-avtcore-avp-codecs		80

Approved thus only publication left thus 95

> draft-ietf-avtext-multiple-clock-rates		70

This one is 90 as it is ready for publication request. It is awaiting
the AVTEXT chair to act on it.

> draft-lennox-rtcweb-rtp-media-type-mux 		80

I wonder why this draft is listed here. Isn't the relevant content of
this gone into the AVTCORE drafts:
- draft-ietf-avtcore-multi-media-rtp-session
- draft-ietf-avtcore-rtp-multi-stream
?


> draft-ietf-avtcore-multi-media-rtp-session 	70
> draft-lennox-avtcore-rtp-multi-stream		70

There is a WG version of this document
draft-ietf-avtcore-rtp-multi-stream which in fact is being split into
one recommendation and one optimization part. Updates to be submitted
prior to deadline.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From andrew.hutton@siemens-enterprise.com  Thu Jul 11 01:28:35 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-use-cases-and-requirements - HTTP Only Firewall.
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Sounds good to me.

Regards
Andy

> -----Original Message-----
> From: Stefan H=E5kansson LK [mailto:stefan.lk.hakansson@ericsson.com]
> Sent: 11 July 2013 07:44
> To: Hutton, Andrew
> Cc: rt >> "rtcweb@ietf.org"
> Subject: Re: draft-ietf-rtcweb-use-cases-and-requirements - HTTP Only
> Firewall.
>=20
> Hi Andrew,
>=20
> thanks for catching this!
>=20
> F37 currently reads:
>=20
> "The browser must be able to send streams and data to a peer in the
> presence of FWs that only allows http(s) traffic."
>=20
> Would a change to:
>=20
> "The browser must be able to send streams and data to a peer in the
> presence of FWs that only allows traffic via a HTTP Proxy."
>=20
> be correct?
>=20
> Br,
> Stefan
>=20
>=20
> On 7/10/13 4:00 PM, Hutton, Andrew wrote:
> > Hi,
> >
> > Regarding the use case in 3.2.3 covering a HTTP only FW the text
> > states:
> >
> > "This use-case is almost identical to the Simple Video Communication
> > Service use-case (Section 3.2.1).  The difference is that one of the
> > users is behind a FW that only allows http traffic".
> >
> > This needs to be changed to allow for the common case when a HTTP
> > Proxy is deployed so I suggest changing the last sentence to
> >
> > "The difference is that one of the users is behind a FW that only
> > allows traffic via a HTTP Proxy".
> >
> > There also needs to be a corresponding change to requirement F37.
> >
> > I believe we have previously discussed changing this but have to
> > admit I could not fine the e-mail chain so maybe it was during a
> > meeting.
> >
> > Regards Andy
> >
> >
> >
> >
> >> -----Original Message----- From: rtcweb-bounces@ietf.org
> >> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan H=E5kansson LK
> >> Sent: 27 June 2013 09:50 To: rt >> "rtcweb@ietf.org" Subject:
> >> [rtcweb] Fwd: New Version Notification for draft-ietf-rtcweb-
> >> use-cases-and-requirements-11.txt
> >>
> >> From the change log:
> >>
> >> o  Described that the API requirements are really from a W3C
> >> perspective and are supplied as an appendix in the introduction.
> >> Moved API requirements to an Appendix.
> >>
> >> o  Removed the "Conventions" section with the key-words and
> >> reference to RFC2119.  Also changed uppercase MUST's/SHOULD's to
> >> lowercase.
> >>
> >> o  Added a note on the proposed use of the document to the
> >> introduction.
> >>
> >> o  Removed the note talking about WS from the "FW that only allows
> >> http" use-case.
> >>
> >> o  Removed the word "Skype" that was used as example in one of the
> >> use-cases.
> >>
> >> o  Clarified F3 (the req saying the everything the browser sends
> >> must be rate controlled).
> >>
> >> o  Removed the TBD saying we need to define reasonable levels from
> >> the requirement saying that quality must be good even in presence
> >> of packet losses (F5), and changed "must" to "should" (Based on a
> >> list discussion involving Bernard).
> >>
> >> o  Removed F6 ("The browser must be able to handle high loss and
> >> jitter levels in a graceful way."), also after a list discussion.
> >>
> >> o  Clarified F7 (used to say that the browser must support fast
> >> stream switches, now says that reference frames must be inserted
> >> when requested). o  Removed the questions from F9 (echo
> >> cancellation), F10 (syncronization), F21 (telephony codec).
> >>
> >> o  Exchanged "restrictive firewalls" for "limited middleboxes" in
> >> F19 (as proposed by Martin).
> >>
> >> o  Expanded DTMF and IVR in F22 (proposed by Martin)
> >>
> >> o  Added ref to RFC5405 in F23 (proposed by Lars Eggert).
> >>
> >> o  Exchanged "service provided" for "web application" in F32.
> >>
> >> o  Changed the text in 3.2.1 that motivates F36 (new text "It is
> >> essential that media and data be encrypted, authenticated ... bound
> >> to the user identity."); and rewrote F36, included a ref to
> >> RFC5479.
> >>
> >> o  Changed "quality of service" to "quality of experience" in F38.
> >>
> >> o  Added F39.
> >>
> >> o  Used new formulation of A17 (proposed by Martin).
> >>
> >> o  Updated A20.
> >>
> >> o  Updated A25.
> >>
> >> Things that have not been done:
> >>
> >> - No use-case on emergency services added (as said already in
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07253.html)
> >>
> >> - No use-case on real-time text added (as said already in
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07254.html)
> >>
> >> - No clarification on what solution(s) related to multiple
> >> resolutions of the same content added (discussed in
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07256.html
> >> but no input received).
> >>
> >> - The order of the requirements (Fn) is still a mess - but I kept
> >> it as is for this version to make diffing simpler. To be fixed in
> >> an upcoming version.
> >>
> >> Stefan
> >>
> >>
> >> -------- Original Message -------- Subject: New Version
> >> Notification for
> >> draft-ietf-rtcweb-use-cases-and-requirements-11.txt Date: 10:40
> >> From: internet-drafts@ietf.org <internet-drafts@ietf.org> To:
> >> Christer Holmberg <christer.holmberg@ericsson.com>,    G=F6ran
> >> Eriksson AP <goran.ap.eriksson@ericsson.com>,    Stefan H=E5kansson
> >> LK <stefan.lk.hakansson@ericsson.com>,    G=F6ran Eriksson AP
> >> <goran.ap.eriksson@ericsson.com>
> >>
> >>
> >> A new version of I-D,
> >> draft-ietf-rtcweb-use-cases-and-requirements- 11.txt has been
> >> successfully submitted by Christer Holmberg and posted to the IETF
> >> repository.
> >>
> >> Filename:	 draft-ietf-rtcweb-use-cases-and-requirements
> Revision:
> >> 11 Title:		 Web Real-Time Communication Use-cases and
> Requirements
> >> Creation date:	 2013-06-27 Group:		 rtcweb Number of
> pages: 30
> >> URL:
> >> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-
> >>
> >>
> requirements-11.txt
> >> Status:
> >> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-
> >> requirements Htmlized:
> >> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-
> >> requirements-11 Diff:
> >> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-use-cases-and-
> >> requirements-11
> >>
> >> Abstract: This document describes web based real-time communication
> >> use- cases. Requirements on the browser functionality are derived
> >> from use- cases.
> >>
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
> >>
> >>
> >>
> >>
> >> _______________________________________________ rtcweb mailing
> >> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> >


From andrew.hutton@siemens-enterprise.com  Thu Jul 11 03:19:17 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Cullen Jennings <fluffy@iii.ca>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Question about the status of various drafts
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Date: Thu, 11 Jul 2013 10:19:07 +0000
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There are things missing from the list.

For example the charter and the requirements indicate that we need to addre=
ss the firewall issue so we need a draft for that.

I hope we will adopt draft-hutton-rtcweb-nat-firewall-considerations to cov=
er this and of course assuming we do then I think it is nearly done but oth=
ers might think differently.=20

I have not been through the requirements document to see if there are other=
 things missing.

Regards
Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: 05 July 2013 01:47
> To: rtcweb@ietf.org
> Subject: [rtcweb] Question about the status of various drafts
>=20
>=20
> As part of putting together some chair slides for Berlin, I'm trying to
> get a very rough idea of where things are. I'd like to get folks input
> on what percent done various drafts are where Done is a published RFC.
>=20
> This is nearly impossible to do at any level of accuracy but I do want
> to get a vague idea. I took a very rough stab below just to have a
> starting point to get the conversation going. I'm sure my numbers are
> mostly wrong but I have tried to ask a few people and sort of take the
> central cluster of the guess. If you think I am way out, please provide
> some feedback of what you think the number actually is.
>=20
> I'm happy to get feedback on specific drafts or feedback of the form
> they are all too low or too high. I do encourage people to
> realistically look at the data tracker to see how long other drafts
> have taken for each stage to get a baseline instead of just going with
> their gut feel.
>=20
> Thanks, Cullen
>=20
> PS - if you are an author and look at the number next to your draft and
> think it is way off, please please, say something.
>=20
>=20
>=20
> draft-ietf-rtcweb-overview		80
> draft-ietf-rtcweb-use-cases-and-requirements	70
> http://dev.w3.org/2011/webrtc/editor/getusermedia.html	70
> draft-burnett-rtcweb-constraints-registry		80
> http://dev.w3.org/2011/webrtc/editor/webrtc.html	50
> draft-nandakumar-rtcweb-stun-uri		80
> draft-petithuguenin-behave-turn-uris		80
> draft-ietf-rtcweb-security		80
> draft-ietf-rtcweb-security-arch		80
> draft-muthu-behave-consent-freshness-02 	50
> draft-ietf-rtcweb-data-channel		70
> draft-ietf-mmusic-sctp-sdp		70
> draft-jesup-rtcweb-data-protocol 		50
> draft-ietf-tsvwg-sctp-dtls-encaps		70
> draft-ietf-rtcweb-rtp-usage		40
> draft-ietf-avtcore-6222bis		80
> draft-ietf-avtcore-avp-codecs		80
> draft-ietf-avtcore-srtp-encrypted-header-ext	80
> draft-ietf-avtext-multiple-clock-rates		70
> draft-lennox-rtcweb-rtp-media-type-mux 		80
> draft-ietf-avtcore-rtp-circuit-breakers		70
> draft-ietf-avtcore-multi-media-rtp-session 	70
> draft-lennox-avtcore-rtp-multi-stream		70
> draft-ietf-rtcweb-jsep		50
> draft-ietf-mmusic-msid		70
> draft-rescorla-mmusic-ice-trickle		25
> draft-ietf-mmusic-sdp-bundle-negotiation	50
> draft-nandakumar-mmusic-sdp-mux-attributes	25
> draft-dhesikan-tsvwg-rtcweb-qos		10 or 90
> draft-ietf-rtcweb-audio		50
> draft-ietf-payload-rtp-opus		90
> draft-ietf-payload-vp8-08		95
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ted.ietf@gmail.com  Thu Jul 11 09:51:30 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] Draft agenda for IETF 87
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--047d7bdc10e446205004e13f33ae
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Greetings,

Below is an initial draft agenda for the upcoming meeting.   Since we have
not yet reached the draft deadline (which is the 15th), there may be new
drafts or updates that result in changes.  We did already receive requests
for NAT/Firewall traversal discussion, and the chairs will be working with
the document authors to get them considered in the appropriate groups.

As folks have probably noticed, we are meeting Thursday and Friday, after
the MMUSIC sessions are complete (they are Tuesday and Wednesday). This
should allow us to discuss the results on our first day.

Please send feedback or change proposals to the list.

thanks,

Ted and Cullen

Day 1:

Should SDES be part of  WebRTC security practice and, if so, how?
Presentations: 30 minutes
Discussion:  40 minutes

Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
Presentation: 30 minutes
Discussion: 30 minutes

Security document updates
Presentation: 10 minutes
Discussion: 10 minutes

Day 2:

Chair Discussion:  10 minutes

Use Case Requirements updates:
Issues list presentation: 20 minutes
Discussion: 20 minutes

Data channel:
Issues list presentation:  45 minutes
Discussion: 45 minutes

--047d7bdc10e446205004e13f33ae
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Greetings,<br><br>Below is an initial draft agenda for the upcoming meeting=
.=A0=A0 Since we have not yet reached the draft deadline (which is the 15th=
), there may be new drafts or updates that result in changes.=A0 We did alr=
eady receive requests for NAT/Firewall traversal discussion, and the chairs=
 will be working with the document authors to get them considered in the ap=
propriate groups.<br>
<br>As folks have probably noticed, we are meeting Thursday and Friday,=20
after the MMUSIC sessions are complete (they are Tuesday and Wednesday). Th=
is should allow us to discuss the results on our first day. <br><br>Please =
send feedback or change proposals to the list.<br><br>thanks,<br><br>Ted an=
d Cullen<br>
<br>Day 1:<br><br>
Should SDES be part of=A0 WebRTC security practice and, if so, how?<br>Pres=
entations: 30 minutes<br>Discussion:=A0 40 minutes<br><br>Post-Plan A/Plan =
B MMUSIC discussion of impact to RTCWEB documents<br>Presentation: 30 minut=
es<br>

Discussion: 30 minutes<br><br>Security document updates<br>Presentation: 10=
 minutes<br>Discussion: 10 minutes<br><br>Day 2:<br><br>Chair Discussion:=
=A0 10 minutes<br><br>Use Case Requirements updates:<br>Issues list present=
ation: 20 minutes<br>
Discussion: 20 minutes<br>
<br>Data channel:<br>Issues list presentation:=A0 45 minutes<br>Discussion:=
 45 minutes<br><br><br>

--047d7bdc10e446205004e13f33ae--

From emcho@sip-communicator.org  Thu Jul 11 10:35:55 2013
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Date: Thu, 11 Jul 2013 19:35:34 +0200
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Subject: Re: [rtcweb] Draft agenda for IETF 87
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--001a11c24eb04e178204e13fd2cc
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Hey Ted,

I am sorry I must have missed your call for presentations.

In you last mail on the subject you mentioned that we will be discussing No
Plan in Berlin together with plans A and B. Could we please add this to the
agenda?

Emil

--sent from my mobile
On Jul 11, 2013 6:51 PM, "Ted Hardie" <ted.ietf@gmail.com> wrote:

> Greetings,
>
> Below is an initial draft agenda for the upcoming meeting.   Since we have
> not yet reached the draft deadline (which is the 15th), there may be new
> drafts or updates that result in changes.  We did already receive requests
> for NAT/Firewall traversal discussion, and the chairs will be working with
> the document authors to get them considered in the appropriate groups.
>
> As folks have probably noticed, we are meeting Thursday and Friday, after
> the MMUSIC sessions are complete (they are Tuesday and Wednesday). This
> should allow us to discuss the results on our first day.
>
> Please send feedback or change proposals to the list.
>
> thanks,
>
> Ted and Cullen
>
> Day 1:
>
> Should SDES be part of  WebRTC security practice and, if so, how?
> Presentations: 30 minutes
> Discussion:  40 minutes
>
> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> Presentation: 30 minutes
> Discussion: 30 minutes
>
> Security document updates
> Presentation: 10 minutes
> Discussion: 10 minutes
>
> Day 2:
>
> Chair Discussion:  10 minutes
>
> Use Case Requirements updates:
> Issues list presentation: 20 minutes
> Discussion: 20 minutes
>
> Data channel:
> Issues list presentation:  45 minutes
> Discussion: 45 minutes
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c24eb04e178204e13fd2cc
Content-Type: text/html; charset=ISO-8859-1
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<p dir=3D"ltr">Hey Ted,</p>
<p dir=3D"ltr">I am sorry I must have missed your call for presentations.</=
p>
<p dir=3D"ltr">In you last mail on the subject you mentioned that we will b=
e discussing No Plan in Berlin together with plans A and B. Could we please=
 add this to the agenda?</p>
<p dir=3D"ltr">Emil</p>
<p dir=3D"ltr">--sent from my mobile</p>
<div class=3D"gmail_quote">On Jul 11, 2013 6:51 PM, &quot;Ted Hardie&quot; =
&lt;<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>&gt; wrote:=
<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Greetings,<br><br>Below is an initial draft agenda for the upcoming meeting=
.=A0=A0 Since we have not yet reached the draft deadline (which is the 15th=
), there may be new drafts or updates that result in changes.=A0 We did alr=
eady receive requests for NAT/Firewall traversal discussion, and the chairs=
 will be working with the document authors to get them considered in the ap=
propriate groups.<br>

<br>As folks have probably noticed, we are meeting Thursday and Friday,=20
after the MMUSIC sessions are complete (they are Tuesday and Wednesday). Th=
is should allow us to discuss the results on our first day. <br><br>Please =
send feedback or change proposals to the list.<br><br>thanks,<br><br>Ted an=
d Cullen<br>

<br>Day 1:<br><br>
Should SDES be part of=A0 WebRTC security practice and, if so, how?<br>Pres=
entations: 30 minutes<br>Discussion:=A0 40 minutes<br><br>Post-Plan A/Plan =
B MMUSIC discussion of impact to RTCWEB documents<br>Presentation: 30 minut=
es<br>


Discussion: 30 minutes<br><br>Security document updates<br>Presentation: 10=
 minutes<br>Discussion: 10 minutes<br><br>Day 2:<br><br>Chair Discussion:=
=A0 10 minutes<br><br>Use Case Requirements updates:<br>Issues list present=
ation: 20 minutes<br>

Discussion: 20 minutes<br>
<br>Data channel:<br>Issues list presentation:=A0 45 minutes<br>Discussion:=
 45 minutes<br><br><br>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--001a11c24eb04e178204e13fd2cc--

From martin.thomson@gmail.com  Thu Jul 11 11:08:13 2013
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I can see some implied outcomes for some of the items.  Would it help
to list some (aspirational) goals.

On 11 July 2013 09:51, Ted Hardie <ted.ietf@gmail.com> wrote:
> Should SDES be part of  WebRTC security practice and, if so, how?
> Presentations: 30 minutes
> Discussion:  40 minutes

Are the chairs confident that this topic can be resolved in this time?
 We managed to fritter a similar amount of time away without
conclusion in the past.  I can see how you plan to accommodate
overruns, but that just opens the possibility more time-wasting.  How
do we ensure that this discussion actually concludes?

> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> Presentation: 30 minutes

How are you planning to prepare for this?  Do you have presenter
candidates lined up?

> Chair Discussion:  10 minutes

I'm curious how you intend to spend this time.

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87
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inline=20

On Jul 11, 2013, at 11:08 AM, Martin Thomson <martin.thomson@gmail.com> wro=
te:

> I can see some implied outcomes for some of the items.  Would it help
> to list some (aspirational) goals.
>=20
> On 11 July 2013 09:51, Ted Hardie <ted.ietf@gmail.com> wrote:
>> Should SDES be part of  WebRTC security practice and, if so, how?
>> Presentations: 30 minutes
>> Discussion:  40 minutes
>=20
> Are the chairs confident that this topic can be resolved in this time?
> We managed to fritter a similar amount of time away without
> conclusion in the past.  I can see how you plan to accommodate
> overruns, but that just opens the possibility more time-wasting.  How
> do we ensure that this discussion actually concludes?

Thought on how to make that happen? Time needed ?


>=20
>> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
>> Presentation: 30 minutes
>=20
> How are you planning to prepare for this?  Do you have presenter
> candidates lined up?

This is a very draft agenda - part of the issues is we are not even at the =
-00 draft deadline yet so I expect that by next week we should be able to f=
igure out more details. We have scheduled the MMUSIC meetings before the RT=
CWeb meetings in Berlin so at some level, some of this may change after the=
 MMUSIC meetings happen. Anyways, expect more details after we get to read =
all the drafts and expect us to be a bit dynamic on this given what happens=
 in the MMUSIC meeting.=20

>=20
>> Chair Discussion:  10 minutes
>=20
> I'm curious how you intend to spend this time.

We would like to inform folks of a few things happening in other WG (TURN, =
NAT, BEHAVE etc) but we would also like to have a discussion about what it =
takes to get to "Done" for the first iteration of all this stuff. We need t=
o get WG feedback on what work is left and what needs to be prioritized.=20


> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 11 July 2013 11:24, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> On Jul 11, 2013, at 11:08 AM, Martin Thomson <martin.thomson@gmail.com> w=
rote:
>> On 11 July 2013 09:51, Ted Hardie <ted.ietf@gmail.com> wrote:
>>> Should SDES be part of  WebRTC security practice and, if so, how?
>>> Presentations: 30 minutes
>>> Discussion:  40 minutes
>>
>> Are the chairs confident that this topic can be resolved in this time?
>> We managed to fritter a similar amount of time away without
>> conclusion in the past.  I can see how you plan to accommodate
>> overruns, but that just opens the possibility more time-wasting.  How
>> do we ensure that this discussion actually concludes?
>
> Thought on how to make that happen? Time needed ?

There are two aspects of this that I, personally, want to see resolved
before I'm happy that we can conclude on this:
 - a story for the gateway-to-other-stuff scenarios (where other-stuff
is mostly just all-the-existing-stuff)
 - a story for conference scenarios

The first two could conclude that we need to re-encrypt at these
points, but that would require some justification.  Or we could
conclude that we need SDES, or maybe that DTLS-SRTP+EKT is the right
answer.

I'm stretching my recall capacity, but I recall promises from several
people to present on various aspects.  If those presentations cover
the ground well enough, then we can go into a discussion
well-prepared.  Time needed will then depend on the time that each
presenter needs.

If we can get to the point where the space is well-enough understood,
we should be able to conclude.  That does require a modest amount of
time for each participant to make an impassioned plea at the mic.  And
pontification takes time.

We may need to consider the alternative decision making process here.

>>> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
>>> Presentation: 30 minutes
>>
>> How are you planning to prepare for this?  Do you have presenter
>> candidates lined up?
>
> This is a very draft agenda - part of the issues is we are not even at th=
e -00 draft deadline yet so I expect that by next week we should be able to=
 figure out more details. We have scheduled the MMUSIC meetings before the =
RTCWeb meetings in Berlin so at some level, some of this may change after t=
he MMUSIC meetings happen. Anyways, expect more details after we get to rea=
d all the drafts and expect us to be a bit dynamic on this given what happe=
ns in the MMUSIC meeting.

You'd best have some candidates lined up to produce some discussion
material in short order.

>>> Chair Discussion:  10 minutes
>>
>> I'm curious how you intend to spend this time.
>
> We would like to inform folks of a few things happening in other WG (TURN=
, NAT, BEHAVE etc) but we would also like to have a discussion about what i=
t takes to get to "Done" for the first iteration of all this stuff. We need=
 to get WG feedback on what work is left and what needs to be prioritized.

Thanks.  That helps.  That might be too short a time for that list.

From fluffy@cisco.com  Thu Jul 11 12:04:59 2013
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On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org>
 wrote:

> Hey Ted,
>=20
> I am sorry I must have missed your call for presentations.
>=20
> In you last mail on the subject you mentioned that we will be discussing =
No Plan in Berlin together with plans A and B. Could we please add this to =
the agenda?
>=20
> Emil
>=20
>=20

No. We believe that conversation needs to happen in the W3C WebRTC WG. I ex=
pect to see a message from W3C chairs on this at some point.=20

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On 11 July 2013 12:04, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org>
>> In you last mail on the subject you mentioned that we will be discussing No Plan in Berlin together with plans A and B. Could we please add this to the agenda?
>
> No. We believe that conversation needs to happen in the W3C WebRTC WG. I expect to see a message from W3C chairs on this at some point.

I'm a little nervous about this.  Where does the decision on the
separation of responsibilities (API vs. SDP) get made?

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Subject: Re: [rtcweb] Draft agenda for IETF 87
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On 7/11/13 14:37, Martin Thomson wrote:
> On 11 July 2013 12:04, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>> On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org>
>>> In you last mail on the subject you mentioned that we will be discussing No Plan in Berlin together with plans A and B. Could we please add this to the agenda?
>> No. We believe that conversation needs to happen in the W3C WebRTC WG. I expect to see a message from W3C chairs on this at some point.
> I'm a little nervous about this.  Where does the decision on the
> separation of responsibilities (API vs. SDP) get made?

Jointly by the chairs of RTCWEB and WebRTC, based on 
<http://www.w3.org/2011/04/webrtc-charter.html> and 
<http://tools.ietf.org/wg/rtcweb/charters>, one would presume.

/a

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On 11 July 2013 12:46, Adam Roach <adam@nostrum.com> wrote:
> Jointly by the chairs of RTCWEB and WebRTC, based on
> <http://www.w3.org/2011/04/webrtc-charter.html> and
> <http://tools.ietf.org/wg/rtcweb/charters>, one would presume.

To be perfectly clear, that's precisely what makes me nervous.

From csp@csperkins.org  Thu Jul 11 14:24:34 2013
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Subject: Re: [rtcweb] #19: Section 7.2: Congestion Control Extensions
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On 20 Apr 2013, at 02:48, rtcweb issue tracker wrote:
> #19: Section 7.2: Congestion Control Extensions
>=20
> 7.2.  RTCP Extensions for Congestion Control
>=20
>    As described in Section 5.1.6, the Temporary Maximum Media Stream =
Bit
>    Rate (TMMBR) request is supported by WebRTC senders.  This request
>    can be used by a media receiver to impose limitations on the media
>    sender based on the receiver's determined bit-rate limitations, to
>    provide a limited means of congestion control.
>=20
>    (tbd: What other RTP/RTCP extensions are needed?)
>=20
>    With proprietary congestion control algorithms issues can arise =
when
>    different algorithms and implementations interact in a =
communication
>    session.  If the different implementations have made different
>    choices in regards to the type of adaptation, for example one =
sender
>    based, and one receiver based, then one could end up in situation
>    where one direction is dual controlled, when the other direction is
>    not controlled.
>=20
>    (tbd: How to ensure that both paths and sender and receiver based
>    solutions can interact)
>=20
> [BA] Some nasty unresolved issues here. In particular, I am concerned
> about potential gaps in worldview between sender-side and =
receiver-side
> congestion control implementations.  Is our goal here to guarantee =
that a sender-side implementation can interoperate with a receiver-side
> implementation? Or that a sender-side implementation has the =
information it needs?


I don't think either is the goal. As we don't have a congestion control =
solution from RMCAT yet, I believe the goal for this draft should be to =
allow signalling of known boundary conditions on the acceptable rate, =
and to ensure the necessary RTP/RTCP features to implement congestion =
control are available in implementations.

TMMBR requests fall into the former category. They're not designed, or =
suitable, for effective congestion control. Rather, they allow =
signalling of known constraints within which congestion control can =
operate. Section 7.1 of the draft, immediately before the text you =
quoted, says this better. I will therefore remove the paragraph about =
TMMBR you quoted from the draft.

The draft says "(tbd: What other RTP/RTCP extensions are needed?)". I =
think this answer here is none: we have RTP/AVPF feedback, RTP header =
extensions, non-compound packets, and the ability to tune the reporting =
interval. These are enough to convey congestion reports in a reasonably =
low-overhead manner. Anything else will depend on the algorithm, and so =
belongs in the draft defining the standard RMCAT congestion control =
algorithm.

The text quoted about proprietary congestion control algorithms, and =
interoperability between sender- and receiver-driven algorithms is a =
real issue, but not one that I think this draft can resolve. I'll move =
the quoted text into Section 7.4, where it fits better, and try to =
clarify that the problem is one that should be addressed by agreement on =
the congestion control algorithm at the signalling layer, and an =
effective standard from RMCAT.

--=20
Colin Perkins
http://csperkins.org/




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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: ken carlberg <carlberg@g11.org.uk>, "Toerless Eckert (eckert)" <eckert@cisco.com>
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Subject: Re: [rtcweb] [tsvwg] Using intelligent dropping to improve video quality	under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Ken,

Thanks very much for the comment and showing an interest in our draft.

Yes, I agree it is beneficial if the intra-flow priorities info can be carr=
ied in some bit field other than DSCP markings. I guess we need to better c=
larify that NM is not necessarily specific to AF PHB but potentially applic=
able to other service classes. (In fact, our implementation did try DPI in =
RTP and H.264 NAL unit header for that, with some limited overheads in pack=
et processing.)

We know there can be a lot of intelligent ways to carrying priorities in pa=
ckets (e.g. your TLV in RTP hdr) which are likely beyond our knowledge and =
expertise, so we just introduced the CB mode in Section 3.1, quoted as foll=
ows:

"When NM operates in "color-blind" (CB) mode, which is optionally
   supported, it reads certain bits field(s) other than the AF-markings
   in the packets to determine the actual drop precedence of that
   packet.  This implies that NM may need DPI in the packets, e.g.
   parsing into H.264 AVC header in each RTP packets, or alternatively
   use some method where the drop precedence is carried from the encoder
   in a customized bits field other than the AF-marking in each packet."

CB might not be the best name... sorry if confusing. Anyway, we hope this a=
llows NM to collaborate better with video endpoints.

Regards,
CJ


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of k=
en carlberg
Sent: Thursday, July 11, 2013 9:05 AM
To: Toerless Eckert (eckert)
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

Toerless,

some food for thought...

about 10 years ago, I put together a draft that added an optional TLV prior=
itization header to RTP to accomplish something similar in terms of having =
the app decide which was the better RTP packet to drop during times of cong=
estion.  My aim was to have these decisions made at forwarding application-=
layer gateways that did not act as a RTP/RTCP endpoint.  The strength of th=
e approach (in my opinion :-) was that it could span pockets of non-diff-se=
rv regions and still carry significance downstream.  The weakness was the n=
eed for DPI if one wanted to make a more informed decision along the path a=
bout dropping a packet.

I wonder if such a dual-layer design might be helpful in your efforts.  In =
any case, I have an interest in the normalizer draft.

-ken


On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> wr=
ote:

> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.
>=20
> One of the main concerns raised against such schemes was one of fairness.=
 If flows=20
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.
>=20
> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>=20
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
> so this is not theoretical).
>=20
> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example=20
>=20
>  a) how well can it work in routers to shift packet drops to low-prio pac=
kets
>  b) whats the evidence that it's good for video quality
>  c) How should we do the marking best going forward
>=20
> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)
>=20
> Thanks!
>    Toerless
>=20
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>=20
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>=20
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>=20
>> Best regards,
>> CJ
>>=20
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>=20
>>=20
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-lai-tsvwg-normalizer
>> Revision:	 00
>> Title:		 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:	 2013-02-08
>> WG ID:		 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>=20
>>=20
>> Abstract:
>>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>   nodes to normalize the distribution of DS codepoint (DSCP) markings
>>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>   useful for traffic conditioning with Active Queue Management (AQM)
>>   when the AF PHB group is deployed for multimedia service classes that
>>   carry video packets with advanced codec technologies.  Using NM can
>>   offer an incentive that is however missing in today's ecosystem of
>>   practical deployment, i.e., when congestion occurs in the AF PHB
>>   group, the network should allow a video codec to benefit from its
>>   effort of generating finer layers of intra-flow precedence (IFP) with
>>   discardable packets in a more balanced distribution.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


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On Thu, Jul 11, 2013 at 3:18 PM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 11 July 2013 12:46, Adam Roach <adam@nostrum.com> wrote:
>> Jointly by the chairs of RTCWEB and WebRTC, based on
>> <http://www.w3.org/2011/04/webrtc-charter.html> and
>> <http://tools.ietf.org/wg/rtcweb/charters>, one would presume.
>
> To be perfectly clear, that's precisely what makes me nervous.

[MB] Me too. [/MB]
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From eckert@cisco.com  Thu Jul 11 17:30:11 2013
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Date: Thu, 11 Jul 2013 17:30:02 -0700
From: Toerless Eckert <eckert@cisco.com>
To: ken carlberg <carlberg@g11.org.uk>
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Subject: Re: [rtcweb] [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Thanks, Ken

Indeed yes, that's exactly the line of thought that we would like to
see proceed as well. Any URL still existing for that draft ?

"application layer gateway" could specifically mean switching MCU using
intelligent dropping in conjunction with outbound congestion control.

I definitely would like to see RTP headers as the ultimatele marking
option even for routers to do intelligent dropping (or other packet
processing "beyond DSCP)). Intra-flow (per-packet) DSCP is just a big pain 
(aka: impossible) for applications. Many OSs don't have any APIs to support
this. And DSCP management in networks is already difficult enough as it
stands.

Would certainly like to see if there is interest to write up r present
more pieces of this puzzle, especially an RTP extension. We just felt that the
fairness in intelligent dropping was the biggest concern our video
application folks had, that's why we wanted to explain how that piece
could work first. 

Cheers
    Toerless


On Thu, Jul 11, 2013 at 12:04:58PM -0400, ken carlberg wrote:
> Toerless,
> 
> some food for thought...
> 
> about 10 years ago, I put together a draft that added an optional TLV prioritization header to RTP to accomplish something similar in terms of having the app decide which was the better RTP packet to drop during times of congestion.  My aim was to have these decisions made at forwarding application-layer gateways that did not act as a RTP/RTCP endpoint.  The strength of the approach (in my opinion :-) was that it could span pockets of non-diff-serv regions and still carry significance downstream.  The weakness was the need for DPI if one wanted to make a more informed decision along the path about dropping a packet.
> 
> I wonder if such a dual-layer design might be helpful in your efforts.  In any case, I have an interest in the normalizer draft.
> 
> -ken
> 
> 
> On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> wrote:
> 
> > When dealing with congestion for real-time datagram (UDP/RTP) based video flows, one idea
> > that we think has shown to provide good value is to intelligently drop in the network
> > preferentially the least important packets within a video flow, for example non-referenced
> > P-frames, because their loss creates the lowest impact to video quality.
> > 
> > One of the main concerns raised against such schemes was one of fairness. If flows 
> > would start to indicate different priorities for packets in them, how would we
> > be able to avoid that flows would indicate all their packets to be of high priority
> > to shift packet drops to other flows. Or even more importantly: how could we motivate
> > flows to indicate that some of their packets are low priority without them having to
> > fear that they would experience more loss than than flows that don't do this.
> > 
> > So, there is one draft that we would like to present (presumably in TSVWG)  explaining
> > how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.txt
> > 
> > (Actually, this is how we did solve the problem in our implementatin of intelligent dropping,
> > so this is not theoretical).
> > 
> > Now granted, the problem of fairness is but one element of building a complete ssytem
> > around the idea of intelligent dropping, so if folks here on the lists would like to
> > see for example 
> > 
> >  a) how well can it work in routers to shift packet drops to low-prio packets
> >  b) whats the evidence that it's good for video quality
> >  c) How should we do the marking best going forward
> > 
> > Then please let us know. We'd be very interested to provide insight into the parts of
> > those pieces that we have also worked out or discuss the ones we have not (primarily c ;-)
> > 
> > Thanks!
> >    Toerless
> > 
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
> > 
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >> 
> >> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> >> 
> >> Best regards,
> >> CJ
> >> 
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> >> 
> >> 
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >> 
> >> Filename:	 draft-lai-tsvwg-normalizer
> >> Revision:	 00
> >> Title:		 Normalization Marker for AF PHB Group in DiffServ
> >> Creation date:	 2013-02-08
> >> WG ID:		 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> >> 
> >> 
> >> Abstract:
> >>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>   nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>   useful for traffic conditioning with Active Queue Management (AQM)
> >>   when the AF PHB group is deployed for multimedia service classes that
> >>   carry video packets with advanced codec technologies.  Using NM can
> >>   offer an incentive that is however missing in today's ecosystem of
> >>   practical deployment, i.e., when congestion occurs in the AF PHB
> >>   group, the network should allow a video codec to benefit from its
> >>   effort of generating finer layers of intra-flow precedence (IFP) with
> >>   discardable packets in a more balanced distribution.
> >> 
> >> 
> >> 
> >> 
> >> The IETF Secretariat
> >> 
> > 
> > -- 
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> > 
> 

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87
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Subject: Re: [rtcweb] Draft agenda for IETF 87
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On Jul 11, 2013, at 5:06 PM, Mary Barnes <mary.ietf.barnes@gmail.com>
 wrote:

> On Thu, Jul 11, 2013 at 3:18 PM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
>> On 11 July 2013 12:46, Adam Roach <adam@nostrum.com> wrote:
>>> Jointly by the chairs of RTCWEB and WebRTC, based on
>>> <http://www.w3.org/2011/04/webrtc-charter.html> and
>>> <http://tools.ietf.org/wg/rtcweb/charters>, one would presume.
>>=20
>> To be perfectly clear, that's precisely what makes me nervous.
>=20
> [MB] Me too. [/MB]

One might read this to imply that this was chair decision about the scope, =
but in this case, I don't think the chairs have very much to do with if the=
 no-plan draft happens to be in scope of this WG or not.  As the authors ex=
plained it, it is purely an API change with no changes to anything on the w=
ire. Everyone in this WG seems to agree on where API works happens and I ha=
ve not heard any disagreement on that. Of course that not surprising that t=
here is not disagreement on this as the charters of WebRTC and RTCWeb seem =
relatively clear on this.

On the decision of what to put on a draft agenda before the -00 deadline, o=
f course Ted and I take responsibility for the current draft agenda.




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Date: Thu, 11 Jul 2013 17:59:30 -0700
From: "Toerless Eckert (eckert)" <eckert@cisco.com>
To: Dave Taht <dave.taht@gmail.com>
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Subject: Re: [rtcweb] [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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On Wed, Jul 10, 2013 at 12:02:26PM -0700, Dave Taht wrote:
> I've begun to look at webrtc of late in the context of fq_codel.
> 
> One thing that might work would be ECN marking important frames and
> not ECN marking others.

True. But i think there are at least 3 layers of priority that would
really be great to have: really important packet (long term reference
frames), "normal" packets, and low-proirity-packets (eg: discardable P).
And ECN could only do two.

Most of the time when you would want intelligent dropping, your drop
rate should be fairly low, eg< 20% instantaneous within the queue. If
you can do discardable P-frames it's of course best to drop those first.
Only when you instantaneously have really big bursts would you want to
drop "normal" packets and refrain to protection really important packets.

This does work really well with different queue-length (or WRED profiles).
If you only had two priorities (via ECN) it's hard to say how you
would best assign these two priorities to at least those three priority
levels.

> Problem overall is that ecn can be gamed and is largely not
> enabled, and so on...

Well. The gaming could be inhibited exactly by this draft because it
would try to guarantee the same amount of drops on a per-flow or "class"
basis, so if you're trying to game the system, you would be put into
your own class and would still see the same amount of drops as other flows.

Well, the main issue is really to redesignate ECN for a different
semantic than what it has today. And indicating packet priority is just
not quite the same as congestion feedback.

> In the case of the fq_codel algorithm I can see a "delayed drop"
> mechanism happening,
> where upon deciding to drop, it could defer a drop up to a few packets
> based on the classification
> of the packet, say 3 packets at most. Since flows are already
> identified, this is kind of easy
> by adding another state variable...

Yes, this is all quite interesting. We also played around with droppping
during enqueuing vs. during dequeuing, so there are a bunch of things
that could optimize intelligent dropping if you can build a more
intelligent queue.

[...]

> 
> So I can see trying to utilize the AF markings as you describe in your
> draft in codel's
> drop decision. While this is also subject to being gamed, the
> disincentives are slight.

Lets separate the marking from the mechanism to inhibit gaming. This
draft really is about the latter, and the marking described with AF is
really just an example. Yes, it would be lovely to agree on markings
too.

> And what codel depends on is a definition of "TCP friendly". That if a
> stream is not going to behave in a tcp friendly manner, then an entirely
> different  strategy for such flows seems necessary, and glomming on some
> additional state isn't going to be the right thing.

Who stood up at RMCAT some time back and claimed to have worked for
> 10 years on TCP and declared TCP not self friendly to itself, so RMCAT
shouldn't bother ? ;-))

Kidding aside: Intelligent dropping IMHO makes perfect sense when being
applied to RT traffic sharing a queue friendly with TCP. And intelligent
dropping can then still improve the performance.

I am pretty positive though that RMCAT traffic in general will behave
better and get lower delay and more video-appropriate loss if it can have
its own queue. Primarily because it will just take decates to root out
all the bad TCP traffic. 

I don't think that the strategies if you do or if you don't share the queue
with TCP would be fundamentally be different. 

> a) how well can it work in routers to shift packet drops to low-prio packets
> 
> I guess where I fall apart here is how to shift the encoders to send thinner
> streams at  any level of packet drop.

Video codec coders are known to be able to do crazy cool things.
There are a bunch of known mechanisms and probably more proprietary ones.

> I'd be interested in specific codecs, example video streams, and test
> scripts to try and fiddle with this, as well as viable metrics.

Sure. Best done in person. If we're going to be at the WG for this, we can
talk on the side how we've been doing this. 

Cheers
    Toerless

> > Thanks!
> >     Toerless
> >
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
> >
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >>
> >> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> >>
> >> Best regards,
> >> CJ
> >>
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> >>
> >>
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >>
> >> Filename:      draft-lai-tsvwg-normalizer
> >> Revision:      00
> >> Title:                 Normalization Marker for AF PHB Group in DiffServ
> >> Creation date:         2013-02-08
> >> WG ID:                 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> >>
> >>
> >> Abstract:
> >>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>    nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>    useful for traffic conditioning with Active Queue Management (AQM)
> >>    when the AF PHB group is deployed for multimedia service classes that
> >>    carry video packets with advanced codec technologies.  Using NM can
> >>    offer an incentive that is however missing in today's ecosystem of
> >>    practical deployment, i.e., when congestion occurs in the AF PHB
> >>    group, the network should allow a video codec to benefit from its
> >>    effort of generating finer layers of intra-flow precedence (IFP) with
> >>    discardable packets in a more balanced distribution.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
> >>
> >
> > --
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> >
> 
> 
> 
> -- 
> Dave Täht
> 
> Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.html

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


From fluffy@iii.ca  Thu Jul 11 18:09:42 2013
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From: Cullen Jennings <fluffy@iii.ca>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Question about the status of various drafts
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On Jul 11, 2013, at 1:28 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> On 2013-07-05 02:47, Cullen Jennings wrote:
>> draft-ietf-rtcweb-rtp-usage		40
>=20
> I disagree that that this is as low as 40%. There are some few thorny
> issues that might take some discussion to resolve. However, most =
things
> are from an authors perspective quite solid. I would say 60-70%.

Not sure how I got 40 on that one.  60-70 makes sense. (and rest of =
email but just wanted to reply to this part right away=

From stefan.lk.hakansson@ericsson.com  Fri Jul 12 00:35:12 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87
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On 7/11/13 9:38 PM, Martin Thomson wrote:=0A=
> On 11 July 2013 12:04, Cullen Jennings (fluffy) <fluffy@cisco.com>=0A=
> wrote:=0A=
>> On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org>=0A=
>>> In you last mail on the subject you mentioned that we will be=0A=
>>> discussing No Plan in Berlin together with plans A and B. Could=0A=
>>> we please add this to the agenda?=0A=
>>=0A=
>> No. We believe that conversation needs to happen in the W3C WebRTC=0A=
>> WG. I expect to see a message from W3C chairs on this at some=0A=
>> point.=0A=
>=0A=
> I'm a little nervous about this.  Where does the decision on the=0A=
> separation of responsibilities (API vs. SDP) get made?=0A=
=0A=
(I have not been able to confer with any of the other chairs, so this is =
=0A=
just my personal opinion):=0A=
=0A=
To me it seems pretty straightforward to a certain point:=0A=
=0A=
* The SDP (if we opt to continue using SDP for this purpose) that goes=0A=
on the signaling wire between the browsers is defined by IETF (and by=0A=
the rtcweb WG I presume even though MMUSIC seems to have some stake)=0A=
=0A=
* JS APIs to:=0A=
** Apply an SDP (e.g. received on the signaling channel) to the browser=0A=
** Hand an SDP generated by the browser over to the application (for=0A=
transmission over the signaling wire presumably)=0A=
** Influencing/modifying the contents of the SDP=0A=
* All belongs to the W3C WebRTC=0A=
=0A=
What seems unclear to me is where we define what modifications to the=0A=
SDP that are allowed - and when. Even though the ambition is to have=0A=
APIs that makes SDP mangling an exception, we will still see that=0A=
happening.=0A=
=0A=
> _______________________________________________ rtcweb mailing list=0A=
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

From andrew.hutton@siemens-enterprise.com  Fri Jul 12 02:01:25 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Regarding the FW traversal discussion then I still think we need a discussi=
on in the RTCWEB WG and I hope to persuade the chairs that this is the case=
.

We have requirements in the use case draft and charter items that need solu=
tions and this is a real issue impacting RTCWeb trials today.

Regards
Andy



From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ted Hardie
Sent: 11 July 2013 17:51
To: rtcweb@ietf.org
Cc: Cullen Jennings
Subject: [rtcweb] Draft agenda for IETF 87

Greetings,

Below is an initial draft agenda for the upcoming meeting.   Since we have =
not yet reached the draft deadline (which is the 15th), there may be new dr=
afts or updates that result in changes.  We did already receive requests fo=
r NAT/Firewall traversal discussion, and the chairs will be working with th=
e document authors to get them considered in the appropriate groups.

As folks have probably noticed, we are meeting Thursday and Friday, after t=
he MMUSIC sessions are complete (they are Tuesday and Wednesday). This shou=
ld allow us to discuss the results on our first day.

Please send feedback or change proposals to the list.

thanks,

Ted and Cullen

Day 1:

Should SDES be part of  WebRTC security practice and, if so, how?
Presentations: 30 minutes
Discussion:  40 minutes

Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
Presentation: 30 minutes
Discussion: 30 minutes

Security document updates
Presentation: 10 minutes
Discussion: 10 minutes

Day 2:

Chair Discussion:  10 minutes

Use Case Requirements updates:
Issues list presentation: 20 minutes
Discussion: 20 minutes

Data channel:
Issues list presentation:  45 minutes
Discussion: 45 minutes


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regarding the FW traversa=
l discussion then I still think we need a discussion in the RTCWEB WG and I=
 hope to persuade the chairs that this is the case.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">We have requirements in t=
he use case draft and charter items that need solutions and this is a real =
issue impacting RTCWeb trials today. &nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Andy<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> 11 July 2013 17:51<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Cc:</b> Cullen Jennings<br>
<b>Subject:</b> [rtcweb] Draft agenda for IETF 87<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Greetings,<br>
<br>
Below is an initial draft agenda for the upcoming meeting.&nbsp;&nbsp; Sinc=
e we have not yet reached the draft deadline (which is the 15th), there may=
 be new drafts or updates that result in changes.&nbsp; We did already rece=
ive requests for NAT/Firewall traversal discussion,
 and the chairs will be working with the document authors to get them consi=
dered in the appropriate groups.<br>
<br>
As folks have probably noticed, we are meeting Thursday and Friday, after t=
he MMUSIC sessions are complete (they are Tuesday and Wednesday). This shou=
ld allow us to discuss the results on our first day.
<br>
<br>
Please send feedback or change proposals to the list.<br>
<br>
thanks,<br>
<br>
Ted and Cullen<br>
<br>
Day 1:<br>
<br>
Should SDES be part of&nbsp; WebRTC security practice and, if so, how?<br>
Presentations: 30 minutes<br>
Discussion:&nbsp; 40 minutes<br>
<br>
Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents<br>
Presentation: 30 minutes<br>
Discussion: 30 minutes<br>
<br>
Security document updates<br>
Presentation: 10 minutes<br>
Discussion: 10 minutes<br>
<br>
Day 2:<br>
<br>
Chair Discussion:&nbsp; 10 minutes<br>
<br>
Use Case Requirements updates:<br>
Issues list presentation: 20 minutes<br>
Discussion: 20 minutes<br>
<br>
Data channel:<br>
Issues list presentation:&nbsp; 45 minutes<br>
Discussion: 45 minutes<br>
<br>
<o:p></o:p></p>
</div>
</div>
</body>
</html>

--_000_9F33F40F6F2CD847824537F3C4E37DDF116406C8MCHP04MSXglobal_--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 12 Jul 2013 11:11:02 +0200
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Subject: Re: [rtcweb] Draft agenda for IETF 87
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2013/7/12 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> * JS APIs to:
> ** Apply an SDP (e.g. received on the signaling channel) to the browser
> ** Hand an SDP generated by the browser over to the application (for
> transmission over the signaling wire presumably)
> ** Influencing/modifying the contents of the SDP
> * All belongs to the W3C WebRTC

I've never seen an API in which the application requests some data to
the core, and then mangles such a data before sending it to the peer.
This is like if I ask the browser to send video, the browser gives me
a media description (I will not assume SDP) with ptime=3D20, and I
mangle the media description to set ptime=3D50, while at the same time I
have no idea whether my browser can send RTP with ptime=3D50 or not.
It's just awesome.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 7/12/13 11:11 AM, I=F1aki Baz Castillo wrote:=0A=
> 2013/7/12 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com>:=0A=
>> * JS APIs to:=0A=
>> ** Apply an SDP (e.g. received on the signaling channel) to the browser=
=0A=
>> ** Hand an SDP generated by the browser over to the application (for=0A=
>> transmission over the signaling wire presumably)=0A=
>> ** Influencing/modifying the contents of the SDP=0A=
>> * All belongs to the W3C WebRTC=0A=
>=0A=
> I've never seen an API in which the application requests some data to=0A=
> the core, and then mangles such a data before sending it to the peer.=0A=
> This is like if I ask the browser to send video, the browser gives me=0A=
> a media description (I will not assume SDP) with ptime=3D20, and I=0A=
> mangle the media description to set ptime=3D50, while at the same time I=
=0A=
> have no idea whether my browser can send RTP with ptime=3D50 or not.=0A=
=0A=
I think this is not at all the way it should be done; if people think =0A=
there is a need to define a specific ptime there should be an API that =0A=
allows for that (and that has means of letting the app know what the =0A=
allowed ranges are, or at least what is actually used). Let's go and do =0A=
that in the WebRTC WG.=0A=
=0A=
> It's just awesome.=0A=
>=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

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2013/7/12 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
>> I've never seen an API in which the application requests some data to
>> the core, and then mangles such a data before sending it to the peer.
>> This is like if I ask the browser to send video, the browser gives me
>> a media description (I will not assume SDP) with ptime=3D20, and I
>> mangle the media description to set ptime=3D50, while at the same time I
>> have no idea whether my browser can send RTP with ptime=3D50 or not.
>
> I think this is not at all the way it should be done; if people think
> there is a need to define a specific ptime there should be an API that
> allows for that (and that has means of letting the app know what the
> allowed ranges are, or at least what is actually used).

That looks a real JS Object based API, right? ;)


> Let's go and do
> that in the WebRTC WG.

Sure.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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--e89a8ff2488125ead604e15172a5
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A minor correction:  I haven't engaged anyone on public-webrtc.  I didn't
even realize there were discussions going on over there about the API,
since I wasn't keeping track and I thought they were all going on over
here.  Sorry about that.  I guess I'll head over there and start engaging
:).


On Tue, Jul 9, 2013 at 1:40 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
>
> On Tue, Jul 9, 2013 at 9:11 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>
>> On 09/07/2013 11:33 AM, Ted Hardie wrote:
>>
>>> Howdy,
>>>
>>> The recent set of API discussions has been spread across both the rtcweb
>>> and public-webrtc mailing lists.  That's making it both harder to follow
>>> and harder for folks to work out who is saying what under which rules.  The
>>> chairs of both groups believe that the right place for the discussion to
>>> continue should be public-webrtc.  Please direct follow-ups on this topic
>>> to that list.
>>>
>>> regards,
>>>
>>> Ted Hardie
>>>
>> Ted,
>>
>>     I agree, with one caveat: virtually none of the high-level
>> stakeholders (spec editors, browser vendors, etc) bother to engage the
>> community on public-webrtc.
>>
>
> I'm not sure I understand this complaint. Is it that the aforementioned
> "high-level stakeholders"
> aren't engaging or merely that they are only engaging on RTCWEB? If it's
> the former, than
> I don't think that's actually true, since in the past week, you've had
> responses from (at least)
> the following people who fall into those categories:
>
> Cullen Jennings (spec editor)
> Adam Bergqvist (spec editor)
> Peter Thatcher (works on Chrome)
> Me (works on Firefox and Chrome; spec editor)
> Christer Holmberg (spec editor)
> Several people from Microsoft.
>
> Who, exactly, are you expecting to engage that hasn't engaged?
>
>
> If your complaint is just that they're engaging on the wrong mailing list,
> well
> that seems to reinforce Ted's point above.
>
> -Ekr
>
>
>
>
>     What's the point of discussing the API on this mailing list if our
>> opinion goes unnoticed? We shouldn't be moving the discussion to
>> public-webrtc as a nice way to filter us out. This discussion requires
>> their attention, be it on one mailing list or the other. I don't mind where
>> we discuss it, so long as they get involved.
>>
>>     Is it their intention to get involved on public-webrtc and summarize
>> the results on rtcweb?
>>
>> Thanks,
>> Gili
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--e89a8ff2488125ead604e15172a5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">A minor correction: =C2=A0I haven&#39;t engaged anyone on =
public-webrtc. =C2=A0I didn&#39;t even realize there were discussions going=
 on over there about the API, since I wasn&#39;t keeping track and I though=
t they were all going on over here. =C2=A0Sorry about that. =C2=A0I guess I=
&#39;ll head over there and start engaging :).<br>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Jul 9=
, 2013 at 1:40 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ek=
r@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote"><div class=3D"im">On Tue, Jul 9, 2013 at 9:11 AM, cowwoc <span dir=
=3D"ltr">&lt;<a href=3D"mailto:cowwoc@bbs.darktech.org" target=3D"_blank">c=
owwoc@bbs.darktech.org</a>&gt;</span> wrote:<br>




<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div>On 09/07/2013 11:33 AM, Ted Hardie=
 wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Howdy,<br>
<br>
The recent set of API discussions has been spread across both the rtcweb an=
d public-webrtc mailing lists. =C2=A0That&#39;s making it both harder to fo=
llow and harder for folks to work out who is saying what under which rules.=
 =C2=A0The chairs of both groups believe that the right place for the discu=
ssion to continue should be public-webrtc. =C2=A0Please direct follow-ups o=
n this topic to that list.<br>





<br>
regards,<br>
<br>
Ted Hardie<br>
</blockquote></div></div>
Ted,<br>
<br>
=C2=A0 =C2=A0 I agree, with one caveat: virtually none of the high-level st=
akeholders (spec editors, browser vendors, etc) bother to engage the commun=
ity on public-webrtc.<br></blockquote><div><br></div></div><div>I&#39;m not=
 sure I understand this complaint. Is it that the aforementioned &quot;high=
-level stakeholders&quot;</div>




<div>aren&#39;t engaging or merely that they are only engaging on RTCWEB? I=
f it&#39;s the former, than</div><div>I don&#39;t think that&#39;s actually=
 true, since in the past week, you&#39;ve had responses from (at least)</di=
v>



<div>the following people who fall into those categories:</div><div><br></d=
iv><div>Cullen Jennings (spec editor)</div><div>Adam Bergqvist (spec editor=
)</div><div>Peter Thatcher (works on Chrome)</div><div>Me (works on Firefox=
 and Chrome; spec editor)</div>



<div>Christer Holmberg (spec editor)</div><div>Several people from Microsof=
t.</div><div><br></div><div>Who, exactly, are you expecting to engage that =
hasn&#39;t engaged?=C2=A0</div><div><br></div>

<div><br></div><div>If your complaint is just that they&#39;re engaging on =
the wrong mailing list, well</div><div>that seems to reinforce Ted&#39;s po=
int above.</div><div><br></div><div>-Ekr</div><div class=3D"im">

<div><br></div><div><br></div><div><br></div><div><br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
=C2=A0 =C2=A0 What&#39;s the point of discussing the API on this mailing li=
st if our opinion goes unnoticed? We shouldn&#39;t be moving the discussion=
 to public-webrtc as a nice way to filter us out. This discussion requires =
their attention, be it on one mailing list or the other. I don&#39;t mind w=
here we discuss it, so long as they get involved.<br>





<br>
=C2=A0 =C2=A0 Is it their intention to get involved on public-webrtc and su=
mmarize the results on rtcweb?<br>
<br>
Thanks,<br>
Gili<br>
<br>
</blockquote></div></div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--e89a8ff2488125ead604e15172a5--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Can you get specific about exactly what you want to discuss? The current so=
lution ins the specs uses ICE, STUN, TURN and works thorough many firewalls=
 but not all. What change would you like to see?

On Jul 12, 2013, at 2:01 AM, "Hutton, Andrew" <andrew.hutton@siemens-enterp=
rise.com>
 wrote:

> Regarding the FW traversal discussion then I still think we need a discus=
sion in the RTCWEB WG and I hope to persuade the chairs that this is the ca=
se.
> =20
> We have requirements in the use case draft and charter items that need so=
lutions and this is a real issue impacting RTCWeb trials today. =20
> =20
> Regards
> Andy
> =20
> =20
> =20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Ted Hardie
> Sent: 11 July 2013 17:51
> To: rtcweb@ietf.org
> Cc: Cullen Jennings
> Subject: [rtcweb] Draft agenda for IETF 87
> =20
> Greetings,
>=20
> Below is an initial draft agenda for the upcoming meeting.   Since we hav=
e not yet reached the draft deadline (which is the 15th), there may be new =
drafts or updates that result in changes.  We did already receive requests =
for NAT/Firewall traversal discussion, and the chairs will be working with =
the document authors to get them considered in the appropriate groups.
>=20
> As folks have probably noticed, we are meeting Thursday and Friday, after=
 the MMUSIC sessions are complete (they are Tuesday and Wednesday). This sh=
ould allow us to discuss the results on our first day.=20
>=20
> Please send feedback or change proposals to the list.
>=20
> thanks,
>=20
> Ted and Cullen
>=20
> Day 1:
>=20
> Should SDES be part of  WebRTC security practice and, if so, how?
> Presentations: 30 minutes
> Discussion:  40 minutes
>=20
> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> Presentation: 30 minutes
> Discussion: 30 minutes
>=20
> Security document updates
> Presentation: 10 minutes
> Discussion: 10 minutes
>=20
> Day 2:
>=20
> Chair Discussion:  10 minutes
>=20
> Use Case Requirements updates:
> Issues list presentation: 20 minutes
> Discussion: 20 minutes
>=20
> Data channel:
> Issues list presentation:  45 minutes
> Discussion: 45 minutes
>=20
>=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Andy, sorry =85 ignore this. I see you provided what I was asking for on an=
other thread.=20

On Jul 12, 2013, at 8:09 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>=20
> Can you get specific about exactly what you want to discuss? The current =
solution ins the specs uses ICE, STUN, TURN and works thorough many firewal=
ls but not all. What change would you like to see?
>=20
> On Jul 12, 2013, at 2:01 AM, "Hutton, Andrew" <andrew.hutton@siemens-ente=
rprise.com>
> wrote:
>=20
>> Regarding the FW traversal discussion then I still think we need a discu=
ssion in the RTCWEB WG and I hope to persuade the chairs that this is the c=
ase.
>>=20
>> We have requirements in the use case draft and charter items that need s=
olutions and this is a real issue impacting RTCWeb trials today. =20
>>=20
>> Regards
>> Andy
>>=20
>>=20
>>=20
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf=
 Of Ted Hardie
>> Sent: 11 July 2013 17:51
>> To: rtcweb@ietf.org
>> Cc: Cullen Jennings
>> Subject: [rtcweb] Draft agenda for IETF 87
>>=20
>> Greetings,
>>=20
>> Below is an initial draft agenda for the upcoming meeting.   Since we ha=
ve not yet reached the draft deadline (which is the 15th), there may be new=
 drafts or updates that result in changes.  We did already receive requests=
 for NAT/Firewall traversal discussion, and the chairs will be working with=
 the document authors to get them considered in the appropriate groups.
>>=20
>> As folks have probably noticed, we are meeting Thursday and Friday, afte=
r the MMUSIC sessions are complete (they are Tuesday and Wednesday). This s=
hould allow us to discuss the results on our first day.=20
>>=20
>> Please send feedback or change proposals to the list.
>>=20
>> thanks,
>>=20
>> Ted and Cullen
>>=20
>> Day 1:
>>=20
>> Should SDES be part of  WebRTC security practice and, if so, how?
>> Presentations: 30 minutes
>> Discussion:  40 minutes
>>=20
>> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
>> Presentation: 30 minutes
>> Discussion: 30 minutes
>>=20
>> Security document updates
>> Presentation: 10 minutes
>> Discussion: 10 minutes
>>=20
>> Day 2:
>>=20
>> Chair Discussion:  10 minutes
>>=20
>> Use Case Requirements updates:
>> Issues list presentation: 20 minutes
>> Discussion: 20 minutes
>>=20
>> Data channel:
>> Issues list presentation:  45 minutes
>> Discussion: 45 minutes
>>=20
>>=20
>=20


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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Date: Fri, 12 Jul 2013 15:17:49 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
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What we want to discuss is a solution to the HTTP (Proxy) only FW use case =
as described in http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-=
requirements-11#section-3.2.3. This is not solved by the current specs.=20

The draft http://tools.ietf.org/html/draft-hutton-rtcweb-nat-firewall-consi=
derations-01 proposes a solution based on using HTTP Connect and some other=
 related browser best practice requirements but also discusses alternatives=
.

We want to discuss solving this use case and hopefully get this draft adopt=
ed.

Andy



> -----Original Message-----
> From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> Sent: 12 July 2013 16:09
> To: Hutton, Andrew
> Cc: Ted Hardie; rtcweb@ietf.org
> Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
>=20
>=20
> Can you get specific about exactly what you want to discuss? The
> current solution ins the specs uses ICE, STUN, TURN and works thorough
> many firewalls but not all. What change would you like to see?
>=20
> On Jul 12, 2013, at 2:01 AM, "Hutton, Andrew" <andrew.hutton@siemens-
> enterprise.com>
>  wrote:
>=20
> > Regarding the FW traversal discussion then I still think we need a
> discussion in the RTCWEB WG and I hope to persuade the chairs that this
> is the case.
> >
> > We have requirements in the use case draft and charter items that
> need solutions and this is a real issue impacting RTCWeb trials today.
> >
> > Regards
> > Andy
> >
> >
> >
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ted Hardie
> > Sent: 11 July 2013 17:51
> > To: rtcweb@ietf.org
> > Cc: Cullen Jennings
> > Subject: [rtcweb] Draft agenda for IETF 87
> >
> > Greetings,
> >
> > Below is an initial draft agenda for the upcoming meeting.   Since we
> have not yet reached the draft deadline (which is the 15th), there may
> be new drafts or updates that result in changes.  We did already
> receive requests for NAT/Firewall traversal discussion, and the chairs
> will be working with the document authors to get them considered in the
> appropriate groups.
> >
> > As folks have probably noticed, we are meeting Thursday and Friday,
> after the MMUSIC sessions are complete (they are Tuesday and
> Wednesday). This should allow us to discuss the results on our first
> day.
> >
> > Please send feedback or change proposals to the list.
> >
> > thanks,
> >
> > Ted and Cullen
> >
> > Day 1:
> >
> > Should SDES be part of  WebRTC security practice and, if so, how?
> > Presentations: 30 minutes
> > Discussion:  40 minutes
> >
> > Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> > Presentation: 30 minutes
> > Discussion: 30 minutes
> >
> > Security document updates
> > Presentation: 10 minutes
> > Discussion: 10 minutes
> >
> > Day 2:
> >
> > Chair Discussion:  10 minutes
> >
> > Use Case Requirements updates:
> > Issues list presentation: 20 minutes
> > Discussion: 20 minutes
> >
> > Data channel:
> > Issues list presentation:  45 minutes
> > Discussion: 45 minutes
> >
> >


From fluffy@cisco.com  Fri Jul 12 08:30:57 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
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So section 3.3.1 of draft-hutton-rtcweb-nat-firewall-considerations provide=
s a brief sketch of an extension to TURN clients to use a HTTP proxy.=20

I would be shocked to see anyone in this WG tell you not to do that, it's b=
een discussed many times in the past. But in the end this WG will either sa=
y we like it, take it to BEHAVE, or it will say we hate it in which case as=
 and individual draft you will still probably take it to BEHAVE. You probab=
ly want to keep an eye on HTTP2 stuff and implications to this too.=20

We had not much list discussion on this draft. We have an agenda very full =
of WG items that are very controversial. We could put it at the end of the =
agenda with "time permitting" but to be honest, I sort of doubt we would ge=
t to it. Is there some question you think you want feedback from this group=
 on that we could start on this list or something?=20

I'm just trying to be realistic about the outcomes of this draft - we could=
 discuss it in RTCWeb for 3 meetings before proposing a  TURN extension in =
behave but in the end I suspect we could just short circuit all that.=20


On Jul 12, 2013, at 8:17 AM, "Hutton, Andrew" <andrew.hutton@siemens-enterp=
rise.com> wrote:

> What we want to discuss is a solution to the HTTP (Proxy) only FW use cas=
e as described in http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-an=
d-requirements-11#section-3.2.3. This is not solved by the current specs.=20
>=20
> The draft http://tools.ietf.org/html/draft-hutton-rtcweb-nat-firewall-con=
siderations-01 proposes a solution based on using HTTP Connect and some oth=
er related browser best practice requirements but also discusses alternativ=
es.
>=20
> We want to discuss solving this use case and hopefully get this draft ado=
pted.
>=20
> Andy
>=20
>=20
>=20
>> -----Original Message-----
>> From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
>> Sent: 12 July 2013 16:09
>> To: Hutton, Andrew
>> Cc: Ted Hardie; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
>>=20
>>=20
>> Can you get specific about exactly what you want to discuss? The
>> current solution ins the specs uses ICE, STUN, TURN and works thorough
>> many firewalls but not all. What change would you like to see?
>>=20
>> On Jul 12, 2013, at 2:01 AM, "Hutton, Andrew" <andrew.hutton@siemens-
>> enterprise.com>
>> wrote:
>>=20
>>> Regarding the FW traversal discussion then I still think we need a
>> discussion in the RTCWEB WG and I hope to persuade the chairs that this
>> is the case.
>>>=20
>>> We have requirements in the use case draft and charter items that
>> need solutions and this is a real issue impacting RTCWeb trials today.
>>>=20
>>> Regards
>>> Andy
>>>=20
>>>=20
>>>=20
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Ted Hardie
>>> Sent: 11 July 2013 17:51
>>> To: rtcweb@ietf.org
>>> Cc: Cullen Jennings
>>> Subject: [rtcweb] Draft agenda for IETF 87
>>>=20
>>> Greetings,
>>>=20
>>> Below is an initial draft agenda for the upcoming meeting.   Since we
>> have not yet reached the draft deadline (which is the 15th), there may
>> be new drafts or updates that result in changes.  We did already
>> receive requests for NAT/Firewall traversal discussion, and the chairs
>> will be working with the document authors to get them considered in the
>> appropriate groups.
>>>=20
>>> As folks have probably noticed, we are meeting Thursday and Friday,
>> after the MMUSIC sessions are complete (they are Tuesday and
>> Wednesday). This should allow us to discuss the results on our first
>> day.
>>>=20
>>> Please send feedback or change proposals to the list.
>>>=20
>>> thanks,
>>>=20
>>> Ted and Cullen
>>>=20
>>> Day 1:
>>>=20
>>> Should SDES be part of  WebRTC security practice and, if so, how?
>>> Presentations: 30 minutes
>>> Discussion:  40 minutes
>>>=20
>>> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
>>> Presentation: 30 minutes
>>> Discussion: 30 minutes
>>>=20
>>> Security document updates
>>> Presentation: 10 minutes
>>> Discussion: 10 minutes
>>>=20
>>> Day 2:
>>>=20
>>> Chair Discussion:  10 minutes
>>>=20
>>> Use Case Requirements updates:
>>> Issues list presentation: 20 minutes
>>> Discussion: 20 minutes
>>>=20
>>> Data channel:
>>> Issues list presentation:  45 minutes
>>> Discussion: 45 minutes
>>>=20
>>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Fri Jul 12 09:04:59 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] division of responsibilities (was: Draft agenda for IETF 87)
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On 12 July 2013 00:35, Stefan H=C3=A5kansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> To me it seems pretty straightforward to a certain point:
>
> * The SDP (if we opt to continue using SDP for this purpose) that goes
> on the signaling wire between the browsers is defined by IETF (and by
> the rtcweb WG I presume even though MMUSIC seems to have some stake)
>
> * JS APIs to:
> ** Apply an SDP (e.g. received on the signaling channel) to the browser
> ** Hand an SDP generated by the browser over to the application (for
> transmission over the signaling wire presumably)
> ** Influencing/modifying the contents of the SDP
> * All belongs to the W3C WebRTC
>
> What seems unclear to me is where we define what modifications to the
> SDP that are allowed - and when. Even though the ambition is to have
> APIs that makes SDP mangling an exception, we will still see that
> happening.

I can see how you would reach that conclusion, but it's only simple at
the most superficial level.

There needs to be a clear understanding about what information is
carried in SDP and what information is supplied by the application.  I
have observed that there is an assumption implicit in a lot of the
work that says that SDP carries everything it can possibly carry.
I'll note that this has been implicit, never explicit.  And the mere
existence of no-plan highlights the fact that there is a fundamental
disagreement on this point.

We've been building a consensus that is based on an assumption.  I'd
rather we had consensus than an untested assumption.  With all respect
to the chairs, that's not something I trust them to rule on at this
point.

From andrew.hutton@siemens-enterprise.com  Fri Jul 12 09:44:25 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87 - FW Issues
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
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I totally don't follow this argument.

What we need is for RTCWEB browsers to behave in a standardized way in envi=
ronments where there is a FW requiring the flow to come from a HTTP Proxy. =
Defining this is clearly within the scope of RTCWEB and if we move the work=
 to BEHAVE we still need a RTCWEB specification to say what rtcweb implemen=
tations are required to implement.

The charter states the wg will.

"Define the solution - protocols and API requirements - for firewall and NA=
T traversal".

It also states:

"This work will be done primarily by using already defined protocols or fun=
ctionalities. If there is identification of missing protocols or functional=
ities, such work can be requested to be done in another working group with =
a suitable charter or by requests for chartering it in this WG or another W=
G".

We have not so far identified any missing protocol or functionality so why =
do you ask the work to be moved to a different group?

With regard to the full agenda I for one think we can find time for this ra=
ther than again circling through the endless discussion on some of the othe=
r topics. I am not sure also why we need 40 mins to discuss the use case up=
dates time would be better spent finding solutions for the existing use cas=
es.

If you think that nobody will object then surely we can get this over very =
quickly.

Agree there has not been enough discussion on the draft but I have received=
 only positive feedback including some comments off list.

Regards
Andy





> -----Original Message-----
> From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> Sent: 12 July 2013 16:31
> To: Hutton, Andrew
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
>=20
>=20
> So section 3.3.1 of draft-hutton-rtcweb-nat-firewall-considerations
> provides a brief sketch of an extension to TURN clients to use a HTTP
> proxy.
>=20
> I would be shocked to see anyone in this WG tell you not to do that,
> it's been discussed many times in the past. But in the end this WG will
> either say we like it, take it to BEHAVE, or it will say we hate it in
> which case as and individual draft you will still probably take it to
> BEHAVE. You probably want to keep an eye on HTTP2 stuff and
> implications to this too.
>=20
> We had not much list discussion on this draft. We have an agenda very
> full of WG items that are very controversial. We could put it at the
> end of the agenda with "time permitting" but to be honest, I sort of
> doubt we would get to it. Is there some question you think you want
> feedback from this group on that we could start on this list or
> something?
>=20
> I'm just trying to be realistic about the outcomes of this draft - we
> could discuss it in RTCWeb for 3 meetings before proposing a  TURN
> extension in behave but in the end I suspect we could just short
> circuit all that.
>=20
>=20
> On Jul 12, 2013, at 8:17 AM, "Hutton, Andrew" <andrew.hutton@siemens-
> enterprise.com> wrote:
>=20
> > What we want to discuss is a solution to the HTTP (Proxy) only FW use
> case as described in http://tools.ietf.org/html/draft-ietf-rtcweb-use-
> cases-and-requirements-11#section-3.2.3. This is not solved by the
> current specs.
> >
> > The draft http://tools.ietf.org/html/draft-hutton-rtcweb-nat-
> firewall-considerations-01 proposes a solution based on using HTTP
> Connect and some other related browser best practice requirements but
> also discusses alternatives.
> >
> > We want to discuss solving this use case and hopefully get this draft
> adopted.
> >
> > Andy
> >
> >
> >
> >> -----Original Message-----
> >> From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> >> Sent: 12 July 2013 16:09
> >> To: Hutton, Andrew
> >> Cc: Ted Hardie; rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Draft agenda for IETF 87 - FW Issues
> >>
> >>
> >> Can you get specific about exactly what you want to discuss? The
> >> current solution ins the specs uses ICE, STUN, TURN and works
> thorough
> >> many firewalls but not all. What change would you like to see?
> >>
> >> On Jul 12, 2013, at 2:01 AM, "Hutton, Andrew"
> <andrew.hutton@siemens-
> >> enterprise.com>
> >> wrote:
> >>
> >>> Regarding the FW traversal discussion then I still think we need a
> >> discussion in the RTCWEB WG and I hope to persuade the chairs that
> this
> >> is the case.
> >>>
> >>> We have requirements in the use case draft and charter items that
> >> need solutions and this is a real issue impacting RTCWeb trials
> today.
> >>>
> >>> Regards
> >>> Andy
> >>>
> >>>
> >>>
> >>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >> Behalf Of Ted Hardie
> >>> Sent: 11 July 2013 17:51
> >>> To: rtcweb@ietf.org
> >>> Cc: Cullen Jennings
> >>> Subject: [rtcweb] Draft agenda for IETF 87
> >>>
> >>> Greetings,
> >>>
> >>> Below is an initial draft agenda for the upcoming meeting.   Since
> we
> >> have not yet reached the draft deadline (which is the 15th), there
> may
> >> be new drafts or updates that result in changes.  We did already
> >> receive requests for NAT/Firewall traversal discussion, and the
> chairs
> >> will be working with the document authors to get them considered in
> the
> >> appropriate groups.
> >>>
> >>> As folks have probably noticed, we are meeting Thursday and Friday,
> >> after the MMUSIC sessions are complete (they are Tuesday and
> >> Wednesday). This should allow us to discuss the results on our first
> >> day.
> >>>
> >>> Please send feedback or change proposals to the list.
> >>>
> >>> thanks,
> >>>
> >>> Ted and Cullen
> >>>
> >>> Day 1:
> >>>
> >>> Should SDES be part of  WebRTC security practice and, if so, how?
> >>> Presentations: 30 minutes
> >>> Discussion:  40 minutes
> >>>
> >>> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> >>> Presentation: 30 minutes
> >>> Discussion: 30 minutes
> >>>
> >>> Security document updates
> >>> Presentation: 10 minutes
> >>> Discussion: 10 minutes
> >>>
> >>> Day 2:
> >>>
> >>> Chair Discussion:  10 minutes
> >>>
> >>> Use Case Requirements updates:
> >>> Issues list presentation: 20 minutes
> >>> Discussion: 20 minutes
> >>>
> >>> Data channel:
> >>> Issues list presentation:  45 minutes
> >>> Discussion: 45 minutes
> >>>
> >>>
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Fri, 12 Jul 2013 10:19:06 -0700
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Subject: Re: [rtcweb] Draft agenda for IETF 87
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On Fri, Jul 12, 2013 at 12:35 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 7/11/13 9:38 PM, Martin Thomson wrote:
> > On 11 July 2013 12:04, Cullen Jennings (fluffy) <fluffy@cisco.com>
> > wrote:
> >> On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org>
> >>> In you last mail on the subject you mentioned that we will be
> >>> discussing No Plan in Berlin together with plans A and B. Could
> >>> we please add this to the agenda?
> >>
> >> No. We believe that conversation needs to happen in the W3C WebRTC
> >> WG. I expect to see a message from W3C chairs on this at some
> >> point.
> >
> > I'm a little nervous about this.  Where does the decision on the
> > separation of responsibilities (API vs. SDP) get made?
>
> (I have not been able to confer with any of the other chairs, so this is
> just my personal opinion):
>
> To me it seems pretty straightforward to a certain point:
>
> * The SDP (if we opt to continue using SDP for this purpose) that goes
> on the signaling wire between the browsers is defined by IETF (and by
> the rtcweb WG I presume even though MMUSIC seems to have some stake)
>
> * JS APIs to:
> ** Apply an SDP (e.g. received on the signaling channel) to the browser
> ** Hand an SDP generated by the browser over to the application (for
> transmission over the signaling wire presumably)
> ** Influencing/modifying the contents of the SDP
> * All belongs to the W3C WebRTC
>

Would it make sense to change this list to say the following?

* JS APIs to:
** Apply a SessionDescription (created by SDP; eg received on the signaling
channel or built by JS) to the browser
** Hand a SessionDescription (serializable to SDP) generated by the browser
over to the application (for
transmission over the signaling wire presumably or for immediately applying
with setLocalDescription for some local effect)
** Influencing/modifying the contents of the SessionDescription (and thus,
SDP, when serialized to SDP)
* All belongs to the W3C WebRTC


My changes highlight a few things:
1.  The API (setLocalDescripiton, setRemoteDescription, creatOffer, etc)
doesn't work with SDP directly.  It works with RTCSessionDescription
objects.  It just happens to be that currently the only way to interact
with RTCSessionDescription objects is through SDP.  I think it's worth
remembering that distinction.

2.  SDP does not necessarily go over the wire.   It only necessarily goes
through the API between JS and browser.   I think it's worth remembering
that distinction.

3.  Calling setLocalDescription with a new SessionDescription object can
have purely local effects that don't need to be sent over the wire.  I
think it's worth remembering those cases.



>
> What seems unclear to me is where we define what modifications to the
> SDP that are allowed - and when. Even though the ambition is to have
> APIs that makes SDP mangling an exception, we will still see that
> happening.
>

I think advanced JS apps are going to use every control knob they can get
to, whether it's anticipated and well-supported or not.    I think there's
a good chance that a a popular WebRTC web app will use some SDP mangling
that wasn't anticipated, but happened to work, but then the browser can't
remove it because it will break certain websites.  It could get even worse
if someone writes a popular WebRTC wrapper library that uses tricky SDP
mangling that is then used by lots of websites.  Certain SDP mangling
techniques might end up becoming a defacto standard API that can't be
removed, even if it was originally a bug.  Or worse, one browser will have
to implement the SDP mangling or even the bugs of another, because WebRTC
apps have come to rely on them.

In fact, it's already the case that Chrome and Firefox support far
different SDP manglings.  I don't think any web apps rely on that yet, but
it's only a matter of time before someone figures out "hey, if I mangle the
SDP on this browser this way and on that browser that way, I can do things
I couldn't do otherwise".  Or worse, an advanced web app developer says
"hey, I can make this work well on browser X via SDP mangling, but not on
browser Y, so I'll put a 'best used with Browser X' icon on my website".
Then that someone writes an abstraction on top of that, and then maybe
shares that with others, and it goes from there.

I think we're in a race with web developers to see if they'll figure out
SDP mangling before we provide a way to avoid SDP mangling.   Who do you
think moves faster?










>
> > _______________________________________________ rtcweb mailing list
> > rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--e89a8fb20834a8c89004e153b7e7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 12, 2013 at 12:35 AM, Stefan H=C3=A5kansson LK <span di=
r=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D=
"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/11/13 9:38 PM, Martin Thomson wrote=
:<br>


&gt; On 11 July 2013 12:04, Cullen Jennings (fluffy) &lt;<a href=3D"mailto:=
fluffy@cisco.com">fluffy@cisco.com</a>&gt;<br>
&gt; wrote:<br>
&gt;&gt; On Jul 11, 2013, at 10:35 AM, Emil Ivov &lt;<a href=3D"mailto:emch=
o@jitsi.org">emcho@jitsi.org</a>&gt;<br>
&gt;&gt;&gt; In you last mail on the subject you mentioned that we will be<=
br>
&gt;&gt;&gt; discussing No Plan in Berlin together with plans A and B. Coul=
d<br>
&gt;&gt;&gt; we please add this to the agenda?<br>
&gt;&gt;<br>
&gt;&gt; No. We believe that conversation needs to happen in the W3C WebRTC=
<br>
&gt;&gt; WG. I expect to see a message from W3C chairs on this at some<br>
&gt;&gt; point.<br>
&gt;<br>
&gt; I&#39;m a little nervous about this. =C2=A0Where does the decision on =
the<br>
&gt; separation of responsibilities (API vs. SDP) get made?<br>
<br>
</div>(I have not been able to confer with any of the other chairs, so this=
 is<br>
just my personal opinion):<br>
<br>
To me it seems pretty straightforward to a certain point:<br>
<br>
* The SDP (if we opt to continue using SDP for this purpose) that goes<br>
on the signaling wire between the browsers is defined by IETF (and by<br>
the rtcweb WG I presume even though MMUSIC seems to have some stake)<br>
<br>
* JS APIs to:<br>
** Apply an SDP (e.g. received on the signaling channel) to the browser<br>
** Hand an SDP generated by the browser over to the application (for<br>
transmission over the signaling wire presumably)<br>
** Influencing/modifying the contents of the SDP<br>
* All belongs to the W3C WebRTC<br></blockquote><div><br></div><div>Would i=
t make sense to change this list to say the following?</div><div><br></div>=
<div>* JS APIs to:<br>** Apply a SessionDescription (created by SDP; eg rec=
eived on the signaling channel or built by JS) to the browser</div>

<div>** Hand a SessionDescription (serializable to SDP) generated by the br=
owser over to the application (for<br>transmission over the signaling wire =
presumably or for immediately applying with setLocalDescription for some lo=
cal effect)<br>

** Influencing/modifying the contents of the SessionDescription (and thus, =
SDP, when serialized to SDP)<br>* All belongs to the W3C WebRTC<br></div><d=
iv><br></div><div><br></div><div>My changes highlight a few things:</div>

<div>1. =C2=A0The API (setLocalDescripiton, setRemoteDescription, creatOffe=
r, etc) doesn&#39;t work with SDP directly. =C2=A0It works with RTCSessionD=
escription objects. =C2=A0It just happens to be that currently the only way=
 to interact with RTCSessionDescription objects is through SDP. =C2=A0I thi=
nk it&#39;s worth remembering that distinction. =C2=A0</div>

<div><br></div><div>2. =C2=A0SDP does not necessarily go over the wire. =C2=
=A0 It only necessarily goes through the API between JS and browser. =C2=A0=
 I think it&#39;s worth remembering that distinction. =C2=A0</div><div><br>=
</div><div>3. =C2=A0Calling setLocalDescription with a new SessionDescripti=
on object can have purely local effects that don&#39;t need to be sent over=
 the wire. =C2=A0I think it&#39;s worth remembering those cases. =C2=A0</di=
v>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex">
<br>
What seems unclear to me is where we define what modifications to the<br>
SDP that are allowed - and when. Even though the ambition is to have<br>
APIs that makes SDP mangling an exception, we will still see that<br>
happening.<br></blockquote><div><br></div><div>I think advanced JS apps are=
 going to use every control knob they can get to, whether it&#39;s anticipa=
ted and well-supported or not. =C2=A0 =C2=A0I think there&#39;s a good chan=
ce that a a popular WebRTC web app will use some SDP mangling that wasn&#39=
;t anticipated, but happened to work, but then the browser can&#39;t remove=
 it because it will break certain websites. =C2=A0It could get even worse i=
f someone writes a popular WebRTC wrapper library that uses tricky SDP mang=
ling that is then used by lots of websites. =C2=A0Certain SDP mangling tech=
niques might end up becoming a defacto standard API that can&#39;t be remov=
ed, even if it was originally a bug. =C2=A0Or worse, one browser will have =
to implement the SDP mangling or even the bugs of another, because WebRTC a=
pps have come to rely on them.</div>

<div><br></div><div>In fact, it&#39;s already the case that Chrome and Fire=
fox support far different SDP manglings. =C2=A0I don&#39;t think any web ap=
ps rely on that yet, but it&#39;s only a matter of time before someone figu=
res out &quot;hey, if I mangle the SDP on this browser this way and on that=
 browser that way, I can do things I couldn&#39;t do otherwise&quot;. =C2=
=A0Or worse, an advanced web app developer says &quot;hey, I can make this =
work well on browser X via SDP mangling, but not on browser Y, so I&#39;ll =
put a &#39;best used with Browser X&#39; icon on my website&quot;. Then tha=
t someone writes an abstraction on top of that, and then maybe shares that =
with others, and it goes from there. =C2=A0</div>

<div><br></div><div>I think we&#39;re in a race with web developers to see =
if they&#39;ll figure out SDP mangling before we provide a way to avoid SDP=
 mangling. =C2=A0 Who do you think moves faster?</div><div><br></div><div><=
br>

</div><div><br></div><div><br></div><div><br></div><div>=C2=A0</div><div><b=
r></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rg=
b(204,204,204);border-left-style:solid;padding-left:1ex">


<div class=3D""><div class=3D"h5"><br>
&gt; _______________________________________________ rtcweb mailing list<br=
>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a href=3D"http=
s://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.iet=
f.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--e89a8fb20834a8c89004e153b7e7--

From stefan.lk.hakansson@ericsson.com  Fri Jul 12 10:27:54 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: division of responsibilities (was: Draft agenda for IETF 87)
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Subject: Re: [rtcweb] division of responsibilities (was: Draft agenda for IETF 87)
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On 7/12/13 6:04 PM, Martin Thomson wrote:=0A=
> On 12 July 2013 00:35, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com> wrote:=0A=
>> To me it seems pretty straightforward to a certain point:=0A=
>>=0A=
>> * The SDP (if we opt to continue using SDP for this purpose) that goes=
=0A=
>> on the signaling wire between the browsers is defined by IETF (and by=0A=
>> the rtcweb WG I presume even though MMUSIC seems to have some stake)=0A=
>>=0A=
>> * JS APIs to:=0A=
>> ** Apply an SDP (e.g. received on the signaling channel) to the browser=
=0A=
>> ** Hand an SDP generated by the browser over to the application (for=0A=
>> transmission over the signaling wire presumably)=0A=
>> ** Influencing/modifying the contents of the SDP=0A=
>> * All belongs to the W3C WebRTC=0A=
>>=0A=
>> What seems unclear to me is where we define what modifications to the=0A=
>> SDP that are allowed - and when. Even though the ambition is to have=0A=
>> APIs that makes SDP mangling an exception, we will still see that=0A=
>> happening.=0A=
>=0A=
> I can see how you would reach that conclusion, but it's only simple at=0A=
> the most superficial level.=0A=
>=0A=
> There needs to be a clear understanding about what information is=0A=
> carried in SDP and what information is supplied by the application.  I=0A=
> have observed that there is an assumption implicit in a lot of the=0A=
> work that says that SDP carries everything it can possibly carry.=0A=
> I'll note that this has been implicit, never explicit.  And the mere=0A=
> existence of no-plan highlights the fact that there is a fundamental=0A=
> disagreement on this point.=0A=
=0A=
This is a very good point IMO. You have brought it up before, and I =0A=
think it is very valid. We need to decide what info is carried over=0A=
=0A=
* in SDP (if SDP is kept)=0A=
* in RTP/RTCP=0A=
* in any other means we standardize=0A=
=0A=
and what is supplied by the application.=0A=
=0A=
>=0A=
> We've been building a consensus that is based on an assumption.  I'd=0A=
> rather we had consensus than an untested assumption.  With all respect=0A=
> to the chairs, that's not something I trust them to rule on at this=0A=
> point.=0A=
=0A=
FWIW, I agree. This needs more discussion.=0A=
=0A=
>=0A=
=0A=

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 12 Jul 2013 10:31:57 -0700
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Tue, Jul 9, 2013 at 4:26 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 2013-07-09 12:18, I=C3=B1aki Baz Castillo wrote:
> > 2013/7/9 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> >> I want to be very clear and careful on what I say. So I am repeating:
> >>
> >> * My comment that I think Eric is right in that there is consensus on
> >> providing APIs that allow most use-cases to be met without SDP manglin=
g
> >> is meant only in that context: SDP O/A is kept (and PeerConnection mor=
e
> >> or less as is and so on).
> >
> > Hi Stefan,
> >
> > You insist that "there is consensus on keeping SDP O/A but providing a
> > better API".
>
> I must be expressing myself quite badly, because the message seems not
> to get through.
>
> * I think we have consensus on adding API that allows most use-cases to
> be met without SDP mangling (given that SDP O/A is kept naturally)
>
> * We discussed last year an alternative API that did not use SDP at all
> (and did not use PeerConnection). The decision at that time was to
> continue developing the API based on PeerConnection and SDP O/A.
>
> Have not stated anything more than that.
>

Thank you for clarifying, Stefan.  It's often very difficult for those of
us who weren't there to understand what was consensus, and what wasn't, and
what was decided and what was decided and what wasn't.  It seems that if
you ask 10 people, you get 10 different answers.


In other words, it's hard to see much consensus on what we have consensus
on.


So thank you for your clarifications.  It's probably not a very fun job for
you :).   Along with that, I have two more questions for you:


1.  You said we have consensus on adding API that allows most use-cases to
be met without SDP mangling.  Since that point in time has there been in
any progress in adding such APIs?  Can you give me a rough feel for how
much time has passed, how many proposals have been made, how many have been
accepted, and how many have been implemented?  I think it's great that
we'll do this "someday", but I'd like to get a feel for how far away
"someday" is.  Thanks in advance.

2.  You said we have consensus against an alternative API.  Was that
consensus against any additions to the API that would allow JS to bypass
SDP, or was that a consensus against just that particular alternative API?
 I'd like to get a feel for how many people voted for "no; let's not go
down that specific road" vs. "no; we cannot go on any road but this one".
I think there's a big difference between the two.  Thanks again in advance.


By the way, Ted asked us to move discussions about the API to
public-webrtc, so should we do that now?



> Stefan
>
> > Given current discussions IMHO it is clear there is not
> > such a consensus (not at all). Or may be you are just talking about
> > two years ago in some IETF meeting (if so I'm sorry).
> >
> > Please review the results in
> >
> https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdH=
lZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1
> >
> >    Bad for advanced stuff: 94%
> >    OK for 1.0: 40%
> >
> > IMHO such a result indicates all but "let's keep SDP O/A and improve a
> > bit the API" (I mean now, in July 2013).
> >
> >
> > Best regards.
> >
> >
> >
> > --
> > I=C3=B1aki Baz Castillo
> > <ibc@aliax.net>
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Jul 9, 2013 at 4:26 AM, Stefan H=C3=A5kansson LK <span dir=
=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"=
_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 2013-07-09 12:18, I=C3=B1aki Baz Cast=
illo wrote:<br>


&gt; 2013/7/9 Stefan H=C3=A5kansson LK &lt;<a href=3D"mailto:stefan.lk.haka=
nsson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;:<br>
&gt;&gt; I want to be very clear and careful on what I say. So I am repeati=
ng:<br>
&gt;&gt;<br>
&gt;&gt; * My comment that I think Eric is right in that there is consensus=
 on<br>
&gt;&gt; providing APIs that allow most use-cases to be met without SDP man=
gling<br>
&gt;&gt; is meant only in that context: SDP O/A is kept (and PeerConnection=
 more<br>
&gt;&gt; or less as is and so on).<br>
&gt;<br>
&gt; Hi Stefan,<br>
&gt;<br>
&gt; You insist that &quot;there is consensus on keeping SDP O/A but provid=
ing a<br>
&gt; better API&quot;.<br>
<br>
</div>I must be expressing myself quite badly, because the message seems no=
t<br>
to get through.<br>
<br>
* I think we have consensus on adding API that allows most use-cases to<br>
be met without SDP mangling (given that SDP O/A is kept naturally)<br>
<br>
* We discussed last year an alternative API that did not use SDP at all<br>
(and did not use PeerConnection). The decision at that time was to<br>
continue developing the API based on PeerConnection and SDP O/A.<br>
<br>
Have not stated anything more than that.<br></blockquote><div><br></div><di=
v>Thank you for clarifying, Stefan. =C2=A0It&#39;s often very difficult for=
 those of us who weren&#39;t there to understand what was consensus, and wh=
at wasn&#39;t, and what was decided and what was decided and what wasn&#39;=
t. =C2=A0It seems that if you ask 10 people, you get 10 different answers. =
=C2=A0</div>

<div><br></div><div><br></div><div>In other words, it&#39;s hard to see muc=
h consensus on what we have consensus on.</div><div><br></div><div><br></di=
v><div>So thank you for your clarifications. =C2=A0It&#39;s probably not a =
very fun job for you :). =C2=A0 Along with that, I have two more questions =
for you:</div>

<div><br></div><div><br></div><div>1. =C2=A0You said we have consensus on a=
dding API that allows most use-cases to be met without SDP mangling. =C2=A0=
Since that point in time has there been in any progress in adding such APIs=
? =C2=A0Can you give me a rough feel for how much time has passed, how many=
 proposals have been made, how many have been accepted, and how many have b=
een implemented? =C2=A0I think it&#39;s great that we&#39;ll do this &quot;=
someday&quot;, but I&#39;d like to get a feel for how far away &quot;someda=
y&quot; is. =C2=A0Thanks in advance.</div>

<div><br></div><div>2. =C2=A0You said we have consensus against an alternat=
ive API. =C2=A0Was that consensus against any additions to the API that wou=
ld allow JS to bypass SDP, or was that a consensus against just that partic=
ular alternative API? =C2=A0I&#39;d like to get a feel for how many people =
voted for &quot;no; let&#39;s not go down that specific road&quot; vs. &quo=
t;no; we cannot go on any road but this one&quot;. =C2=A0 I think there&#39=
;s a big difference between the two. =C2=A0Thanks again in advance.</div>

<div><br></div><div><br></div><div>By the way, Ted asked us to move discuss=
ions about the API to public-webrtc, so should we do that now?</div><div><b=
r></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bo=
rder-left-style:solid;padding-left:1ex">


<span class=3D""><font color=3D"#888888"><br>
Stefan<br>
</font></span><div class=3D""><div class=3D"h5"><br>
&gt; Given current discussions IMHO it is clear there is not<br>
&gt; such a consensus (not at all). Or may be you are just talking about<br=
>
&gt; two years ago in some IETF meeting (if so I&#39;m sorry).<br>
&gt;<br>
&gt; Please review the results in<br>
&gt; <a href=3D"https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0=
AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1" target=3D"_blank">http=
s://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9R=
N0xSWFhybVl4anJLRkVPV0E#gid=3D1</a><br>


&gt;<br>
&gt; =C2=A0 =C2=A0Bad for advanced stuff: 94%<br>
&gt; =C2=A0 =C2=A0OK for 1.0: 40%<br>
&gt;<br>
&gt; IMHO such a result indicates all but &quot;let&#39;s keep SDP O/A and =
improve a<br>
&gt; bit the API&quot; (I mean now, in July 2013).<br>
&gt;<br>
&gt;<br>
&gt; Best regards.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; I=C3=B1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--047d7b5d438c9a998404e153e5d7--

From martin.thomson@gmail.com  Fri Jul 12 10:41:45 2013
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Subject: [rtcweb] Another reason not to use SDP (was: Draft agenda for IETF 87)
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On 12 July 2013 10:19, Peter Thatcher <pthatcher@google.com> wrote:
> I think advanced JS apps are going to use every control knob they can get
> to, whether it's anticipated and well-supported or not.    I think there's a
> good chance that a a popular WebRTC web app will use some SDP mangling that
> wasn't anticipated, but happened to work, but then the browser can't remove
> it because it will break certain websites.  It could get even worse if
> someone writes a popular WebRTC wrapper library that uses tricky SDP
> mangling that is then used by lots of websites.  Certain SDP mangling
> techniques might end up becoming a defacto standard API that can't be
> removed, even if it was originally a bug.  Or worse, one browser will have
> to implement the SDP mangling or even the bugs of another, because WebRTC
> apps have come to rely on them.
>
> In fact, it's already the case that Chrome and Firefox support far different
> SDP manglings.  I don't think any web apps rely on that yet, but it's only a
> matter of time before someone figures out "hey, if I mangle the SDP on this
> browser this way and on that browser that way, I can do things I couldn't do
> otherwise".  Or worse, an advanced web app developer says "hey, I can make
> this work well on browser X via SDP mangling, but not on browser Y, so I'll
> put a 'best used with Browser X' icon on my website". Then that someone
> writes an abstraction on top of that, and then maybe shares that with
> others, and it goes from there.
>
> I think we're in a race with web developers to see if they'll figure out SDP
> mangling before we provide a way to avoid SDP mangling.   Who do you think
> moves faster?

This is one of the arguments we've made against the use of SDP.  There
is a remedy for this (aside from comment 22 ;), but it's a fair amount
of work:

 - provide very explicit and detailed instructions on what SDP to
produce under all starting conditions, and

 - provide very explicit and detailed instructions on what to do with
every single bit of SDP that is provided.

That is what I believe to be necessary if this API is to have any hope
of real success.

Unfortunately, when I sat down to do this, I barely made it to t=
sections.  I simply cannot justify spending the time required for that
sort of chore.

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On Jul 12, 2013, at 9:04 AM, Martin Thomson <martin.thomson@gmail.com>
 wrote:

> With all respect
> to the chairs, that's not something I trust them to rule on at this
> point.

just to be clear, I don't think the RTCWeb chairs have done anything remote=
ly close to a ruling on this. We have taken censuses calls on the use of 32=
64 O/A.



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On 12 July 2013 10:42, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> On Jul 12, 2013, at 9:04 AM, Martin Thomson <martin.thomson@gmail.com>
>  wrote:
>> With all respect
>> to the chairs, that's not something I trust them to rule on at this
>> point.
>
> just to be clear, I don't think the RTCWeb chairs have done anything remotely close to a ruling on this. We have taken censuses calls on the use of 3264 O/A.

I didn't mean to imply that a ruling has been made.  To a large
extent, we haven't even discussed the issue that no-plan represents.
That's a large part of where my concern derives from.

It's a tricky situation: chairs have to make rulings on the
appropriate venue for discussion, but when that ruling seems to be
predicated on having a technical decision, it creates a bind.

From pthatcher@google.com  Fri Jul 12 11:22:05 2013
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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 12 Jul 2013 11:21:21 -0700
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Another reason not to use SDP (was: Draft agenda for IETF 87)
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--bcaec51f99e54f49c504e15496f0
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On Fri, Jul 12, 2013 at 10:41 AM, Martin Thomson
<martin.thomson@gmail.com>wrote:

> On 12 July 2013 10:19, Peter Thatcher <pthatcher@google.com> wrote:
> > I think advanced JS apps are going to use every control knob they can get
> > to, whether it's anticipated and well-supported or not.    I think
> there's a
> > good chance that a a popular WebRTC web app will use some SDP mangling
> that
> > wasn't anticipated, but happened to work, but then the browser can't
> remove
> > it because it will break certain websites.  It could get even worse if
> > someone writes a popular WebRTC wrapper library that uses tricky SDP
> > mangling that is then used by lots of websites.  Certain SDP mangling
> > techniques might end up becoming a defacto standard API that can't be
> > removed, even if it was originally a bug.  Or worse, one browser will
> have
> > to implement the SDP mangling or even the bugs of another, because WebRTC
> > apps have come to rely on them.
> >
> > In fact, it's already the case that Chrome and Firefox support far
> different
> > SDP manglings.  I don't think any web apps rely on that yet, but it's
> only a
> > matter of time before someone figures out "hey, if I mangle the SDP on
> this
> > browser this way and on that browser that way, I can do things I
> couldn't do
> > otherwise".  Or worse, an advanced web app developer says "hey, I can
> make
> > this work well on browser X via SDP mangling, but not on browser Y, so
> I'll
> > put a 'best used with Browser X' icon on my website". Then that someone
> > writes an abstraction on top of that, and then maybe shares that with
> > others, and it goes from there.
> >
> > I think we're in a race with web developers to see if they'll figure out
> SDP
> > mangling before we provide a way to avoid SDP mangling.   Who do you
> think
> > moves faster?
>
> This is one of the arguments we've made against the use of SDP.  There
> is a remedy for this (aside from comment 22 ;), but it's a fair amount
> of work:
>
>  - provide very explicit and detailed instructions on what SDP to
> produce under all starting conditions, and
>
>  - provide very explicit and detailed instructions on what to do with
> every single bit of SDP that is provided.
>

If we write it down with such detailed instructions, is a browser banned
from accepting more than the explicitly allowed SDP in calls to
setLocalDescription and setRemoteDescription?  For example, what if lots of
web apps get SDP from a particularly popular gateway device that adds a
particular SDP attribute, and if the browser "listens" to that SDP
attribute, things work better (for example, it might be a bandwidth-related
attribute).  Can the browser "listen" to that SDP attribute even though it
isn't explicitly supported by the WG?  Or does it have to wait for it to be
added to the explicit instructions?



> That is what I believe to be necessary if this API is to have any hope
> of real success.
>
>
It depends on how you define "real success".   I think the API has already
had success for simple use cases, seeing as two browsers have working,
interoperable implementations for the simple use cases.  It's the advanced
ones I see being problematic.





> Unfortunately, when I sat down to do this, I barely made it to t=
> sections.  I simply cannot justify spending the time required for that
> sort of chore.
>

Well, then you're almost finished.  All you really have left are the m=
sections :).

--bcaec51f99e54f49c504e15496f0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 12, 2013 at 10:41 AM, Martin Thomson <span dir=3D"ltr">=
&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.th=
omson@gmail.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 12 July 2013 10:19, Peter Thatcher &lt;<a=
 href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.co=
m</a>&gt; wrote:<br>



&gt; I think advanced JS apps are going to use every control knob they can =
get<br>
&gt; to, whether it&#39;s anticipated and well-supported or not. =C2=A0 =C2=
=A0I think there&#39;s a<br>
&gt; good chance that a a popular WebRTC web app will use some SDP mangling=
 that<br>
&gt; wasn&#39;t anticipated, but happened to work, but then the browser can=
&#39;t remove<br>
&gt; it because it will break certain websites. =C2=A0It could get even wor=
se if<br>
&gt; someone writes a popular WebRTC wrapper library that uses tricky SDP<b=
r>
&gt; mangling that is then used by lots of websites. =C2=A0Certain SDP mang=
ling<br>
&gt; techniques might end up becoming a defacto standard API that can&#39;t=
 be<br>
&gt; removed, even if it was originally a bug. =C2=A0Or worse, one browser =
will have<br>
&gt; to implement the SDP mangling or even the bugs of another, because Web=
RTC<br>
&gt; apps have come to rely on them.<br>
&gt;<br>
&gt; In fact, it&#39;s already the case that Chrome and Firefox support far=
 different<br>
&gt; SDP manglings. =C2=A0I don&#39;t think any web apps rely on that yet, =
but it&#39;s only a<br>
&gt; matter of time before someone figures out &quot;hey, if I mangle the S=
DP on this<br>
&gt; browser this way and on that browser that way, I can do things I could=
n&#39;t do<br>
&gt; otherwise&quot;. =C2=A0Or worse, an advanced web app developer says &q=
uot;hey, I can make<br>
&gt; this work well on browser X via SDP mangling, but not on browser Y, so=
 I&#39;ll<br>
&gt; put a &#39;best used with Browser X&#39; icon on my website&quot;. The=
n that someone<br>
&gt; writes an abstraction on top of that, and then maybe shares that with<=
br>
&gt; others, and it goes from there.<br>
&gt;<br>
&gt; I think we&#39;re in a race with web developers to see if they&#39;ll =
figure out SDP<br>
&gt; mangling before we provide a way to avoid SDP mangling. =C2=A0 Who do =
you think<br>
&gt; moves faster?<br>
<br>
This is one of the arguments we&#39;ve made against the use of SDP. =C2=A0T=
here<br>
is a remedy for this (aside from comment 22 ;), but it&#39;s a fair amount<=
br>
of work:<br>
<br>
=C2=A0- provide very explicit and detailed instructions on what SDP to<br>
produce under all starting conditions, and<br>
<br>
=C2=A0- provide very explicit and detailed instructions on what to do with<=
br>
every single bit of SDP that is provided.<br></blockquote><div><br></div><d=
iv>If we write it down with such detailed instructions, is a browser banned=
 from accepting more than the explicitly allowed SDP in calls to setLocalDe=
scription and setRemoteDescription? =C2=A0For example, what if lots of web =
apps get SDP from a particularly popular gateway device that adds a particu=
lar SDP attribute, and if the browser &quot;listens&quot; to that SDP attri=
bute, things work better (for example, it might be a bandwidth-related attr=
ibute). =C2=A0Can the browser &quot;listen&quot; to that SDP attribute even=
 though it isn&#39;t explicitly supported by the WG? =C2=A0Or does it have =
to wait for it to be added to the explicit instructions?</div>

<div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
That is what I believe to be necessary if this API is to have any hope<br>
of real success.<br>
<br></blockquote><div><br></div><div>It depends on how you define &quot;rea=
l success&quot;. =C2=A0 I think the API has already had success for simple =
use cases, seeing as two browsers have working, interoperable implementatio=
ns for the simple use cases. =C2=A0It&#39;s the advanced ones I see being p=
roblematic.</div>

<div><br></div><div><br></div><div><br></div>
<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
Unfortunately, when I sat down to do this, I barely made it to t=3D<br>
sections. =C2=A0I simply cannot justify spending the time required for that=
<br>
sort of chore.<br></blockquote><div><br></div><div>Well, then you&#39;re al=
most finished. =C2=A0All you really have left are the m=3D sections :).</di=
v></div><br></div></div>

--bcaec51f99e54f49c504e15496f0--

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Just to be clear: use of SDP O/A is precisely the intended topic of the
slot we are requesting.

--sent from my mobile
On Jul 12, 2013 7:42 PM, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
wrote:

>
> On Jul 12, 2013, at 9:04 AM, Martin Thomson <martin.thomson@gmail.com>
>  wrote:
>
> > With all respect
> > to the chairs, that's not something I trust them to rule on at this
> > point.
>
> just to be clear, I don't think the RTCWeb chairs have done anything
> remotely close to a ruling on this. We have taken censuses calls on the use
> of 3264 O/A.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p dir=3D"ltr">Just to be clear: use of SDP O/A is precisely the intended t=
opic of the slot we are requesting.</p>
<p dir=3D"ltr">--sent from my mobile</p>
<div class=3D"gmail_quote">On Jul 12, 2013 7:42 PM, &quot;Cullen Jennings (=
fluffy)&quot; &lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&=
gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
On Jul 12, 2013, at 9:04 AM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com">martin.thomson@gmail.com</a>&gt;<br>
=A0wrote:<br>
<br>
&gt; With all respect<br>
&gt; to the chairs, that&#39;s not something I trust them to rule on at thi=
s<br>
&gt; point.<br>
<br>
just to be clear, I don&#39;t think the RTCWeb chairs have done anything re=
motely close to a ruling on this. We have taken censuses calls on the use o=
f <a href=3D"tel:3264" value=3D"+333264">3264</a> O/A.<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div>

--047d7bd6c2c4b7f75e04e154acf0--

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Subject: Re: [rtcweb] Another reason not to use SDP (was: Draft agenda for IETF 87)
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On 12 July 2013 11:21, Peter Thatcher <pthatcher@google.com> wrote:
> If we write it down with such detailed instructions, is a browser banned
> from accepting more than the explicitly allowed SDP in calls to
> setLocalDescription and setRemoteDescription?  For example, what if lots of
> web apps get SDP from a particularly popular gateway device that adds a
> particular SDP attribute, and if the browser "listens" to that SDP
> attribute, things work better (for example, it might be a bandwidth-related
> attribute).  Can the browser "listen" to that SDP attribute even though it
> isn't explicitly supported by the WG?  Or does it have to wait for it to be
> added to the explicit instructions?

That depends entirely on the nature of the instructions.  Allowing for
extensibility is always really, really hard, but maybe there is enough
experience with SDP now that this sort of scenario could be supported.

From ibc@aliax.net  Fri Jul 12 11:30:41 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 12 Jul 2013 20:30:15 +0200
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To: Emil Ivov <emcho@jitsi.org>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] division of responsibilities (was: Draft agenda for IETF 87)
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2013/7/12 Emil Ivov <emcho@jitsi.org>:
> Just to be clear: use of SDP O/A is precisely the intended topic of the s=
lot
> we are requesting.

Do you mean "use or non-use of SDP O/A"?


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 12 Jul 2013 12:00:32 -0700
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Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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--047d7b111f5f64ab3104e15522fe
Content-Type: text/plain; charset=UTF-8

I wasn't aware of that.  Thanks for letting me know.


On Mon, Jul 8, 2013 at 9:02 AM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

>  Some of us work for browser vendors who cannot follow that link, you
> know.****
>
> ** **
>
> Matthew Kaufman****
>
> ** **
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Peter Thatcher
> *Sent:* Monday, July 8, 2013 8:01 AM
> *To:* Silvia Pfeiffer
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] New Draft - WebRTC JS Object API Model****
>
> ** **
>
> ** **
>
> ** **
>
> On Sat, Jul 6, 2013 at 3:59 AM, Silvia Pfeiffer <silviapfeiffer1@gmail.com>
> wrote:****
>
> An interesting read. It makes me wonder if there is a full description
> available of the SDP that browsers (Chrome, Firefox) currently
> support.****
>
>  ** **
>
> Here's the description for Chrome:****
>
> ** **
>
>
> https://code.google.com/p/libjingle/source/browse/trunk/talk/app/webrtc/webrtcsdp.cc
> ****
>
> ** **
>
> :)****
>
>  ****
>
>
> Looking forward to your proposal (I assume it will be on the W3C list?).
>
> Cheers,
> Silvia.****
>
>
> On Wed, Jun 26, 2013 at 2:37 PM, Robin Raymond <robin@hookflash.com>
> wrote:
> >
> > According to many, real adoption of WebRTC will not happen if we
> continue to
> > force everyone to use this SDP Offer/Answer methodology.  It is clearly
> > blocking our way forward and the amount of specification documentation
> > remaining needed for the browser vendors to produce a compatible SDP
> based
> > WebRTC engine in a browser is much more daunting than most are willing to
> > admit.
> >
> >
> > The draft rationale behind a JavaScript Object API model:
> >
> >
> http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00.txt
> >
> > Whereas:
> >
> > The WebRTC JavaScript Object API & Shim document will be along shortly.
> >
> >
> > We wanted to get this initial rationale draft informational document out
> > right away. Everyone that has any interest should review the reasons and
> > concepts and comment before we get too far along on the details of an
> actual
> > API, the initial API will follow in the coming days.
> >
> >
> > Best regards,
> >
> > Robin Raymond
> >
> >****
>
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb****
>
>  ** **
>

--047d7b111f5f64ab3104e15522fe
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">I wasn&#39;t aware of that. =C2=A0Thanks for letting me kn=
ow.<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon=
, Jul 8, 2013 at 9:02 AM, Matthew Kaufman (SKYPE) <span dir=3D"ltr">&lt;<a =
href=3D"mailto:matthew.kaufman@skype.net" target=3D"_blank">matthew.kaufman=
@skype.net</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><a name=3D"13fbf05938c678bf__MailEndCompose"><span s=
tyle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&q=
uot;;color:#1f497d">Some of us work for browser vendors who cannot follow t=
hat link, you know.<u></u><u></u></span></a></p>


<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Matthew Kaufman<u></u><u>=
</u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #e1e1e1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"> <a hre=
f=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.=
org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank=
">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Peter Thatcher<br>
<b>Sent:</b> Monday, July 8, 2013 8:01 AM<br>
<b>To:</b> Silvia Pfeiffer<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] New Draft - WebRTC JS Object API Model<u></u><=
u></u></span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<p class=3D"MsoNormal">On Sat, Jul 6, 2013 at 3:59 AM, Silvia Pfeiffer &lt;=
<a href=3D"mailto:silviapfeiffer1@gmail.com" target=3D"_blank">silviapfeiff=
er1@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal">An interesting read. It makes me wonder if there is =
a full description<br>
available of the SDP that browsers (Chrome, Firefox) currently<br>
support.<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Here&#39;s the description for Chrome:<u></u><u></u>=
</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><a href=3D"https://code.google.com/p/libjingle/sourc=
e/browse/trunk/talk/app/webrtc/webrtcsdp.cc" target=3D"_blank">https://code=
.google.com/p/libjingle/source/browse/trunk/talk/app/webrtc/webrtcsdp.cc</a=
><u></u><u></u></p>


</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">:)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal"><br>
Looking forward to your proposal (I assume it will be on the W3C list?).<br=
>
<br>
Cheers,<br>
Silvia.<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
On Wed, Jun 26, 2013 at 2:37 PM, Robin Raymond &lt;<a href=3D"mailto:robin@=
hookflash.com" target=3D"_blank">robin@hookflash.com</a>&gt; wrote:<br>
&gt;<br>
&gt; According to many, real adoption of WebRTC will not happen if we conti=
nue to<br>
&gt; force everyone to use this SDP Offer/Answer methodology. =C2=A0It is c=
learly<br>
&gt; blocking our way forward and the amount of specification documentation=
<br>
&gt; remaining needed for the browser vendors to produce a compatible SDP b=
ased<br>
&gt; WebRTC engine in a browser is much more daunting than most are willing=
 to<br>
&gt; admit.<br>
&gt;<br>
&gt;<br>
&gt; The draft rationale behind a JavaScript Object API model:<br>
&gt;<br>
&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-we=
brtc-js-obj-api-rationale-00.txt" target=3D"_blank">
http://www.ietf.org/internet-drafts/draft-raymond-rtcweb-webrtc-js-obj-api-=
rationale-00.txt</a><br>
&gt;<br>
&gt; Whereas:<br>
&gt;<br>
&gt; The WebRTC JavaScript Object API &amp; Shim document will be along sho=
rtly.<br>
&gt;<br>
&gt;<br>
&gt; We wanted to get this initial rationale draft informational document o=
ut<br>
&gt; right away. Everyone that has any interest should review the reasons a=
nd<br>
&gt; concepts and comment before we get too far along on the details of an =
actual<br>
&gt; API, the initial API will follow in the coming days.<br>
&gt;<br>
&gt;<br>
&gt; Best regards,<br>
&gt;<br>
&gt; Robin Raymond<br>
&gt;<br>
&gt;<u></u><u></u></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&gt; _______________________________________________=
<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br></div></div>

--047d7b111f5f64ab3104e15522fe--

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Subject: Re: [rtcweb] Another reason not to use SDP (was: Draft agenda for IETF 87)
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On Fri, Jul 12, 2013 at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> That depends entirely on the nature of the instructions.  Allowing for
> extensibility is always really, really hard, but maybe there is enough
> experience with SDP now that this sort of scenario could be supported.
>
>
Unless my past 10 years of experience building SIP systems mislead me, the
current way of dealing with SDP extensibility is to define all features
supported by network or device, go through the interop with every new
network or device you need to connect to, and then, when in doubt, put an
SBC in front of it. We currently need to go through interop testing every
time we setup a new connection with a new VoIP service provider, and this
is for simple PSTN to VoIP G.711+RFC2833 calls. Adding an unusual codec,
like getting a direct interconnect in AMR-WB to cell phone networks, or
SILK to Skype, requires years of testing and still not typically available
commercially. SIP interworking for anything more complex (like even simple
video end points) is virtually non existent. So, I guess, we are building
on a very solid foundation of interprable solutions which require extensive
testing after every sneeze.
_____________
Roman Shpount

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<div class=3D"gmail_quote">On Fri, Jul 12, 2013 at 2:28 PM, Martin Thomson =
<span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D=
"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
<div class=3D"im">That depends entirely on the nature of the instructions. =
=A0Allowing for</div>
extensibility is always really, really hard, but maybe there is enough<br>
experience with SDP now that this sort of scenario could be supported.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></blockquote><div><=
br></div><div>Unless my past 10 years of experience building SIP systems mi=
slead me, the current way of dealing with SDP extensibility is to define al=
l features supported by network or device, go through the interop with ever=
y new network or device you need to connect to, and then, when in doubt, pu=
t an SBC in front of it. We currently need to go through interop testing ev=
ery time we setup a new connection with a new VoIP service provider, and th=
is is for simple PSTN to VoIP G.711+RFC2833 calls. Adding an unusual codec,=
 like getting a direct interconnect in AMR-WB to cell phone networks, or SI=
LK to Skype, requires years of testing and still not typically available co=
mmercially. SIP interworking for anything more complex (like even simple vi=
deo end points) is virtually non existent. So, I guess, we are building on =
a very solid foundation of interprable solutions which require extensive te=
sting after every sneeze.</div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--047d7b874000b4895104e15649c5--

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Subject: Re: [rtcweb] Another reason not to use SDP (was: Draft agenda for IETF 87)
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--001a11c3c96eed040104e157aa1e
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On Fri, Jul 12, 2013 at 3:23 PM, Roman Shpount <roman@telurix.com> wrote:

> On Fri, Jul 12, 2013 at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>wrote:
>
>> That depends entirely on the nature of the instructions.  Allowing for
>> extensibility is always really, really hard, but maybe there is enough
>> experience with SDP now that this sort of scenario could be supported.
>>
>>
> Unless my past 10 years of experience building SIP systems mislead me, the
> current way of dealing with SDP extensibility is to define all features
> supported by network or device, go through the interop with every new
> network or device you need to connect to, and then, when in doubt, put an
> SBC in front of it. We currently need to go through interop testing every
> time we setup a new connection with a new VoIP service provider, and this
> is for simple PSTN to VoIP G.711+RFC2833 calls. Adding an unusual codec,
> like getting a direct interconnect in AMR-WB to cell phone networks, or
> SILK to Skype, requires years of testing and still not typically available
> commercially. SIP interworking for anything more complex (like even simple
> video end points) is virtually non existent. So, I guess, we are building
> on a very solid foundation of interprable solutions which require extensive
> testing after every sneeze.
>

I think much of it goes back to supporting a specific subset of SDP for a
core set of use cases.  In the Asterisk world, we generally try to support
nearly anything and everything that operates within our set of core use
cases.  But once you trespass too far outside of them, you can only fall
back to support the parts of the session description that are interpretable
by your implementation.

Given all of the different devices that we have come to interoperate with
over the years (just with SDP, not even necessarily with SIP or other
higher layer signalling), I too fear that a lack of specific restraints on
what is permissible within SDP as utilized within WebRTC will cause
interoperability and fragmentation problems, at least in more advanced use
cases.

Matthew Fredrickson

--001a11c3c96eed040104e157aa1e
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<div dir=3D"ltr">On Fri, Jul 12, 2013 at 3:23 PM, Roman Shpount <span dir=
=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@t=
elurix.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=
=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"gmail_quote"><div class=3D"im"=
>On Fri, Jul 12, 2013 at 2:28 PM, Martin Thomson <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gm=
ail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div>That depends entirely on the nature of the instructions. =A0Allowing f=
or</div>
extensibility is always really, really hard, but maybe there is enough<br>
experience with SDP now that this sort of scenario could be supported.<br>
<div><div><br></div></div></blockquote><div><br></div></div><div>Unless my =
past 10 years of experience building SIP systems mislead me, the current wa=
y of dealing with SDP extensibility is to define all features supported by =
network or device, go through the interop with every new network or device =
you need to connect to, and then, when in doubt, put an SBC in front of it.=
 We currently need to go through interop testing every time we setup a new =
connection with a new VoIP service provider, and this is for simple PSTN to=
 VoIP G.711+RFC2833 calls. Adding an unusual codec, like getting a direct i=
nterconnect in AMR-WB to cell phone networks, or SILK to Skype, requires ye=
ars of testing and still not typically available commercially. SIP interwor=
king for anything more complex (like even simple video end points) is virtu=
ally non existent. So, I guess, we are building on a very solid foundation =
of interprable solutions which require extensive testing after every sneeze=
.</div>
</div></blockquote><div><br></div><div>I think much of it goes back to supp=
orting a specific subset of SDP for a core set of use cases. =A0In the Aste=
risk world, we generally try to support nearly anything and everything that=
 operates within our set of core use cases. =A0But once you trespass too fa=
r outside of them, you can only fall back to support the parts of the sessi=
on description that are interpretable by your implementation.</div>
<div><br></div><div>Given all of the different devices that we have come to=
 interoperate with over the years (just with SDP, not even necessarily with=
 SIP or other higher layer signalling), I too fear that a lack of specific =
restraints on what is permissible within SDP as utilized within WebRTC will=
 cause interoperability and fragmentation problems, at least in more advanc=
ed use cases.</div>
<div><br></div><div>Matthew Fredrickson</div></div></div></div>

--001a11c3c96eed040104e157aa1e--

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On Jul 11, 2013, at 2:08 PM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> I can see some implied outcomes for some of the items.  Would it help
> to list some (aspirational) goals.
>=20
> On 11 July 2013 09:51, Ted Hardie <ted.ietf@gmail.com> wrote:
>> Should SDES be part of  WebRTC security practice and, if so, how?
>> Presentations: 30 minutes
>> Discussion:  40 minutes
>=20
> Are the chairs confident that this topic can be resolved in this time?
> We managed to fritter a similar amount of time away without
> conclusion in the past.  I can see how you plan to accommodate
> overruns, but that just opens the possibility more time-wasting.  How
> do we ensure that this discussion actually concludes?

I can't guarantee that we'll conclude of course, but I'm hopeful we will =
get to a hum/vote one way or the other.  In my presentation I plan to =
propose a compromise, for example.  I think those against SDES will be =
ok with the compromise, but I'm not sure about those who want SDES.

-hadriel


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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] A compromise for SDES
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Howdy,
Someone's asked me to explain the compromise I plan on presenting in =
Berlin now, so we have more time to argue over it or chew on it.  The =
reason I was going to wait is because I wanted to explain how I got to =
the point of thinking such a compromise would make sense.  So I'll try =
to do that in this email, which will make this email long.  Nothin' much =
I can do about that.

Obviously this is all my personal opinion, not that of my employer, and =
not a statement claiming to be objective fact, etc.

Background:
I think many people would agree that SDES has some benefits compared to =
DTLS for those of us wishing to interface to the SIP world.  Some of =
those benefits are commercial/cost factors, some are more tangible like =
reducing early media clipping.  Whether you agree with such benefits or =
not, so long as WebRTC could support SDES in a sufficiently-secure =
manner, there'd be less concern over having it.  Most of the arguments =
against SDES have been directed at whether it could in fact be =
"sufficiently secure".  Personally I find most of the arguments against =
it being secure enough to be essentially FUD, and/or also apply to =
DTLS-SRTP.

On the flip-side, I think the commercial/cost factor for SDES is not a =
strong cause, because I believe the market will bear whatever the extra =
cost may be. (I wasn't sure of this 2 years ago when this all started, =
but I'm fairly confident of it now)  And I think having only DTLS-SRTP =
as MTI would have a marketing benefit for WebRTC as a technology, which =
I value highly.

Given those personal feelings, I didn't much care either way, so long as =
a decision was made - although I still preferred having SDES to avoid =
the early media clipping problem.  So when I was going to present on =
this topic back in the Vancouver meeting, my last slide basically said: =
"If we support SDES and it turns out later we were wrong, we can always =
rescind it".  This was based on the notion that browsers get updated all =
the time, especially for security issues.

But later it occurred to me that's actually the Achilles' heel of =
supporting SDES in WebRTC, for those of us wanting to gateway this stuff =
to SIP.  Imagine if it turns out we were wrong, and some =
expected-or-unexpected vulnerability is exploited against some large =
WebRTC service provider, and makes the news/slashdot.  Browser vendors =
would instantly have new code removing SDES support, and browsers in =
devices would be updated virtually overnight - some users may take a =
long time to upgrade their specific browsers on their specific devices - =
but many of them would do so within days.  That would be untenable for a =
WebRTC service provider.  Even the smaller Enterprises take a long time =
to upgrade code on their servers, so even for them changing their =
gateways to suddenly support DTLS-SRTP would be unrealistic.  Imagine =
what it would be for larger providers, who might even have to deploy =
more servers to handle the sudden additional overhead of DTLS-SRTP.  =
Meanwhile a growing percentage of their users can no longer use the =
service.  That's bad mojo.

The things that a WebRTC-to-SIP service provider can control, and =
arguably the much larger use-case for this stuff to begin with, are not =
really true "browsers" anyway - they're web-based-framework device Apps, =
provided by the service provider to begin with.  I.e., the apps they =
build using web application framework things like PhoneGap/Cordova, =
Appcelerator Titanium, etc.  I mean using a real Browser is nice for =
ad-hoc stuff, like being able to click on a "talk to a rep" button on a =
website, or a webex/meetecho type thing; but for real communication =
"service", whether it be as a subscriber of a traditional carrier, or =
Skype, or an employee of an Enterprise, or a call-center attendant, or =
whatever - for those most people would never want to have to open a =
browser just to receive/make calls.  They'd give you an app to use =
instead.

So it's the app use-case that would benefit the most from SDES, =
especially for issues like media clipping.  And the app use-case doesn't =
have the security concerns nor concerns for unforeseen overnight =
updates.

The Compromise:
So given that background, I was planning to propose that the security =
doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a =
statement that web-based application frameworks SHOULD also support =
SDES. (with text about why and how, etc.)

I don't think this is too controversial, because web-based frameworks =
are never beholden to following browser behavior anyway - they're used =
to build a native application, and native applications have a very =
different security/threat model in practice.  They're written for a =
specific purpose, and installed by users from known sources for that =
known purpose.  Technically, afaik, nothing we do in RTCWEB WG or W3C's =
WEBRTC group have any requirements/mandates for native applications =
anyway - an app maker would just ignore something they don't think =
applies to them - but I think web-based frameworks do generally try to =
follow W3C models for Javascript APIs, and will likely read the IETF =
RFCs for the media-layer stuff too.  So I think having this SHOULD =
statement would be beneficial.

Or if the WG prefers, we could even write a separate doc on what a =
web-based framework maker should consider supporting/not-supporting. (I =
can imagine a few other things they might want to offer that a browser =
can't)

-hadriel


From stefan.lk.hakansson@ericsson.com  Sat Jul 13 11:39:26 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Draft agenda for IETF 87
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87
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On 7/12/13 7:19 PM, Peter Thatcher wrote:=0A=
>=0A=
>=0A=
>=0A=
> On Fri, Jul 12, 2013 at 12:35 AM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     On 7/11/13 9:38 PM, Martin Thomson wrote:=0A=
>      > On 11 July 2013 12:04, Cullen Jennings (fluffy) <fluffy@cisco.com=
=0A=
>     <mailto:fluffy@cisco.com>>=0A=
>      > wrote:=0A=
>      >> On Jul 11, 2013, at 10:35 AM, Emil Ivov <emcho@jitsi.org=0A=
>     <mailto:emcho@jitsi.org>>=0A=
>      >>> In you last mail on the subject you mentioned that we will be=0A=
>      >>> discussing No Plan in Berlin together with plans A and B. Could=
=0A=
>      >>> we please add this to the agenda?=0A=
>      >>=0A=
>      >> No. We believe that conversation needs to happen in the W3C WebRT=
C=0A=
>      >> WG. I expect to see a message from W3C chairs on this at some=0A=
>      >> point.=0A=
>      >=0A=
>      > I'm a little nervous about this.  Where does the decision on the=
=0A=
>      > separation of responsibilities (API vs. SDP) get made?=0A=
>=0A=
>     (I have not been able to confer with any of the other chairs, so this=
 is=0A=
>     just my personal opinion):=0A=
>=0A=
>     To me it seems pretty straightforward to a certain point:=0A=
>=0A=
>     * The SDP (if we opt to continue using SDP for this purpose) that goe=
s=0A=
>     on the signaling wire between the browsers is defined by IETF (and by=
=0A=
>     the rtcweb WG I presume even though MMUSIC seems to have some stake)=
=0A=
>=0A=
>     * JS APIs to:=0A=
>     ** Apply an SDP (e.g. received on the signaling channel) to the brows=
er=0A=
>     ** Hand an SDP generated by the browser over to the application (for=
=0A=
>     transmission over the signaling wire presumably)=0A=
>     ** Influencing/modifying the contents of the SDP=0A=
>     * All belongs to the W3C WebRTC=0A=
>=0A=
>=0A=
> Would it make sense to change this list to say the following?=0A=
>=0A=
> * JS APIs to:=0A=
> ** Apply a SessionDescription (created by SDP; eg received on the=0A=
> signaling channel or built by JS) to the browser=0A=
> ** Hand a SessionDescription (serializable to SDP) generated by the=0A=
> browser over to the application (for=0A=
> transmission over the signaling wire presumably or for immediately=0A=
> applying with setLocalDescription for some local effect)=0A=
> ** Influencing/modifying the contents of the SessionDescription (and=0A=
> thus, SDP, when serialized to SDP)=0A=
> * All belongs to the W3C WebRTC=0A=
=0A=
That might well be a better way to put it.=0A=
=0A=
>=0A=
>=0A=
> My changes highlight a few things:=0A=
> 1.  The API (setLocalDescripiton, setRemoteDescription, creatOffer, etc)=
=0A=
> doesn't work with SDP directly.  It works with RTCSessionDescription=0A=
> objects.  It just happens to be that currently the only way to interact=
=0A=
> with RTCSessionDescription objects is through SDP.  I think it's worth=0A=
> remembering that distinction.=0A=
>=0A=
> 2.  SDP does not necessarily go over the wire.   It only necessarily=0A=
> goes through the API between JS and browser.   I think it's worth=0A=
> remembering that distinction.=0A=
>=0A=
> 3.  Calling setLocalDescription with a new SessionDescription object can=
=0A=
> have purely local effects that don't need to be sent over the wire.  I=0A=
> think it's worth remembering those cases.=0A=
>=0A=
>=0A=
>     What seems unclear to me is where we define what modifications to the=
=0A=
>     SDP that are allowed - and when. Even though the ambition is to have=
=0A=
>     APIs that makes SDP mangling an exception, we will still see that=0A=
>     happening.=0A=
>=0A=
>=0A=
> I think advanced JS apps are going to use every control knob they can=0A=
> get to, whether it's anticipated and well-supported or not.    I think=0A=
> there's a good chance that a a popular WebRTC web app will use some SDP=
=0A=
> mangling that wasn't anticipated, but happened to work, but then the=0A=
> browser can't remove it because it will break certain websites.  It=0A=
> could get even worse if someone writes a popular WebRTC wrapper library=
=0A=
> that uses tricky SDP mangling that is then used by lots of websites.=0A=
>   Certain SDP mangling techniques might end up becoming a defacto=0A=
> standard API that can't be removed, even if it was originally a bug.  Or=
=0A=
> worse, one browser will have to implement the SDP mangling or even the=0A=
> bugs of another, because WebRTC apps have come to rely on them.=0A=
>=0A=
> In fact, it's already the case that Chrome and Firefox support far=0A=
> different SDP manglings.  I don't think any web apps rely on that yet,=0A=
> but it's only a matter of time before someone figures out "hey, if I=0A=
> mangle the SDP on this browser this way and on that browser that way, I=
=0A=
> can do things I couldn't do otherwise".  Or worse, an advanced web app=0A=
> developer says "hey, I can make this work well on browser X via SDP=0A=
> mangling, but not on browser Y, so I'll put a 'best used with Browser X'=
=0A=
> icon on my website". Then that someone writes an abstraction on top of=0A=
> that, and then maybe shares that with others, and it goes from there.=0A=
>=0A=
> I think we're in a race with web developers to see if they'll figure out=
=0A=
> SDP mangling before we provide a way to avoid SDP mangling.   Who do you=
=0A=
> think moves faster?=0A=
=0A=
Let's go over to the public-webrtc W3C space and work on this.=0A=
=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>      > _______________________________________________ rtcweb mailing lis=
t=0A=
>      > rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>      >=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
=0A=

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
Thread-Index: AQHOeDFbTNRhHYI3NkSZeqhrbqmeQg==
Date: Sat, 13 Jul 2013 18:40:23 +0000
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References: <CAJrXDUGMohpBdi-ft-o_uE7ewFkw7wRY9x7gYEncjov7qi-Bew@mail.gmail.com> <CABcZeBPa4wBS8pYq=0wesMOfL6TkeC7QGAZ8pWwOcnkhkJqWfA@mail.gmail.com> <CAJrXDUFxo8P8wxh8jX3019yPQOuwQ0eVdsFmRXsbWdWinnc5oA@mail.gmail.com> <CABcZeBOTKpmFC34waqZ4kA-P8t+E6yY9gX1JFCHhsBH0+CF-Qw@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30BC0F@ESESSMB209.ericsson.se> <CAD5OKxtKLMf_d=8GSMrqfNhDHPe9MFP2ZTKzZHFn9CyMr-gSVQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30C833@ESESSMB209.ericsson.se> <CAJrXDUHOZf21aXgQMrdjTV8Fok+fVp-2SuhTra0JGy0Jq=Wi0Q@mail.gmail.com> <CAHp8n2mNdNiXCCNUdEvKgAsh_pPn=jNs+56VCg4PMKbUmOGztQ@mail.gmail.com> <CABkgnnUZMAuDocwZWXn9a+Xj3kkcX0uyRgjDmy-hQxpTDKWj3w@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30CF49@ESESSMB209.ericsson.se> <CALiegfmiRsOXL97XDzMRQ_Vvbk9zaDBBvCPxr_=zbDJbnMZ_8A@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30D110@ESESSMB209.ericsson.se> <CAJrXDUGNCrN5kgmrd4YK=7FtDd-LwA54aykJUP9_3sGSB42TkQ@mail.gmail.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/12/13 7:32 PM, Peter Thatcher wrote:=0A=
>=0A=
>=0A=
>=0A=
> On Tue, Jul 9, 2013 at 4:26 AM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     On 2013-07-09 12:18, I=F1aki Baz Castillo wrote:=0A=
>      > 2013/7/9 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com=
=0A=
>     <mailto:stefan.lk.hakansson@ericsson.com>>:=0A=
>      >> I want to be very clear and careful on what I say. So I am=0A=
>     repeating:=0A=
>      >>=0A=
>      >> * My comment that I think Eric is right in that there is=0A=
>     consensus on=0A=
>      >> providing APIs that allow most use-cases to be met without SDP=0A=
>     mangling=0A=
>      >> is meant only in that context: SDP O/A is kept (and=0A=
>     PeerConnection more=0A=
>      >> or less as is and so on).=0A=
>      >=0A=
>      > Hi Stefan,=0A=
>      >=0A=
>      > You insist that "there is consensus on keeping SDP O/A but=0A=
>     providing a=0A=
>      > better API".=0A=
>=0A=
>     I must be expressing myself quite badly, because the message seems no=
t=0A=
>     to get through.=0A=
>=0A=
>     * I think we have consensus on adding API that allows most use-cases =
to=0A=
>     be met without SDP mangling (given that SDP O/A is kept naturally)=0A=
>=0A=
>     * We discussed last year an alternative API that did not use SDP at a=
ll=0A=
>     (and did not use PeerConnection). The decision at that time was to=0A=
>     continue developing the API based on PeerConnection and SDP O/A.=0A=
>=0A=
>     Have not stated anything more than that.=0A=
>=0A=
>=0A=
> Thank you for clarifying, Stefan.  It's often very difficult for those=0A=
> of us who weren't there to understand what was consensus, and what=0A=
> wasn't, and what was decided and what was decided and what wasn't.  It=0A=
> seems that if you ask 10 people, you get 10 different answers.=0A=
>=0A=
>=0A=
> In other words, it's hard to see much consensus on what we have=0A=
> consensus on.=0A=
>=0A=
>=0A=
> So thank you for your clarifications.  It's probably not a very fun job=
=0A=
> for you :).   Along with that, I have two more questions for you:=0A=
>=0A=
>=0A=
> 1.  You said we have consensus on adding API that allows most use-cases=
=0A=
> to be met without SDP mangling.  Since that point in time has there been=
=0A=
> in any progress in adding such APIs?  Can you give me a rough feel for=0A=
> how much time has passed, how many proposals have been made, how many=0A=
> have been accepted, and how many have been implemented?  I think it's=0A=
> great that we'll do this "someday", but I'd like to get a feel for how=0A=
> far away "someday" is.  Thanks in advance.=0A=
>=0A=
> 2.  You said we have consensus against an alternative API.  Was that=0A=
> consensus against any additions to the API that would allow JS to bypass=
=0A=
> SDP, or was that a consensus against just that particular alternative=0A=
> API?  I'd like to get a feel for how many people voted for "no; let's=0A=
> not go down that specific road" vs. "no; we cannot go on any road but=0A=
> this one".   I think there's a big difference between the two.  Thanks=0A=
> again in advance.=0A=
>=0A=
>=0A=
> By the way, Ted asked us to move discussions about the API to=0A=
> public-webrtc, so should we do that now?=0A=
=0A=
Yes. I will respond there (probably tomorrow).=0A=
=0A=
>=0A=
>=0A=
>=0A=
>     Stefan=0A=
>=0A=
>      > Given current discussions IMHO it is clear there is not=0A=
>      > such a consensus (not at all). Or may be you are just talking abou=
t=0A=
>      > two years ago in some IETF meeting (if so I'm sorry).=0A=
>      >=0A=
>      > Please review the results in=0A=
>      >=0A=
>     https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkH=
MSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1=0A=
>      >=0A=
>      >    Bad for advanced stuff: 94%=0A=
>      >    OK for 1.0: 40%=0A=
>      >=0A=
>      > IMHO such a result indicates all but "let's keep SDP O/A and=0A=
>     improve a=0A=
>      > bit the API" (I mean now, in July 2013).=0A=
>      >=0A=
>      >=0A=
>      > Best regards.=0A=
>      >=0A=
>      >=0A=
>      >=0A=
>      > --=0A=
>      > I=F1aki Baz Castillo=0A=
>      > <ibc@aliax.net <mailto:ibc@aliax.net>>=0A=
>      >=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
=0A=

From bernard_aboba@hotmail.com  Sat Jul 13 14:02:36 2013
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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 13 Jul 2013 14:02:29 -0700
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Subject: Re: [rtcweb] A compromise for SDES
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--_01c938cf-2a92-4fe6-85ae-639d68afd89d_
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Hadriel --
I think you've brought up a critical point=2C which is very worth keeping i=
n mind throughout the discussions in IETF RTCWEB.  That is that WebRTC is r=
eally two distinct innovations -- one=2C a profile of "RAI 2.0" functionali=
ty=2C defining a core set of RTP functionality developed over the last deca=
de (e.g. end-to-end security=2C adaptive HD codecs=2C AVPF=2C etc.)=2C and =
the other=2C a set of Javascript APIs for Web browsers defined by the W3C. =
  IMHO=2C  the "RAI 2.0" aspect is likely to prove more important (and long=
-lived) than the W3C WEBRTC API aspect. =20
Since the "RAI 2.0" profile developed by the IETF RTCWEB WG is likely to li=
ve on way beyond the useful life of the W3C WEBRTC APIs=2C it is critical t=
hat the protocol profiles not be distorted so as to cater to API concerns t=
hat will be difficult to justify=2C let alone remember=2C a decade from now=
.   If the damage brought about by an SDP-based API could be restricted sol=
ely to Web browsers without reflecting itself in expected "on the wire" beh=
avior=2C I would care a lot less about it.  After all=2C the Internet knows=
 how to deal with bad designs -  it just "treats them as damage and routes =
around them".    So to the extent that IETF RTCWEB can contain the SDP API =
swamp we'll all be better off -- and the swamp will be that much easier to =
drain down the road once it becomes clear to all that nobody really wants t=
o live in that part of town anyway. =20
As you state=2C the vast majority of customers I talk to are interested pri=
marily in the "RAI 2.0" aspects of WebRTC=2C and only secondarily=2C if at =
all=2C in the W3C JS APIs.  Given the displacement of feature phones by sma=
rtphones=2C the growth in availability of wireless data as well as the incr=
easing availability of connectivity that can support interactive video (all=
 areas of very rapid growth over the next few years)=2C the focus of realti=
me communications app developers I talk to is not on browser=2C but on the =
development of native applications.  And unless HTML5-based approaches such=
 as Firefox OS gain traction=2C that focus is likely to remain.  =20
Given this=2C the most frequent request I hear is for a usable mobile SDK t=
hat can offer protocol compatibility with WebRTC implementations on browser=
s.  To succeed=2C that mobile SDK had better not look anything like the W3C=
 WEBRTC API=2C and it also better support extensibility.  In reality=2C all=
 that the IETF RTCWEB profile defines is the core set of functionality -- t=
hat set of features that MUST be supported to enable interoperability.  How=
ever=2C there is a lot more that a developer might want to have available f=
or their particular application.  That might include a codec (such as AMR-N=
B) or a security variant such as SDES/SRTP.   Since as you note many develo=
pers aren't going to be using the W3C WEBRTC APIs anyway=2C whether those a=
dditional functions are supported in the W3C APIs is largely irrelevant - a=
nd so are discussions about whether this or that additional feature needs t=
o be a "SHOULD" or a "MAY". =20

> So given that background=2C I was planning to propose that the security d=
oc keep DTLS-SRTP as the only MTI mechanism for browsers=2C BUT to add a st=
atement that web-based application frameworks SHOULD also support SDES. (wi=
th text about why and how=2C etc.)
>=20
> I don't think this is too controversial=2C because web-based frameworks a=
re never beholden to following browser behavior anyway - they're used to bu=
ild a native application=2C and native applications have a very different s=
ecurity/threat model in practice.  They're written for a specific purpose=
=2C and installed by users from known sources for that known purpose.  Tech=
nically=2C afaik=2C nothing we do in RTCWEB WG or W3C's WEBRTC group have a=
ny requirements/mandates for native applications anyway - an app maker woul=
d just ignore something they don't think applies to them - but I think web-=
based frameworks do generally try to follow W3C models for Javascript APIs=
=2C and will likely read the IETF RFCs for the media-layer stuff too.  So I=
 think having this SHOULD statement would be beneficial.

 		 	   		  =

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<html>
<head>
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<body class=3D'hmmessage'><div dir=3D'ltr'><div>Hadriel --</div><div><br></=
div><div>I think you've brought up a critical point=2C which is very worth =
keeping in mind throughout the discussions in IETF RTCWEB. &nbsp=3BThat is =
that WebRTC is really two distinct innovations -- one=2C a profile of "RAI =
2.0" functionality=2C defining a core set of RTP functionality developed ov=
er the last decade (e.g. end-to-end security=2C adaptive HD codecs=2C AVPF=
=2C etc.)=2C and the other=2C a set of Javascript APIs for Web browsers def=
ined by the W3C. &nbsp=3B&nbsp=3B<span style=3D"font-size: 12pt=3B ">IMHO=
=2C &nbsp=3Bthe "RAI 2.0" aspect is likely to prove more important (and lon=
g-lived) than the W3C WEBRTC API aspect. &nbsp=3B</span></div><div><span st=
yle=3D"font-size: 12pt=3B "><br></span></div><div><span style=3D"font-size:=
 12pt=3B ">Since the "RAI 2.0" profile developed by the IETF RTCWEB WG is l=
ikely to live on way beyond the useful life of the W3C WEBRTC APIs=2C it is=
 critical that the protocol profiles not be distorted so as to cater to API=
 concerns that will be difficult to justify=2C let alone remember=2C a deca=
de from now. &nbsp=3B If t</span><span style=3D"font-size: 12pt=3B ">he dam=
age brought about by an SDP-based API could be restricted solely to Web bro=
wsers without reflecting itself in expected "on the wire" behavior=2C I wou=
ld care a lot less about it. &nbsp=3BAfter all=2C&nbsp=3B</span><span style=
=3D"font-size: 12pt=3B ">the Internet knows how to deal with bad designs - =
&nbsp=3Bit just "treats them as damage and routes around them". &nbsp=3B &n=
bsp=3BSo to the extent that IETF RTCWEB can contain the SDP API swamp we'll=
 all be better off -- and the swamp will be that much easier to drain down =
the road once it becomes clear to all that nobody really wants to live in t=
hat part of town anyway. &nbsp=3B</span></div><div><span style=3D"font-size=
: 12pt=3B "><br></span></div><div><span style=3D"font-size: 12pt=3B ">As yo=
u state=2C the vast majority of customers I talk to are interested primaril=
y in the "RAI 2.0" aspects of WebRTC=2C and only secondarily=2C if at all=
=2C in the W3C JS APIs. &nbsp=3BGiven the displacement of feature phones by=
 smartphones=2C the growth in availability of wireless data as well as the =
increasing availability of connectivity that can support interactive video =
(all areas of very rapid growth over the next few years)=2C the focus of re=
altime communications app developers I talk to is not on browser=2C but on =
the development of native applications. &nbsp=3BAnd unless HTML5-based appr=
oaches such as Firefox OS gain traction=2C that focus is likely to remain. =
&nbsp=3B&nbsp=3B</span></div><div><span style=3D"font-size: 12pt=3B "><br><=
/span></div><div><span style=3D"font-size: 12pt=3B ">Given this=2C the most=
 frequent request I hear is for a</span><span style=3D"font-size: 12pt=3B "=
>&nbsp=3Busable mobile SDK that can offer protocol compatibility with WebRT=
C implementations on browsers. &nbsp=3BTo succeed=2C that mobile SDK had be=
tter not look anything like the W3C WEBRTC API=2C and it also better suppor=
t extensibility. &nbsp=3BIn reality=2C all that the IETF RTCWEB profile def=
ines is the core set of functionality -- that set of features that MUST be =
supported to enable interoperability. &nbsp=3BHowever=2C there is a lot mor=
e that a developer might want to have available for their particular applic=
ation. &nbsp=3B</span><span style=3D"font-size: 12pt=3B ">That might includ=
e a codec (such as AMR-NB) or a security variant such as SDES/SRTP. &nbsp=
=3B Since as you note many developers aren't going to be using the W3C WEBR=
TC APIs anyway=2C whether those additional functions are supported in the W=
3C APIs is largely irrelevant - and so are discussions about whether this o=
r that additional feature needs to be a "SHOULD" or a "MAY". &nbsp=3B</span=
></div><div><br></div><div><br></div><div>&gt=3B So given that background=
=2C I was planning to propose that the security doc keep DTLS-SRTP as the o=
nly MTI mechanism for browsers=2C BUT to add a statement that web-based app=
lication frameworks SHOULD also support SDES. (with text about why and how=
=2C etc.)<br>&gt=3B <br>&gt=3B I don't think this is too controversial=2C b=
ecause web-based frameworks are never beholden to following browser behavio=
r anyway - they're used to build a native application=2C and native applica=
tions have a very different security/threat model in practice.  They're wri=
tten for a specific purpose=2C and installed by users from known sources fo=
r that known purpose.  Technically=2C afaik=2C nothing we do in RTCWEB WG o=
r W3C's WEBRTC group have any requirements/mandates for native applications=
 anyway - an app maker would just ignore something they don't think applies=
 to them - but I think web-based frameworks do generally try to follow W3C =
models for Javascript APIs=2C and will likely read the IETF RFCs for the me=
dia-layer stuff too.  So I think having this SHOULD statement would be bene=
ficial.<br><br></div> 		 	   		  </div></body>
</html>=

--_01c938cf-2a92-4fe6-85ae-639d68afd89d_--

From stefan.lk.hakansson@ericsson.com  Sat Jul 13 23:58:42 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] A compromise for SDES
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On 7/13/13 11:02 PM, Bernard Aboba wrote:=0A=
> Hadriel --=0A=
>=0A=
> I think you've brought up a critical point, which is very worth keeping=
=0A=
> in mind throughout the discussions in IETF RTCWEB.  That is that WebRTC=
=0A=
> is really two distinct innovations -- one, a profile of "RAI 2.0"=0A=
> functionality, defining a core set of RTP functionality developed over=0A=
> the last decade (e.g. end-to-end security, adaptive HD codecs, AVPF,=0A=
> etc.), and the other, a set of Javascript APIs for Web browsers defined=
=0A=
> by the W3C.IMHO,  the "RAI 2.0" aspect is likely to prove more=0A=
 > important (and long-lived) than the W3C WEBRTC API aspect.=0A=
=0A=
=0A=
I think that the Javascript APIs are going to be very important, but I =0A=
fully agree that the "RAI 2.0" profile defined will be used beyond =0A=
browsers. And I think that the more self contained (i.e. not dependent =0A=
on other signaling during operation) we can make that RAI 2.0 profile =0A=
the better.=0A=
=0A=
So I am again arguing for pushing things like pause/resume signaling to =0A=
RTP/RCTP, and not depend on out of band exchanges of signaling messages.=0A=
=0A=
=0A=
=0A=

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To: =?ISO-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  Hadriel Kaplan <hadriel.kaplan@oracle.com>, Bernard Aboba <bernard_aboba@hotmail.com>
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On Sun, Jul 14, 2013 at 2:58 AM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 7/13/13 11:02 PM, Bernard Aboba wrote:
> > Hadriel --
> >
> > I think you've brought up a critical point, which is very worth keeping
> > in mind throughout the discussions in IETF RTCWEB.  That is that WebRTC
> > is really two distinct innovations -- one, a profile of "RAI 2.0"
> > functionality, defining a core set of RTP functionality developed over
> > the last decade (e.g. end-to-end security, adaptive HD codecs, AVPF,
> > etc.), and the other, a set of Javascript APIs for Web browsers defined
> > by the W3C.IMHO,  the "RAI 2.0" aspect is likely to prove more
>  > important (and long-lived) than the W3C WEBRTC API aspect.
>
>
> I think that the Javascript APIs are going to be very important, but I
> fully agree that the "RAI 2.0" profile defined will be used beyond
> browsers. And I think that the more self contained (i.e. not dependent
> on other signaling during operation) we can make that RAI 2.0 profile
> the better.
>
>
This point, even though valid, might have unintended consequences. For
instance, if support for RTP based real time text in web browsers is
questionable, due to other ways to implement it via existing communications
channels, it should probably be part of "RAI 2.0". I think there are going
to be quite a few features, which will not be part of core webrtc
functionality which will fall into "RAI 2.0" category. Discussing these
features can slow down this group even more then it already is, and is most
likely be outside of this group charter.

Furthermore, according to the same logic, Hadriel's compromise can be
boiled down to not supporting SDES in WebRTC, with all the other language
essentially discussing things that will not controlled by WebRTC drafts and
lay outside this group charter.
_____________
Roman Shpount

--001a11c3669e6cf06804e179b1d5
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<div>On Sun, Jul 14, 2013 at 2:58 AM, Stefan H=E5kansson LK <span dir=3D"lt=
r">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank=
">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:</div><div class=3D=
"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/13/13 11:02 PM, Berna=
rd Aboba wrote:<br>
&gt; Hadriel --<br>
&gt;<br>
&gt; I think you&#39;ve brought up a critical point, which is very worth ke=
eping<br>
&gt; in mind throughout the discussions in IETF RTCWEB. =A0That is that Web=
RTC<br>
&gt; is really two distinct innovations -- one, a profile of &quot;RAI 2.0&=
quot;<br>
&gt; functionality, defining a core set of RTP functionality developed over=
<br>
&gt; the last decade (e.g. end-to-end security, adaptive HD codecs, AVPF,<b=
r>
&gt; etc.), and the other, a set of Javascript APIs for Web browsers define=
d<br>
</div>&gt; by the W3C.IMHO, =A0the &quot;RAI 2.0&quot; aspect is likely to =
prove more<br>
<div class=3D"im">=A0&gt; important (and long-lived) than the W3C WEBRTC AP=
I aspect.<br>
<br>
<br>
</div>I think that the Javascript APIs are going to be very important, but =
I<br>
fully agree that the &quot;RAI 2.0&quot; profile defined will be used beyon=
d<br>
browsers. And I think that the more self contained (i.e. not dependent<br>
on other signaling during operation) we can make that RAI 2.0 profile<br>
the better.<br>
<br></blockquote><div><br></div><div>This point, even though valid, might h=
ave unintended consequences. For instance, if support for RTP based real ti=
me text in web browsers is questionable, due to other ways to implement it =
via existing communications channels, it should probably be part of &quot;R=
AI 2.0&quot;. I think there are going to be quite a few features, which wil=
l not be part of core webrtc functionality which will fall into &quot;RAI 2=
.0&quot; category. Discussing these features can slow down this group even =
more then it already is, and is most likely be outside of this group charte=
r.</div>
<div><br></div><div>Furthermore, according to the same logic, Hadriel&#39;s=
 compromise can be boiled down to not supporting SDES in WebRTC, with all t=
he other language essentially discussing things that will not controlled by=
 WebRTC drafts and lay outside this group charter.</div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--001a11c3669e6cf06804e179b1d5--

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Very interesting discussion and important thread. I have a comment on JS
API's.

On Sun, Jul 14, 2013 at 2:32 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> Hadriel --
>
> .....
>
> As you state, the vast majority of customers I talk to are interested
> primarily in the "RAI 2.0" aspects of WebRTC, and only secondarily, if at
> all, in the W3C JS APIs.  Given the displacement of feature phones by
> smartphones, the growth in availability of wireless data as well as the
> increasing availability of connectivity that can support interactive video
> (all areas of very rapid growth over the next few years), the focus of
> realtime communications app developers I talk to is not on browser, but on
> the development of native applications.  And unless HTML5-based approaches
> such as Firefox OS gain traction, that focus is likely to remain.
>
>
WebRTC JS API's has the potential to make video/audio based web apps the
first class citizens of internet. Focus on interop at the media plane and
signaling plane is important. But it is critical to focus and bring better
JS API's to browser and browser frameworks. I am startup and many customers
I talk to are asking for video web apps on Internet explorer which
unfortunately is lacking as compared to Chrome and Firefox (both of which
are available on mobile too).

regards,
Rishi
OneKlikStreet.com




>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div>Very interesting discussion and important thread. I h=
ave a comment on JS API&#39;s.</div><br><div class=3D"gmail_extra">On Sun, =
Jul 14, 2013 at 2:32 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_aboba@hotmail.com<=
/a>&gt;</span> wrote:<br>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">


<div><div dir=3D"ltr"><div>Hadriel --</div><div><br></div><div>.....</div><=
div><span style=3D"font-size:12pt"><br></span></div><div><span style=3D"fon=
t-size:12pt">As you state, the vast majority of customers I talk to are int=
erested primarily in the &quot;RAI 2.0&quot; aspects of WebRTC, and only se=
condarily, if at all, in the W3C JS APIs. =A0Given the displacement of feat=
ure phones by smartphones, the growth in availability of wireless data as w=
ell as the increasing availability of connectivity that can support interac=
tive video (all areas of very rapid growth over the next few years), the fo=
cus of realtime communications app developers I talk to is not on browser, =
but on the development of native applications. =A0And unless HTML5-based ap=
proaches such as Firefox OS gain traction, that focus is likely to remain. =
=A0=A0</span></div>
<div><br></div></div></div></blockquote><div><br></div><div>WebRTC JS API&#=
39;s has the potential to make video/audio based web apps the first class c=
itizens of internet. Focus on interop at the media plane and signaling plan=
e is important. But it is critical to focus and bring better JS API&#39;s t=
o browser and browser frameworks. I am startup and many customers I talk to=
 are asking for video web apps on Internet explorer which unfortunately is =
lacking as compared to Chrome and Firefox (both of which are available on m=
obile too).=A0</div>
<div><br></div><div>regards,</div><div>Rishi</div><div>OneKlikStreet.com</d=
iv><div>=A0</div><div><br></div><div>=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-co=
lor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div><div dir=3D"ltr"><div></div><div class=3D"im"><div><br></div> 		 	   	=
	  </div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--20cf3010ea091ee74604e17bb964--

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 15 Jul 2013 05:46:34 -0700
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On Sat, Jul 13, 2013 at 2:02 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> Hadriel --
>
> I think you've brought up a critical point, which is very worth keeping in
> mind throughout the discussions in IETF RTCWEB.  That is that WebRTC is
> really two distinct innovations -- one, a profile of "RAI 2.0"
> functionality, defining a core set of RTP functionality developed over the
> last decade (e.g. end-to-end security, adaptive HD codecs, AVPF, etc.), and
> the other, a set of Javascript APIs for Web browsers defined by the W3C.   IMHO,
>  the "RAI 2.0" aspect is likely to prove more important (and long-lived)
> than the W3C WEBRTC API aspect.
>
> Since the "RAI 2.0" profile developed by the IETF RTCWEB WG is likely to
> live on way beyond the useful life of the W3C WEBRTC APIs, it is critical
> that the protocol profiles not be distorted so as to cater to API concerns
> that will be difficult to justify, let alone remember, a decade from now.
> If the damage brought about by an SDP-based API could be restricted
> solely to Web browsers without reflecting itself in expected "on the wire"
> behavior, I would care a lot less about it.  After all, the Internet
> knows how to deal with bad designs -  it just "treats them as damage and
> routes around them".    So to the extent that IETF RTCWEB can contain the
> SDP API swamp we'll all be better off -- and the swamp will be that much
> easier to drain down the road once it becomes clear to all that nobody
> really wants to live in that part of town anyway.
>
>



> As you state, the vast majority of customers I talk to are interested
> primarily in the "RAI 2.0" aspects of WebRTC, and only secondarily, if at
> all, in the W3C JS APIs.  Given the displacement of feature phones by
> smartphones, the growth in availability of wireless data as well as the
> increasing availability of connectivity that can support interactive video
> (all areas of very rapid growth over the next few years), the focus of
> realtime communications app developers I talk to is not on browser, but on
> the development of native applications.  And unless HTML5-based approaches
> such as Firefox OS gain traction, that focus is likely to remain.
>
> Given this, the most frequent request I hear is for a usable mobile SDK
> that can offer protocol compatibility with WebRTC implementations on
> browsers.  To succeed, that mobile SDK had better not look anything like
> the W3C WEBRTC API, and it also better support extensibility.
>

Chrome's implementation of WebRTC, which can be found at webrtc.org, has a
mobile SDK that can be used to build mobile apps (Android and iOS, at
least).  Some well-known and popular RTC mobile apps are in large part
built on that code and can interop to a degree with browsers.  So in some
sense, you already have what you want.

The big problem is that the high-level API offered by that SDK exactly
matches the one in the browser, which means it's the gross SDP-based one.
 And none of the mobile apps I know, beyond an example app, use that API,
or have plans to, precisely because it's gross.  Instead, as far as I know,
they dig deeper into the code and avoid that part. (why jam everything
through SDP if you don't have to?).

I think the optimal scenario for us to target is to define a good API that
is the same between mobile and web browsers.  I think that would help
developers the most.  Making such a "web and mobile" API is probably out of
scope for any WG, but I think that if we defined a good (non SDP-based) API
for the browser, it could easily be exposed as a native SDK (just like it
is now), but be much less gross and more usable.

I don't know if such a thing (a good API for both mobile and web) can
realistically ever happen, but in the meantime, you can at least pick up
the code at webrtc.org and make a mobile app.  It's open source, too, so if
you want to make a better API than what's there, go ahead and write one :).





In reality, all that the IETF RTCWEB profile defines is the core set of
> functionality -- that set of features that MUST be supported to enable
> interoperability.  However, there is a lot more that a developer might want
> to have available for their particular application.  That might include a
> codec (such as AMR-NB) or a security variant such as SDES/SRTP.   Since as
> you note many developers aren't going to be using the W3C WEBRTC APIs
> anyway, whether those additional functions are supported in the W3C APIs is
> largely irrelevant - and so are discussions about whether this or that
> additional feature needs to be a "SHOULD" or a "MAY".
>

FYI, the code at webrtc.org supports SDES/SRTP, and while it doesn't have
support for AMR-NB (as far as I know), it does have a system for adding new
codecs, so you could theoretically take an implementation of AMR-NB and
compile it in yourself to make a mobile app.  It may not be very easy, but
it is possible.


>
>
> > So given that background, I was planning to propose that the security
> doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
> statement that web-based application frameworks SHOULD also support SDES.
> (with text about why and how, etc.)
> >
> > I don't think this is too controversial, because web-based frameworks
> are never beholden to following browser behavior anyway - they're used to
> build a native application, and native applications have a very different
> security/threat model in practice. They're written for a specific purpose,
> and installed by users from known sources for that known purpose.
> Technically, afaik, nothing we do in RTCWEB WG or W3C's WEBRTC group have
> any requirements/mandates for native applications anyway - an app maker
> would just ignore something they don't think applies to them - but I think
> web-based frameworks do generally try to follow W3C models for Javascript
> APIs, and will likely read the IETF RFCs for the media-layer stuff too. So
> I think having this SHOULD statement would be beneficial.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7b111cff82ace904e18c427f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sat, Jul 13, 2013 at 2:02 PM, Bernard Aboba <span dir=3D"ltr">&l=
t;<a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_ab=
oba@hotmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">


<div><div dir=3D"ltr"><div>Hadriel --</div><div><br></div><div>I think you&=
#39;ve brought up a critical point, which is very worth keeping in mind thr=
oughout the discussions in IETF RTCWEB. =C2=A0That is that WebRTC is really=
 two distinct innovations -- one, a profile of &quot;RAI 2.0&quot; function=
ality, defining a core set of RTP functionality developed over the last dec=
ade (e.g. end-to-end security, adaptive HD codecs, AVPF, etc.), and the oth=
er, a set of Javascript APIs for Web browsers defined by the W3C. =C2=A0=C2=
=A0<span style=3D"font-size:12pt">IMHO, =C2=A0the &quot;RAI 2.0&quot; aspec=
t is likely to prove more important (and long-lived) than the W3C WEBRTC AP=
I aspect. =C2=A0</span></div>

<div><span style=3D"font-size:12pt"><br></span></div><div><span style=3D"fo=
nt-size:12pt">Since the &quot;RAI 2.0&quot; profile developed by the IETF R=
TCWEB WG is likely to live on way beyond the useful life of the W3C WEBRTC =
APIs, it is critical that the protocol profiles not be distorted so as to c=
ater to API concerns that will be difficult to justify, let alone remember,=
 a decade from now. =C2=A0 If t</span><span style=3D"font-size:12pt">he dam=
age brought about by an SDP-based API could be restricted solely to Web bro=
wsers without reflecting itself in expected &quot;on the wire&quot; behavio=
r, I would care a lot less about it. =C2=A0After all,=C2=A0</span><span sty=
le=3D"font-size:12pt">the Internet knows how to deal with bad designs - =C2=
=A0it just &quot;treats them as damage and routes around them&quot;. =C2=A0=
 =C2=A0So to the extent that IETF RTCWEB can contain the SDP API swamp we&#=
39;ll all be better off -- and the swamp will be that much easier to drain =
down the road once it becomes clear to all that nobody really wants to live=
 in that part of town anyway. =C2=A0</span></div>

<div><span style=3D"font-size:12pt"><br></span></div></div></div></blockquo=
te><div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-lef=
t-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div><div dir=3D"ltr"><div><span style=3D"font-size:12pt"></span></div><div=
><span style=3D"font-size:12pt">As you state, the vast majority of customer=
s I talk to are interested primarily in the &quot;RAI 2.0&quot; aspects of =
WebRTC, and only secondarily, if at all, in the W3C JS APIs. =C2=A0Given th=
e displacement of feature phones by smartphones, the growth in availability=
 of wireless data as well as the increasing availability of connectivity th=
at can support interactive video (all areas of very rapid growth over the n=
ext few years), the focus of realtime communications app developers I talk =
to is not on browser, but on the development of native applications. =C2=A0=
And unless HTML5-based approaches such as Firefox OS gain traction, that fo=
cus is likely to remain. =C2=A0=C2=A0</span></div>

<div><span style=3D"font-size:12pt"><br></span></div><div><span style=3D"fo=
nt-size:12pt">Given this, the most frequent request I hear is for a</span><=
span style=3D"font-size:12pt">=C2=A0usable mobile SDK that can offer protoc=
ol compatibility with WebRTC implementations on browsers. =C2=A0To succeed,=
 that mobile SDK had better not look anything like the W3C WEBRTC API, and =
it also better support extensibility. =C2=A0</span></div>

</div></div></blockquote><div><br></div><div>Chrome&#39;s implementation of=
 WebRTC, which can be found at <a href=3D"http://webrtc.org">webrtc.org</a>=
, has a mobile SDK that can be used to build mobile apps (Android and iOS, =
at least). =C2=A0Some well-known and popular RTC mobile apps are in large p=
art built on that code and can interop to a degree with browsers. =C2=A0So =
in some sense, you already have what you want.</div>

<div><br></div><div>The big problem is that the high-level API offered by t=
hat SDK exactly matches the one in the browser, which means it&#39;s the gr=
oss SDP-based one. =C2=A0And none of the mobile apps I know, beyond an exam=
ple app, use that API, or have plans to, precisely because it&#39;s gross. =
=C2=A0Instead, as far as I know, they dig deeper into the code and avoid th=
at part. (why jam everything through SDP if you don&#39;t have to?). =C2=A0=
</div>

<div><br></div><div>I think the optimal scenario for us to target is to def=
ine a good API that is the same between mobile and web browsers. =C2=A0I th=
ink that would help developers the most. =C2=A0Making such a &quot;web and =
mobile&quot; API is probably out of scope for any WG, but I think that if w=
e defined a good (non SDP-based) API for the browser, it could easily be ex=
posed as a native SDK (just like it is now), but be much less gross and mor=
e usable. =C2=A0</div>

<div><br></div><div>I don&#39;t know if such a thing (a good API for both m=
obile and web) can realistically ever happen, but in the meantime, you can =
at least pick up the code at <a href=3D"http://webrtc.org">webrtc.org</a> a=
nd make a mobile app. =C2=A0It&#39;s open source, too, so if you want to ma=
ke a better API than what&#39;s there, go ahead and write one :).</div>

<div><br></div><div><br></div><div><br></div><div><br></div><div><br></div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">

<div><div dir=3D"ltr"><div><span style=3D"font-size:12pt">In reality, all t=
hat the IETF RTCWEB profile defines is the core set of functionality -- tha=
t set of features that MUST be supported to enable interoperability. =C2=A0=
However, there is a lot more that a developer might want to have available =
for their particular application. =C2=A0</span><span style=3D"font-size:12p=
t">That might include a codec (such as AMR-NB) or a security variant such a=
s SDES/SRTP. =C2=A0 Since as you note many developers aren&#39;t going to b=
e using the W3C WEBRTC APIs anyway, whether those additional functions are =
supported in the W3C APIs is largely irrelevant - and so are discussions ab=
out whether this or that additional feature needs to be a &quot;SHOULD&quot=
; or a &quot;MAY&quot;. =C2=A0</span></div>

</div></div></blockquote><div><br></div><div>FYI, the code at <a href=3D"ht=
tp://webrtc.org">webrtc.org</a> supports SDES/SRTP, and while it doesn&#39;=
t have support for AMR-NB (as far as I know), it does have a system for add=
ing new codecs, so you could theoretically take an implementation of AMR-NB=
 and compile it in yourself to make a mobile app. =C2=A0It may not be very =
easy, but it is possible.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex"><div><div dir=3D"ltr"><div class=3D"im"><=
div><br>
</div>
<div><br></div><div>&gt; So given that background, I was planning to propos=
e that the security doc keep DTLS-SRTP as the only MTI mechanism for browse=
rs, BUT to add a statement that web-based application frameworks SHOULD als=
o support SDES. (with text about why and how, etc.)<br>

&gt; <br>&gt; I don&#39;t think this is too controversial, because web-base=
d frameworks are never beholden to following browser behavior anyway - they=
&#39;re used to build a native application, and native applications have a =
very different security/threat model in practice.  They&#39;re written for =
a specific purpose, and installed by users from known sources for that know=
n purpose.  Technically, afaik, nothing we do in RTCWEB WG or W3C&#39;s WEB=
RTC group have any requirements/mandates for native applications anyway - a=
n app maker would just ignore something they don&#39;t think applies to the=
m - but I think web-based frameworks do generally try to follow W3C models =
for Javascript APIs, and will likely read the IETF RFCs for the media-layer=
 stuff too.  So I think having this SHOULD statement would be beneficial.<b=
r>

<br></div> 		 	   		  </div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--047d7b111cff82ace904e18c427f--

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Security Considerations for WebRTC
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-05.txt
	Pages           : 24
	Date            : 2013-07-15

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers, generally called "WebRTC".  The major use cases
   for WebRTC technology are real-time audio and/or video calls, Web
   conferencing, and direct data transfer.  Unlike most conventional
   real-time systems (e.g., SIP-based soft phones) WebRTC communications
   are directly controlled by a Web server, which poses new security
   challenges.  For instance, a Web browser might expose a JavaScript
   API which allows a server to place a video call.  Unrestricted access
   to such an API would allow any site which a user visited to "bug" a
   user's computer, capturing any activity which passed in front of
   their camera.  This document defines the WebRTC threat model and
   analyzes the security threats of WebRTC in that model.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-security-05


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : WebRTC Security Architecture
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-07.txt
	Pages           : 43
	Date            : 2013-07-15

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (commonly
   called "WebRTC").  This document defines the security architecture
   for


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-security-arch-07


Internet-Drafts are also available by anonymous FTP at:
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I've just posted updated security documents reflecting most, but not all
the last call comments. There's still a few TODOs I need to resolve
F2F, (+ the SDES thing which I haven't done anything with, since we're
discussing it in Berlin).

Below is a consolidated response to people's comments. Please let me
know if you think I haven't responded adequately to yours.


DRAFT-IETF-RTCWEB-SECURITY
--------------------------
 WESTERLUND
> I have reviewed the security draft (-04) as an individual and have the
> following comments.

Thanks for your comments. Responses below.


> 1. Usage of acronym RTCWEB vs WebRTC. I thought we earlier had a
> conclusion that we would use WebRTC as the acronym for the complete
> solution? Am I remembering incorrectly?

Changed, except when referring to the WG.


> 2. I think the introduction should contain some pointer to the overview
> draft for reverse lookup purpose if one stumbles on the document.

Done.


> 3. Section 3.1:
> Similarly, while Flash SWFs can access the camera and microphone,
>    they explicitly require that the user consent to that access.
>
> Can you please expand the SWF acronym and perhaps add a reference?

The acronym isn't very useful, since I think it refers ot
"Shockwave Flash". I changed this to programs ("SWFs")

Added a reference.


> 4. Section 3.2:
> Many other resources are accessible but isolated.  For instance,
>    while scripts are allowed to make HTTP requests via the
>    XMLHttpRequest() API those requests are not allowed to be made to any
>    server, but rather solely to the same ORIGIN from whence the script
>    came.[RFC6454] (although CORS [CORS] and WebSockets [RFC6455]
>    provides a escape hatch from this restriction, as described below.)
>
> This above looks strange around the first period, which is followed by
> RFC6454 reference. If is is a new sentence, the main sentence contains
> only a reference.

Fixed.


> 5. Section 3.2
> This SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks
>    on server B via the user's browser, which protects both the user
>    (e.g., from misuse of his credentials) and the server (e.g., from DoS
>    attack).
>
> I think one can make clear that the server one protects are B in the end
> of the sentence. Although it reasonably clear from context, I think
> writing it like this:
>
> This SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks
>    on server B via the user's browser, which protects both the user
>    (e.g., from misuse of his credentials) and the server B (e.g., from
>    DoS attack).
>
> Is clearer.

Changed.


> 6. Section 4.1.1.3:
> In both of the previous cases, the user has a direct relationship
>    (though perhaps a transient one) with the target of the call.
>
> Is it really the "target of the call" or is it the calling service that
> the user has a direct relation ship with?

I removed this section entirely since we decided to do nothing
about this and I have seen other people be confused by it.


> 7. Section 4.2.1:
>
> [[ OPEN ISSUE:  Do we need some way of verifying the expected traffic
>    rate, not just consent to receive traffic at all.]]
>
> Regarding this issue, I think you could add something which makes this
> not required, namely that by requiring congestion control of all flows
> established by the browser, the browser tries to avoid persistent
> congestion and any overload attack is less effective. The attack is more
> in the vain of reducing any competing flows supported rate to a lower
> fair share rate, which in worst case is below what is required to
> support the service being the target of the attack.
>
> Thus I think the open issue question is a nice to have, but not required
> feature.

Agreed, and I think we have mostly come to consensus on this point,
so I have removed the open issue.


> 8. Section 4.2.4:
>
> I think this document is missing to discuss the privacy threats beyond
> IP location privacy? I think that should be added, especially as we are
> addressing some issues we know of.
>
> These other threats that we have discussed are associated with anonymous
> calling services and linking and fingerprinting browsers or users with
> anonymous calls. The threats we do know exist are for example RTCP CNAME
> generation, DTLS certifcates, and API fingerprinting. I think these
> needs to be raised as a threat.

Added a privacy considerations section.

-Ekr

-------------------------------------------------
OHLSSON
> Some minor comments on draft-ietf-rtcweb-security:

Thanks. Responses below.


> - Abstract
>
>     This document defines the RTC-Web threat model and defines an
>     architecture which provides security within that threat model
>
>   Isn't the architecture defined in draft-ietf-rtcweb-security-arch?

Good catch. Fixed.


> - Section 4.1.1.3, 3rd paragraph
>
>     However, this obviously presents a privacy challenge, as sites which
>     host advertisements in IFRAMEs often learn very little about whether
>     individual users clicked through to the ads, or even which ads were
>     presented.
>
>   What exactly is the privacy issue here? Is it that the hosting site
>   can eavesdrop on the call between the user and the advertiser since
>   IFRAMEs are not used?

I removed this section.


> - Section 4.3.1, 3rd paragraph
>
>     In addition, the system MUST NOT provide any APIs to extract either
>     long-term keying material or to directly access any stored traffic
keys.
>
>   Does this mean that SDES is out of the picture or does the sentence onl=
y
>   apply to DTLS(-SRTP)?

I don't think this really impacts SDES, in context, but I tried to add
some clarifying text.



JOHNSON
> Abstract and the duplicate text in Introduction: RTC-web turn into
> either RTCWEB or WebRTC, with WebRTC being preferred.
>
> "some Web server" - slangy "a web server"

Changed.


> Figure 1: might also want to show WebSockets as this is becoming very
> common and is mentioned later in the document without much context.

I added WebSockets.


> Also, do we want to show the signaling channel (the one referenced by
> the W3C WebRTC spec that transports the SDP offers and answers between
> browsers)?

I think it clutters the diagram, but I added some text.


> JS, popunder, and SWF are not defined in the draft

Defined JS and SWF and rewrote to not use popunder.


> 4.1.2. has a reference to draft-ietf-rescorla-rtcweb-generic-idp.
> What is the status of this draft?

Per WG discussion it has been folded into the security architecture.
Fixed.


> 4.3.  RFC 3711 SRTP is not mentioned in the opening paragraph, which
> seems strange, since it is the actual protocol providing the COMSEC
> properties mentioned, rather then DTLS or DTLS-SRTP for media.

Agreed. Fixed.


> 4.3.1. Might be useful to mention that this is sometimes called a
> "passive attack" whereas 4.3.2 is an "active attack".

Fixed.

> 4.3.2.2.There are a number of misleading statements about SAS in this
> section which should be corrected.  For example:
>
>  "This SAS is designed to be read over the voice channel and if
>  confirmed by both sides precludes MITM attack."  The SAS is to be
>  *compared* by the users - reading it out loud is only one possible
>  mechanism.

Rewritten as:

  ZRTP <xref target=3D"RFC6189"/> uses a "short authentication string" (SAS=
)
which is derived
  from the key agreement protocol. This SAS is designed to be compared
  by the users (e.g., read aloud over the the voice channel or
  transmitted via an out of band channel) and if confirmed by both sides
precludes MITM
  attack. The intention is that the SAS is used once and then key
  continuity (though a different mechanism from that discussed
  above) is used thereafter.



> "Moreover, it is possible for an attacker who controls the browser to
> allow the SAS to succeed and then simulate call failure and reconnect,
> trusting that the user will not notice that the "no SAS" indicator has
> been set (which seems likely)." - I don't quite know what this means.
> If SAS is used, the UI is the browser chrome, so if this is possible,
> it is just a badly designed UI not a protocol failure or issue.

Removed.


> "Even were SAS secure if used, it seems exceedingly unlikely that
> users will actually use it." - This reads like speculation.  There are
> users today using SAS, open source users and commercial users.  One
> thing is certain, if we don't provide SAS, then users will not use it.

Yes, I think this is fair. I rewrote this to be a bit more neutral.

   Additionally, it is unclear that users will actually use an SAS.
   As discussed above, the browser UI constraints preclude requiring
   the SAS exchange prior to completing the call and so it must be
   voluntary; at most the browser will provide some UI indicator that the
   SAS has not yet been checked. However, it it is well-known that when
   faced with optional security mechanisms, many users simply
   ignore them <xref target=3D"whitten-johnny"/>.

LMK if you still object.


> The whole paragraph is about SAS, but then adds "or fingerprints" in
> the last sentence.  There are very significant differences between an
> SAS and a fingerprint (which I will call a Way Too Long Authentication
> String).  If fingerprints are going to be discussed in this section,
> then the differences between an SAS and a WTLAS need to be explained.

Fair enough. I removed the graf.


ABOBA
-----
Bernard filed a bunch of issues:

Issue 4:
> It would be nice to see some editorial consistency among the document set=
.
> For example, the overview document uses the term "RTCWEB" to refer to the
> IETF WG or protocol specifications whereas this document uses "RTC-Web".

Changed to RTCWEB


Issue 5:
 > There also needs to be some mechanism for the browser to verify that
 >    the target of the traffic continues to wish to receive it.
 >    Obviously, some ICE-based mechanism will work here, but it has been
 >    observed that because ICE keepalives are indications, they will not
 >    work here, so some other mechanism is needed.
 >
 >    [[ OPEN ISSUE:  Do we need some way of verifying the expected traffic
 >    rate, not just consent to receive traffic at all.]]
 >
 > [BA] Since I believe we've figured this out, can we clean up the above
 > paragraph and OPEN ISSUE?

Agreed. Rewrote anda dded a link to draft-muthu.


>  [Note:  current thinking in the RTCWEB WG is not to support TCP and to
>  support SCTP over DTLS, thus removing the need for masking.]
>
>  [BA] This section seems somewhat "overtaken by events" given that the
data
>  channel will run over DTLS. How about the following?
>
>  4.2.2. Masking
>
>     Once consent is verified, there still is some concern about
>     misinterpretation attacks as described by Huang et al.[huang-w2sp].
>     Where TCP is used the risk is substantial due to the potential
>     presence of transparent proxies and therefore if TCP is to be used,
>     then WebSockets style masking MUST be employed.
>
>     Since DTLS (with the anti-chosen plaintext mechanisms required by
>     TLS 1.1) does not allow the attacker to generate predictable
>     ciphertext, there is no need for masking of protocols running over
>     DTLS (e.g. SCTP over DTLS, UDP over DTLS, etc.).

Done.



>     It is this consideration that makes an
>     automatic, public key-based key exchange mechanism imperative for
>     RTC-Web (this is a good idea for any communications security system)
>     and this mechanism SHOULD provide perfect forward secrecy (PFS).
>
>  [BA] Do we mean "SHOULD support" PFS or "SHOULD use"?  I don't believe
>  that DTLS/SRTP-EKT provides PFS.  Also, is there any implication that th=
e
>  user should be able to somehow influence whether PFS is required or not?

Changed per discussion in the meeting.


> 4.3.2. Protecting Against During-Call Attack
>
>     Protecting against attacks during a call is a more difficult
>     proposition.  Even if the calling service cannot directly access
>     keying material (as recommended in the previous section), it can
>     simply mount a man-in-the-middle attack on the connection, telling
>     Alice that she is calling Bob and Bob that he is calling Alice, while
>     in fact the calling service is acting as a calling bridge and
>     capturing all the traffic.  While in theory it is possible to
>     construct techniques which protect against this form of attack, in
>     practice these techniques all require far too much user intervention
>     to be practical, given the user interface constraints described in
>     [abarth-rtcweb].
>
>  [BA] I think it's more than a user intervention/user interface issue.
>  Aside from snooping the signaling to see if the callee includes an
>  "isfocus" tag, how can the browser know if it is calling a conference
>  bridge or not? Personally, I'd remove the "in theory" sentence.

I rewrote to describe how this could in principle happen with
a fingerprint or an IdP.


>  4.2.4. IP Location Privacy
>
>     Note that as soon as the callee sends their ICE candidates, the
>     caller learns the callee's IP addresses.  The callee's server
>     reflexive address reveals a lot of information about the callee's
>     location.  In order to avoid tracking, implementations may wish to
>     suppress the start of ICE negotiation until the callee has answered.
>     In addition, either side may wish to hide their location entirely by
>     forcing all traffic through a TURN server.
>
>  [BA] Might be useful to say explicitly that the concern about location
>  privacy is restricted to media; hiding the client's location from the We=
b
>  server is handled by things like ToR, and signaling privacy is
constrained
>  by the signaling protocol (e.g. SIP privacy, etc.).

I added some material here.



> #11: Security vs. Security-Arch
>
>  At various points within the Security Architecture document, I had a
>  feeling of deju vu, encountering material that was also in the Security
>  doc.  The overlap made me wonder if we should either move some material
>  from Arch to Security, or whether we couldn't live with a single doc
>  instead of two.

I agree it's a bit awkward. I'm taking my guidance on one versus two
documents from the chairs. If the WG wants me to merge, I'll do so.


>  Abstract
>
>  All but the last sentence of the abstract is identical to that of the
>  Security document.  Would it make some sense to shorten the abstract?
>  Also, references aren't allowed in abstracts.

Shortened it.


>  Section 1
>
>  The first paragraph duplicates material from the abstract as well as
>  Section 1 of the Security doc, and figure 1 is also included in both
docs.
>  However, the security doc doesn't discuss the multidomain case.  Would i=
t
>  make sense to move material from Section 1 of this document to the
>  security doc?  If this were done, would we still need Section 1 in this
>  doc?

As long as these are still separate documents, I'd like to keep this
one as-is. I tried to keep the security document high-level, so I think
this is probably the right split...



>  Section 5.2
>
>  This section, with its requirements on APIs, seemed similar in spirit to
>  Section 4.1 of the Security doc.

Agreed. The idea of Section 4.1 is to provide analysis and this to
provide nomative language. Agreed it's somewhat imperfect.


>  Section 5.4
>
>  Material in this section (such as the discussion of ToR) seems like it
>  might better belong in Section 4.2.4 of the Security doc.

I added some more high-level material into the security doc.



> #10: Additional Threats
>
>  Aside from the threats described in the document, a few others come to
>  mind.  It might be useful for the document to state explicitly why or wh=
y
>  not these are out of scope:
>
>  a. Live versus replayed streams.  While you might have media security,
>  this doesn't tell you whether the stream is actually originating from a
>  device or is being replayed.
>
>  b. Prank calling.  While the document talks about permission to make
>  calls, it doesn't mention permission to receive or blocking of unwanted
>  calls.

I added a section about malicious peers.



DRAFT-IETF-RTCWEB-SECURITY-ARCH
-------------------------------
> WESTERLUND
> I have made an individual review of the security architecture document
> version 6 and have the following comments.
>
> 1. In title and in other places, should we use WebRTC as the handle to
> the complete solution?

Done.


> 2. Section 7.2 is wrongly labeled, I assume it is -04 version

Unfortunately not.

"Version -04 was a version control mistake.  Please ignore."


> 3. Section 4:
> As is conventional in the Web, all identities are
>    ultimately rooted that system.
>
> I guess this should be ... rooted in that system?

Fixed.


> 4. Figure 4. The arrow between Bob's browser and Bob's IdP is missalinged=
.

Fixed.

> 5. Section 4.1:
> Because this is an audio/video call, it creates
>    two MediaStreams, one connected to an audio input and one connected
>    to a video input.
>
> I think this normally would be one mediaStream but with two
> MediaStreamTracks.

Agreed.


> 6. Section 4.1:
> If Bob agrees [I am ignoring early media for now], a PeerConnection
>    is instantiated with the message from Alice's side.
>
> I think you can remove the parenthesis. First of all I don't think it
> relavant in the paragraph. Secondly, early media doesn't exist in the
> context of a single peerConnection.

Removed.


> 7. Section 4.3
>
> If Alice and Bob authenticated via their IdPs,
>    then they also know that the signaling service is not mounting a man-
>    in-the-middle attack on theor traffic.
>
> theor / their?

Fixed.


> 8. Section 5.1:
>
> It is RECOMMENDED that browsers which allow
>    active mixed content nevertheless disable RTCWEB functionality in
>    mixed content settings. [[ OPEN ISSUE:  Should this be a 2119 MUST?
>    It's not clear what set of conditions would make this OK, other than
>    that browser manufacturers have traditionally been permissive here
>    here.]]
>
> I am actually quite worried about not making this a MUST. Mixed content
> clearly causes significant security holes that I think really should be
> avoided.

Agreed. Changed per-wg discussion.


> 9. Section 5.1, similarly I think this applies to the second open issue
> in this section.

Changed. If people disagree, please objec.t


> 10. Section 5.3:
>
> [Note:
>    this document takes no position on the split between ICE in JS and
>    ICE in the browser.  The above text is written the way it is for
>    editorial convenience and will be modified appropriately if the WG
>    decides on ICE in the JS.]  The JS MUST NOT be permitted to control
>    the local ufrag and password, though it of course knows it.
>
> I think you can clean up this statement as there is consensus to leave
> ICE in the browser.

Done.


> 11. Section 5.3:
> A separate document will profile the ICE
>    timers to be used; see [I-D.muthu-behave-consent-freshness].
>
> We should clarify the WG consensus on this document and the scope the
> document has.

I'm not sure I understand this. Is this a proposed change to the
document or a WG action item?


> 12. Section 5.4:
>
>    API Requirement:  The API MUST provide a mechanism for the calling
>       application JS to indicate that only TURN candidates are to be
>       used.  This prevents the peer from learning one's IP address at
>       all.
>
> What about the ICE candidates related address information, that should
> be mentioned I think also as possible risk of leaking and what to set it
to.

Good point!


> 13. I am missing a discussion on how to avoid the linkage issues. I
> think we need to point out the known sources to linkage that WebRTC
> adds. I am aware of RTCP CNAMEs and as well as the certificate used in
> DTLS(-SRTP). Their should be recommendations on how to avoid these issues=
.

See S 5.5 below. Are you looking for something else.




> 14. Section 5.5
>    [OPEN ISSUE:  What should the settings be here?  MUST?]
>    Implementations MAY support SDES for media traffic for backward
>    compatibility purposes.
>
> This needs to be resolved. I see no issues with retaining the
> possibility for SDES and other schemes. I think however, the downgrade
> issues may need more discussion here. This is a somewhat discussed in
> the security considerations section later.

I plan to hold off on this till after the discussion in Berlin.


> 15. Section 5.5
>    API Requirement:  The API MUST provide a mechanism to indicate that a
>       fresh DTLS key pair is to be generated for a specific call.  This
>       is intended to allow for unlinkability.  Note that there are also
>       settings where it is attractive to use the same keying material
>       repeatedly, especially those with key continuity-based
>       authentication.
>
> I think this has to do with 13. For example should really the same
> certificate be used with different origins, unless it is a intentionally
> added endpoint certificate that provide verifiable source?

No, it shouldn't be used with different origins except for thjat.


> 16. Section 5.5
> [largely derived from
>       [I-D.kaufman-rtcweb-security-ui]
>
> Maybe this is more suitable in a contributors section instead of inline
> in text.

Done.


> 17. Section 5.5:
>
>       *  The "security characteristics" MUST indicate the cryptographic
>          algorithms in use (For example:  "AES-CBC" or "Null Cipher".)
>
> Does there need to be additional discussion or clarifications on high
> level indications when one uses NULL cipher that doesn't provide
> confidentiality?

Good point. Added a new requirement here.


> 18. Section 5.6.2:
>
> The details of the mechanism are described in the W3C API
>    specification,
>
> I think a reference needs to be added here.
>
> 19. Section 5.6.3
> Todo REF!

Added.


> 20. Section 5.6.4.1:
>
>    The "algorithm" and digest values correspond directly to the
>    algorithm and digest in the a=3Dfingerprint line of the SDP.
>
> Should there be a reference to where a=3Dfingerprint is defined here?

Done.


> Also, does it need to be clarified what "algorithm" and digest
> reference, the ABNF contructs or value for those parameters.

I added "values" here. Is that what you were looking for?


> 21. Section 5.6.4.2:
> This SDP attribute is underspecified. No clear definition of what is
> allowed, no IANA consideration section registering it.

It's just a base64-encoded identity attribute.


> Also, should this SDP attribute also carry the origin of the IdP? I just
> wonder how a non browser can determine which IdP has been providing the
> assertion.

It's in the assertion. Do you think it would be better in the
a=3Dline?


> 22. Section 5.6.5.1 and following.
>
> I would like that all these JSON objects would have a bit more crisp
> definitions. Sure, the JSON has certain limiations in what values can be
> used, but it is unclear if the full UTF-8 values can be used in all
> cases, or if there is restrictions on the URL provided, or if it should
> be URI's in fact?

OK. I will circle around with Ted (as a URL expert) and anyone else
who feels expert to try to nail this down.

> 23. I am missing a security discussion around the fact that I can claim
> to have an IdP and load whatever JS code into the proxy. What security
> implications deos this have. Section 5.6.5.2 does provide some basic
> restrictions here. Are more needed?

I tried to hit this in 5.7.4. Generally if you are a site you can
always load new JS into the browser, so this is just part of the
threat model. Can you give me an example of what you're looking for.


> 24. Section 5.6.5.2.2:
>
>    idp:  A dictionary containing the domain name of the provider and the
>       protocol string
>
> What is the definition of dictionary here?

I'm not sure enough of the terminology here. Isn't dictionary
what we call JS associative-array type structures.



> 25. Are there any time limits on responding to SIGN or Verify, or can
> you get applications to hand simply by not responding to a IdP message?

I would hope the browser would produce an error at some point,
but I'm not sure we need to specify here.


> 26. Section 5.6.5.2.3:
>
> Are the fields required to be included or not. This is a bit unclear.

Currently it says:

"MUST contain a message field consisting of a dictionary/hash with the
following fields:"

Would you like it to say "which MUST contain the following fields"



> Section 27. Section 5.6.5.2.3:
> Open issue to resolve. Please try to resolve this issue by talking to
> some people.

Agreed. This is now on my TODO list.


> 27. Section 5.6.5.2.3.1
>
> Open issue needing resolution. No opinion.

Will raise this in Berlin.


> 28. Section 5.7.2:
>
> Missing the linkability issues discussion here.

Added some material.


> 29. Section 5.7.2:
>
> Combined RTCWEB/Tor
>    implementations SHOULD arrange to route the media as well as the
>    signaling through Tor. [Currently this will produce very suboptimal
>    performance.]

Done.


> I think you can remove the parenthis but keep the text here.
>
> 30. Page 40 SDP example looks strange with no a=3Didenity attribute,
> instead just the JSON object.
Fixed.



OHLSSON
> When reading draft-ietf-rtcweb-security-arch I noticed a couple of things
that might be missing:
>
> - According to section 4.3.2.4 in draft-ietf-rtcweb-security there shold
>   be a mechanism to verify that the remote browser has engaged a "secure
>   media mode". This mechanism is not (yet) described in the document.

Good catch. I have to write something here but I haven't yet.


> - Except for paragraph 6 in section 4.1, there is not much said about
>   the implications of the "secure media mode".

This is referring to isolated streams, which is now in gUM. How much
do you think I should say here?


> - An enterprise network with an HTTP proxy may not want external web
sites to
>   learn the IP addresses of its hosts using the PeerConnection API. I'm
>   not sure if this kind of attack is relevant or not. If it is then
>   it should be mentioned in Section 5.4. IP Location Privacy.

Added some text.
o
> I also have some minor comments on the rest of the document:
>
> - Section 3, 1st paragraph
>
>    The basic assumption of this architecture is that network resources
>    exist in a hierarchy of trust, rooted in the browser, which serves as
>    the user's TRUSTED COMPUTING BASE (TCB).
>
>   The term "hierarchy of trust" is unclear in this context. Are there
other
>   nodes in this hierarchy except for the browser and web server?

I would claim browser, web server, idp (if any), other network elements.



> - Section 3, 2nd paragraph
>
>     This is a natural extension of the end-to-end principle.
>
>   Perhaps it's just me but I don't understand what this end-to-end
principle is.

I just removed it since it doesn't add value.


> - Section 4.3, 1st paragraph
>
>     The total number of channels depends on the amount of
>     muxing; in the most likely case we are using both RTP/RTCP mux and
>     muxing multiple media streams on the same channel, in which case
>     there is only one DTLS handshake.
>
>    Won't there be separate handshake for the data channel?

My understanding was that we could mux the datachannel onto
the same 5-tuple.


> - Section 5.4, 1st paragraph
>
>     [...] which leaks large amounts of location information,
>     especially for mobile devices.
>
>    Why are mobile devices different in this aspect?

They're not. Removed.


> - Section 5.6.4
>
>   The SDP example contains RTP/AVP but I thought SRTP was made mandatory?

Bit rot. Fixed.


> - Section 5.6.5.1, 3rd paragraph
>
>    All requests from the PeerConnection object MUST contain an "id"
>    field which MUST be unique for that PeerConnection object.  Any
>    responses from the IdP proxy MUST contain the same id in response,
>    which allows the PeerConnection to correlate requests and responses.
>
>   Couldn't the browser implementation handle the message routing
>   without the id field?

This isn't about routing, it's about what happens if there are
multiple outstanding queries.


> - Section 5.6.5.1.1
>
>   The mandatory id field is missing in the example

Fixed.


> - Section 5.6.5.2.2
>
>   How is the assertion field transformed into the a=3Didentity attribute?
>   I guess you=B4re using some form of B64 encoding but this is not expain=
ed
>   anywhere.

It's earlier, but I added it.


> - Section 5.6.5.2.3
>
>   I think the reason for including request_origin should be explained her=
e
>   instead of in the end of the document.

I added a forward reference instead to avoid breaking up the document
flow.


> - Section 5.7.1, 3rd paragraph
>
>   WebCrypto API lacks a reference.
>
> - Section 5.7.4.5.2
>
>     Any IdP which uses cookies to persist logins will be broken
>     by third-party cookie blocking.
>
>   Shouldn't it say "Any IdP which uses third-party cookies"?

IdPs are inherently third-party in this context.


> - Appendix A and B
>
>   Both appendices appear unfinished. For example, ROAP is still mentioned
>   in othe BrowserID example and it's unclear what client credentials
>   Bob uses in the OAuth example.

Agreed. I have a TODO to clean these up.




JOHNSTON
--------
> Again, lots of RTCWEB, RTCWeb, etc that should be WebRTC.

Fixed.


o> The document introduction seems to be specific to audio and video
  WebRTC sessions.  Later, it becomes clear that the analysis applies
  to the data channel as well.  This should be stated clearly up front
  and any differences between the two should be explained (i.e. DTLS
  vs SRTP encryption).

I added a sentence in S 4.3. This is actually the first significant
discussion of SRTP, so I think this works best.



> "SIP or XMPP" - XMPP isn't a signaling protocol but Jingle is, so that is
probably what should be mentioned, but Jingle doesn't use SDP.

I just left it as SIP.


> "Web sites whose origin we can verify (optimally via HTTPS, but in some
cases because we are on a topologically restricted network, such as behind
a firewall)"  - what is the 2nd case - no verification?  Verification using
something other than HTTPS?

I added "and can infer authentication from firewall behavior"

> 3.1  middle para graph is basically saying that security policy can be
applied after authentication - might be worth stating this after the Dr.
Evil analogy.

Added some text.

> 4. "Specifically, Alice and Bob have relationships with
>    some Identity Provider (IdP) that supports a protocol such as OpenID
>    or BrowserID) that can be used to demonstrate their identity to other
>    parties." - needs one more ( to parse correctly.
>
> Figures 3 & 4.  Would be better to have IdP1 and IdP2 to make clear that
this can be 2 different providers. Also, Secure WebSockets should be shown
with HTTPS.

Done.


> In Figures 3 & 4, media is shown as DTLS-SRTP when the media is in fact
SRTP which is keyed with DTLS-SRTP.  In fact, RFC 3711 is not even
referenced, when it should be a normative reference, which is potentially
very confusing and misleading for implementors.  There are many places
where DTLS-SRTP described as if it is more than a key agreement protocol
for SRTP.

I changed this to be "DTLS+SRTP", added a reference to RFC 3711, and
went through each use of DTLS-SRTP for appropriateness, modifying
where I thought necessary. Please let me know if there are places I
missed.


> Figure 4 is the Trapezoid of the overview and should be referenced as
such.

Added a back reference to figure 2.

> JS is used but not defined.

Defined it back in the intro.


> 4.1  The first example in this document of microphone and webcam
permissions is that of a permanent grant.  Is this the model we want to
encourage?  Shouldn't the first mention be of the much safer per call
grant?  Later the per call grant is mentioned, but deep in the document.

Rewritten.


> 4.1 "This message is sent to the signaling server, e.g., by
XMLHttpRequest[XmlHttpRequest] or by WebSockets [RFC6455] "  Sentence is
missing a period.  Do we really mean HTTPS and Secure WebSockets here?

I added "preferably over TLS".


> 4.1 "This allows the browser to display a trusted element in the browser
chrome indicating that a call is coming in from Alice."  Can this assertion
be made now, prior to the DTLS-SRTP handshake completing?  If this identity
assertion is made at this stage, then a different fingerprint shows up in
the DTLS-SRTP handshake, does the browser chrome then indicate 'sorry, I
made a mistake, it isn't Alice calling'?

I think at that point you just show a network error. The identity
assertion indicates that Alice was trying to call and if someone
is attacking, then that's just a failure.


> There are many cases of using [] instead of () for parentheical text.

I try to use [] for Notes or open issues.


> 4.2 ICE is not defined or referenced in its first occurance.

Fixed in 4.1.

> typo "theor"
>
> 5.1 "[[ OPEN ISSUE:  Should this be a 2119 MUST?
>    It's not clear what set of conditions would make this OK, other than
>    that browser manufacturers have traditionally been permissive here
>    here.]] "
>
>    [[ OPEN ISSUE::  Should we be more aggressive about this?]]
>
> Do these issues need to be discussed in a W3C document where browser
expertise might help us do the right thing?

I think we resolved this in Orlando and the document reflects what I think
the consensus was.


> 5.2  Do we really expect browsers to have X.509 certificates?  Does
anyone do this with DTLS-SRTP today?  Do browsers have UI to manage these
certs?

No, we don't. This was a request from Martin because he was expecting
servers might. I added some text to try to make that clear.



> 5.2 The API and UI requirements and elsewhere - should these be in a W3C
document instead of/in addition to this document?

My sense is that IETF was responsible for the security analysis so they
belong here.


> 5.2 "Implementations which support some form of direct user authenticatio=
n
>    SHOULD also provide a policy by which a user can authorize calls only
>    to specific counterparties."  What is a counterparty?

Changed to "communicating peers."


> 5.3   ICE-Lite is not defined or referenced.  Since this is a MUST, we
need a normative reference - is it what is described as ICE Lite in RFC
5245 or draft-rescorla-mmusic-ice-lite?

It's 5245 again. I added a cite.


> Question:  If a browser has multiple public/private keys, including an
X.509 one, can the JS suggest which one to use for a particular
PeerConnection?

I think the answer needs to be yes here, but we probably need to clean
up the W3C API a bit to match.


> 5.5 Again, the media channel is secured using SRTP, not DTLS-SRTP,
> which is one way to key SRTP.  In the future, new and better key
> agreements will be developed, and if we write all our specs assuming
> one particular keying method, then they will become obsolete.

Good point. Here is my new text which I hope is clearer:

   Implementations MUST implement SRTP [RFC3711].  Implementations MUST
   implement DTLS [RFC4347] and DTLS-SRTP [RFC5763][RFC5764] for SRTP
   keying.  Implementations MUST implement
   [I-D.ietf-tsvwg-sctp-dtls-encaps].

   All media channels MUST be secured via SRTP.  Media traffic MUST NOT
   be sent over plain (unencrypted) RTP.  DTLS-SRTP MUST be offered for
   every media channel and MUST be the default; i.e., if an
   implementation receives an offer for DTLS-SRTP and SDES, DTLS-SRTP
   MUST be selected.

   All data channels MUST be secured via DTLS.


> Section 5.6 with Appendix A reads like a completely separate
> document.  It is also very hard reading as there are lots of new
> concepts and ideas.  Should perhaps this be in a separate document?
> The basic security architecture for WebRTC is unlikely to change,
> but given that we have no experience with this IdP system, it seems
> likely that it will evolve and perhaps change significantly, which
> is an argument for a separate document.

I agree the writing is a bit awkward. I'm rewriting App A to make it
clearer and I'll see what I can do to about making S 5.6 merge better
as well.


> 5.6.4.2  Is this document defining a new SDP attribute?  Shouldn't this
be done in MMUSIC?

I have a TODO here. Will consult with the chairs.


> typos "implemementations"  "termnating" "restrcitions"

Fixed.


> WebIntents mentioned without reference or explanation.

WebIntents is OBE. Removed.


> Missing normative reference: RFC 3711.

Fixed.


-Ekr

--047d7b6da78246f54a04e18ce9ab
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I&#39;ve just posted updated security documents reflecting=
 most, but not all<div>the last call comments. There&#39;s still a few TODO=
s I need to resolve</div><div>F2F, (+ the SDES thing which I haven&#39;t do=
ne anything with, since we&#39;re</div>

<div>discussing it in Berlin).</div><div><br></div><div>Below is a consolid=
ated response to people&#39;s comments. Please let me</div><div>know if you=
 think I haven&#39;t responded adequately to yours.</div><div><br></div>

<div><br></div><div><div>DRAFT-IETF-RTCWEB-SECURITY</div><div>-------------=
-------------</div><div>=A0WESTERLUND</div><div>&gt; I have reviewed the se=
curity draft (-04) as an individual and have the</div><div>&gt; following c=
omments.</div>

<div><br></div><div>Thanks for your comments. Responses below.</div><div><b=
r></div><div><br></div><div>&gt; 1. Usage of acronym RTCWEB vs WebRTC. I th=
ought we earlier had a</div><div>&gt; conclusion that we would use WebRTC a=
s the acronym for the complete</div>

<div>&gt; solution? Am I remembering incorrectly?</div><div><br></div><div>=
Changed, except when referring to the WG.</div><div><br></div><div><br></di=
v><div>&gt; 2. I think the introduction should contain some pointer to the =
overview</div>

<div>&gt; draft for reverse lookup purpose if one stumbles on the document.=
</div><div><br></div><div>Done.</div><div><br></div><div><br></div><div>&gt=
; 3. Section 3.1:</div><div>&gt; Similarly, while Flash SWFs can access the=
 camera and microphone,</div>

<div>&gt; =A0 =A0they explicitly require that the user consent to that acce=
ss.</div><div>&gt;=A0</div><div>&gt; Can you please expand the SWF acronym =
and perhaps add a reference?</div><div><br></div><div>The acronym isn&#39;t=
 very useful, since I think it refers ot</div>

<div>&quot;Shockwave Flash&quot;. I changed this to programs (&quot;SWFs&qu=
ot;)</div><div><br></div><div>Added a reference.</div><div><br></div><div><=
br></div><div>&gt; 4. Section 3.2:</div><div>&gt; Many other resources are =
accessible but isolated. =A0For instance,</div>

<div>&gt; =A0 =A0while scripts are allowed to make HTTP requests via the</d=
iv><div>&gt; =A0 =A0XMLHttpRequest() API those requests are not allowed to =
be made to any</div><div>&gt; =A0 =A0server, but rather solely to the same =
ORIGIN from whence the script</div>

<div>&gt; =A0 =A0came.[RFC6454] (although CORS [CORS] and WebSockets [RFC64=
55]</div><div>&gt; =A0 =A0provides a escape hatch from this restriction, as=
 described below.)</div><div>&gt;=A0</div><div>&gt; This above looks strang=
e around the first period, which is followed by</div>

<div>&gt; RFC6454 reference. If is is a new sentence, the main sentence con=
tains</div><div>&gt; only a reference.</div><div><br></div><div>Fixed.</div=
><div><br></div><div><br></div><div>&gt; 5. Section 3.2</div><div>&gt; This=
 SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks</div>

<div>&gt; =A0 =A0on server B via the user&#39;s browser, which protects bot=
h the user</div><div>&gt; =A0 =A0(e.g., from misuse of his credentials) and=
 the server (e.g., from DoS</div><div>&gt; =A0 =A0attack).</div><div>&gt;=
=A0</div><div>

&gt; I think one can make clear that the server one protects are B in the e=
nd</div><div>&gt; of the sentence. Although it reasonably clear from contex=
t, I think</div><div>&gt; writing it like this:</div><div>&gt;=A0</div><div=
>

&gt; This SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks<=
/div><div>&gt; =A0 =A0on server B via the user&#39;s browser, which protect=
s both the user</div><div>&gt; =A0 =A0(e.g., from misuse of his credentials=
) and the server B (e.g., from</div>

<div>&gt; =A0 =A0DoS attack).</div><div>&gt;=A0</div><div>&gt; Is clearer.<=
/div><div><br></div><div>Changed.</div><div><br></div><div><br></div><div>&=
gt; 6. Section <a href=3D"http://4.1.1.3">4.1.1.3</a>:</div><div>&gt; In bo=
th of the previous cases, the user has a direct relationship</div>

<div>&gt; =A0 =A0(though perhaps a transient one) with the target of the ca=
ll.</div><div>&gt;=A0</div><div>&gt; Is it really the &quot;target of the c=
all&quot; or is it the calling service that</div><div>&gt; the user has a d=
irect relation ship with?</div>

<div><br></div><div>I removed this section entirely since we decided to do =
nothing</div><div>about this and I have seen other people be confused by it=
.</div><div><br></div><div><br></div><div>&gt; 7. Section 4.2.1:</div>
<div>
&gt;=A0</div><div>&gt; [[ OPEN ISSUE: =A0Do we need some way of verifying t=
he expected traffic</div><div>&gt; =A0 =A0rate, not just consent to receive=
 traffic at all.]]</div><div>&gt;=A0</div><div>&gt; Regarding this issue, I=
 think you could add something which makes this</div>

<div>&gt; not required, namely that by requiring congestion control of all =
flows</div><div>&gt; established by the browser, the browser tries to avoid=
 persistent</div><div>&gt; congestion and any overload attack is less effec=
tive. The attack is more</div>

<div>&gt; in the vain of reducing any competing flows supported rate to a l=
ower</div><div>&gt; fair share rate, which in worst case is below what is r=
equired to</div><div>&gt; support the service being the target of the attac=
k.</div>

<div>&gt;=A0</div><div>&gt; Thus I think the open issue question is a nice =
to have, but not required</div><div>&gt; feature.</div><div><br></div><div>=
Agreed, and I think we have mostly come to consensus on this point,</div>

<div>so I have removed the open issue.</div><div><br></div><div><br></div><=
div>&gt; 8. Section 4.2.4:</div><div>&gt;=A0</div><div>&gt; I think this do=
cument is missing to discuss the privacy threats beyond</div><div>&gt; IP l=
ocation privacy? I think that should be added, especially as we are</div>

<div>&gt; addressing some issues we know of.</div><div>&gt;=A0</div><div>&g=
t; These other threats that we have discussed are associated with anonymous=
</div><div>&gt; calling services and linking and fingerprinting browsers or=
 users with</div>

<div>&gt; anonymous calls. The threats we do know exist are for example RTC=
P CNAME</div><div>&gt; generation, DTLS certifcates, and API fingerprinting=
. I think these</div><div>&gt; needs to be raised as a threat.</div><div>

<br></div><div>Added a privacy considerations section.</div><div><br></div>=
<div>-Ekr</div><div><br></div><div>----------------------------------------=
---------</div><div>OHLSSON</div><div>&gt; Some minor comments on draft-iet=
f-rtcweb-security:</div>

<div><br></div><div>Thanks. Responses below.</div><div><br></div><div><br><=
/div><div>&gt; - Abstract</div><div>&gt;=A0</div><div>&gt; =A0 =A0 This doc=
ument defines the RTC-Web threat model and defines an</div><div>&gt; =A0 =
=A0 architecture which provides security within that threat model</div>

<div>&gt;=A0</div><div>&gt; =A0 Isn&#39;t the architecture defined in draft=
-ietf-rtcweb-security-arch?</div><div><br></div><div>Good catch. Fixed.</di=
v><div><br></div><div><br></div><div>&gt; - Section 4.1.1.3, 3rd paragraph<=
/div>

<div>&gt;=A0</div><div>&gt; =A0 =A0 However, this obviously presents a priv=
acy challenge, as sites which</div><div>&gt; =A0 =A0 host advertisements in=
 IFRAMEs often learn very little about whether</div><div>&gt; =A0 =A0 indiv=
idual users clicked through to the ads, or even which ads were</div>

<div>&gt; =A0 =A0 presented.</div><div>&gt;=A0</div><div>&gt; =A0 What exac=
tly is the privacy issue here? Is it that the hosting site</div><div>&gt; =
=A0 can eavesdrop on the call between the user and the advertiser since</di=
v><div>

&gt; =A0 IFRAMEs are not used?</div><div><br></div><div>I removed this sect=
ion.</div><div><br></div><div><br></div><div>&gt; - Section 4.3.1, 3rd para=
graph</div><div>&gt;=A0</div><div>&gt; =A0 =A0 In addition, the system MUST=
 NOT provide any APIs to extract either</div>

<div>&gt; =A0 =A0 long-term keying material or to directly access any store=
d traffic keys.</div><div>&gt;=A0</div><div>&gt; =A0 Does this mean that SD=
ES is out of the picture or does the sentence only</div><div>&gt; =A0 apply=
 to DTLS(-SRTP)?</div>

<div><br></div><div>I don&#39;t think this really impacts SDES, in context,=
 but I tried to add</div><div>some clarifying text.</div><div><br></div><di=
v><br></div><div><br></div><div>JOHNSON</div><div>&gt; Abstract and the dup=
licate text in Introduction: RTC-web turn into</div>

<div>&gt; either RTCWEB or WebRTC, with WebRTC being preferred.</div><div>&=
gt;=A0</div><div>&gt; &quot;some Web server&quot; - slangy &quot;a web serv=
er&quot;</div><div><br></div><div>Changed.</div><div><br></div><div><br>
</div>
<div>&gt; Figure 1: might also want to show WebSockets as this is becoming =
very</div><div>&gt; common and is mentioned later in the document without m=
uch context.</div><div><br></div><div>I added WebSockets.</div><div><br>

</div><div><br></div><div>&gt; Also, do we want to show the signaling chann=
el (the one referenced by</div><div>&gt; the W3C WebRTC spec that transport=
s the SDP offers and answers between</div><div>&gt; browsers)?</div><div>

<br></div><div>I think it clutters the diagram, but I added some text.</div=
><div><br></div><div><br></div><div>&gt; JS, popunder, and SWF are not defi=
ned in the draft</div><div><br></div><div>Defined JS and SWF and rewrote to=
 not use popunder.</div>

<div><br></div><div><br></div><div>&gt; 4.1.2. has a reference to draft-iet=
f-rescorla-rtcweb-generic-idp.</div><div>&gt; What is the status of this dr=
aft?</div><div><br></div><div>Per WG discussion it has been folded into the=
 security architecture.</div>

<div>Fixed.</div><div><br></div><div><br></div><div>&gt; 4.3. =A0RFC 3711 S=
RTP is not mentioned in the opening paragraph, which</div><div>&gt; seems s=
trange, since it is the actual protocol providing the COMSEC</div><div>&gt;=
 properties mentioned, rather then DTLS or DTLS-SRTP for media.</div>

<div><br></div><div>Agreed. Fixed.</div><div><br></div><div><br></div><div>=
&gt; 4.3.1. Might be useful to mention that this is sometimes called a</div=
><div>&gt; &quot;passive attack&quot; whereas 4.3.2 is an &quot;active atta=
ck&quot;.</div>

<div><br></div><div>Fixed.</div><div><br></div><div>&gt; 4.3.2.2.There are =
a number of misleading statements about SAS in this</div><div>&gt; section =
which should be corrected. =A0For example:</div><div>&gt;=A0</div><div>&gt;=
 =A0&quot;This SAS is designed to be read over the voice channel and if</di=
v>

<div>&gt; =A0confirmed by both sides precludes MITM attack.&quot; =A0The SA=
S is to be</div><div>&gt; =A0*compared* by the users - reading it out loud =
is only one possible</div><div>&gt; =A0mechanism.</div><div><br></div><div>=
Rewritten as:</div>

<div><br></div><div>=A0 ZRTP &lt;xref target=3D&quot;RFC6189&quot;/&gt; use=
s a &quot;short authentication string&quot; (SAS) which is derived</div><di=
v>=A0 from the key agreement protocol. This SAS is designed to be compared<=
/div>

<div>=A0 by the users (e.g., read aloud over the the voice channel or</div>=
<div>=A0 transmitted via an out of band channel) and if confirmed by both s=
ides precludes MITM</div><div>=A0 attack. The intention is that the SAS is =
used once and then key</div>

<div>=A0 continuity (though a different mechanism from that discussed</div>=
<div>=A0 above) is used thereafter.=A0</div><div><br></div><div><br></div><=
div><br></div><div>&gt; &quot;Moreover, it is possible for an attacker who =
controls the browser to</div>

<div>&gt; allow the SAS to succeed and then simulate call failure and recon=
nect,</div><div>&gt; trusting that the user will not notice that the &quot;=
no SAS&quot; indicator has</div><div>&gt; been set (which seems likely).&qu=
ot; - I don&#39;t quite know what this means.</div>

<div>&gt; If SAS is used, the UI is the browser chrome, so if this is possi=
ble,</div><div>&gt; it is just a badly designed UI not a protocol failure o=
r issue.</div><div><br></div><div>Removed.</div><div><br></div><div><br>

</div><div>&gt; &quot;Even were SAS secure if used, it seems exceedingly un=
likely that</div><div>&gt; users will actually use it.&quot; - This reads l=
ike speculation. =A0There are</div><div>&gt; users today using SAS, open so=
urce users and commercial users. =A0One</div>

<div>&gt; thing is certain, if we don&#39;t provide SAS, then users will no=
t use it.</div><div><br></div><div>Yes, I think this is fair. I rewrote thi=
s to be a bit more neutral.</div><div><br></div><div>=A0 =A0Additionally, i=
t is unclear that users will actually use an SAS.</div>

<div>=A0 =A0As discussed above, the browser UI constraints preclude requiri=
ng</div><div>=A0 =A0the SAS exchange prior to completing the call and so it=
 must be</div><div>=A0 =A0voluntary; at most the browser will provide some =
UI indicator that the</div>

<div>=A0 =A0SAS has not yet been checked. However, it it is well-known that=
 when</div><div>=A0 =A0faced with optional security mechanisms, many users =
simply</div><div>=A0 =A0ignore them &lt;xref target=3D&quot;whitten-johnny&=
quot;/&gt;.</div>

<div><br></div><div>LMK if you still object.=A0</div><div><br></div><div><b=
r></div><div>&gt; The whole paragraph is about SAS, but then adds &quot;or =
fingerprints&quot; in</div><div>&gt; the last sentence. =A0There are very s=
ignificant differences between an</div>

<div>&gt; SAS and a fingerprint (which I will call a Way Too Long Authentic=
ation</div><div>&gt; String). =A0If fingerprints are going to be discussed =
in this section,</div><div>&gt; then the differences between an SAS and a W=
TLAS need to be explained.</div>

<div><br></div><div>Fair enough. I removed the graf.</div><div><br></div><d=
iv><br></div><div>ABOBA</div><div>-----</div><div>Bernard filed a bunch of =
issues:</div><div><br></div><div>Issue 4:</div><div>&gt; It would be nice t=
o see some editorial consistency among the document set.</div>

<div>&gt; For example, the overview document uses the term &quot;RTCWEB&quo=
t; to refer to the</div><div>&gt; IETF WG or protocol specifications wherea=
s this document uses &quot;RTC-Web&quot;.</div><div><br></div><div>Changed =
to RTCWEB</div>

<div><br></div><div><br></div><div>Issue 5:</div><div>=A0&gt; There also ne=
eds to be some mechanism for the browser to verify that</div><div>=A0&gt; =
=A0 =A0the target of the traffic continues to wish to receive it.</div><div=
>=A0&gt; =A0 =A0Obviously, some ICE-based mechanism will work here, but it =
has been</div>

<div>=A0&gt; =A0 =A0observed that because ICE keepalives are indications, t=
hey will not</div><div>=A0&gt; =A0 =A0work here, so some other mechanism is=
 needed.</div><div>=A0&gt;=A0</div><div>=A0&gt; =A0 =A0[[ OPEN ISSUE: =A0Do=
 we need some way of verifying the expected traffic</div>

<div>=A0&gt; =A0 =A0rate, not just consent to receive traffic at all.]]</di=
v><div>=A0&gt;=A0</div><div>=A0&gt; [BA] Since I believe we&#39;ve figured =
this out, can we clean up the above</div><div>=A0&gt; paragraph and OPEN IS=
SUE?</div>

<div><br></div><div>Agreed. Rewrote anda dded a link to draft-muthu.</div><=
div><br></div><div><br></div><div>&gt; =A0[Note: =A0current thinking in the=
 RTCWEB WG is not to support TCP and to</div><div>&gt; =A0support SCTP over=
 DTLS, thus removing the need for masking.]</div>

<div>&gt;=A0</div><div>&gt; =A0[BA] This section seems somewhat &quot;overt=
aken by events&quot; given that the data</div><div>&gt; =A0channel will run=
 over DTLS. How about the following?</div><div>&gt;=A0</div><div>&gt; =A04.=
2.2. Masking</div>

<div>&gt;=A0</div><div>&gt; =A0 =A0 Once consent is verified, there still i=
s some concern about</div><div>&gt; =A0 =A0 misinterpretation attacks as de=
scribed by Huang et al.[huang-w2sp].</div><div>&gt; =A0 =A0 Where TCP is us=
ed the risk is substantial due to the potential</div>

<div>&gt; =A0 =A0 presence of transparent proxies and therefore if TCP is t=
o be used,</div><div>&gt; =A0 =A0 then WebSockets style masking MUST be emp=
loyed.</div><div>&gt;=A0</div><div>&gt; =A0 =A0 Since DTLS (with the anti-c=
hosen plaintext mechanisms required by</div>

<div>&gt; =A0 =A0 TLS 1.1) does not allow the attacker to generate predicta=
ble</div><div>&gt; =A0 =A0 ciphertext, there is no need for masking of prot=
ocols running over</div><div>&gt; =A0 =A0 DTLS (e.g. SCTP over DTLS, UDP ov=
er DTLS, etc.).</div>

<div><br></div><div>Done.</div><div><br></div><div><br></div><div><br></div=
><div>&gt; =A0 =A0 It is this consideration that makes an</div><div>&gt; =
=A0 =A0 automatic, public key-based key exchange mechanism imperative for</=
div><div>

&gt; =A0 =A0 RTC-Web (this is a good idea for any communications security s=
ystem)</div><div>&gt; =A0 =A0 and this mechanism SHOULD provide perfect for=
ward secrecy (PFS).</div><div>&gt;=A0</div><div>&gt; =A0[BA] Do we mean &qu=
ot;SHOULD support&quot; PFS or &quot;SHOULD use&quot;? =A0I don&#39;t belie=
ve</div>

<div>&gt; =A0that DTLS/SRTP-EKT provides PFS. =A0Also, is there any implica=
tion that the</div><div>&gt; =A0user should be able to somehow influence wh=
ether PFS is required or not?</div><div><br></div><div>Changed per discussi=
on in the meeting.</div>

<div><br></div><div><br></div><div>&gt; 4.3.2. Protecting Against During-Ca=
ll Attack</div><div>&gt;=A0</div><div>&gt; =A0 =A0 Protecting against attac=
ks during a call is a more difficult</div><div>&gt; =A0 =A0 proposition. =
=A0Even if the calling service cannot directly access</div>

<div>&gt; =A0 =A0 keying material (as recommended in the previous section),=
 it can</div><div>&gt; =A0 =A0 simply mount a man-in-the-middle attack on t=
he connection, telling</div><div>&gt; =A0 =A0 Alice that she is calling Bob=
 and Bob that he is calling Alice, while</div>

<div>&gt; =A0 =A0 in fact the calling service is acting as a calling bridge=
 and</div><div>&gt; =A0 =A0 capturing all the traffic. =A0While in theory i=
t is possible to</div><div>&gt; =A0 =A0 construct techniques which protect =
against this form of attack, in</div>

<div>&gt; =A0 =A0 practice these techniques all require far too much user i=
ntervention</div><div>&gt; =A0 =A0 to be practical, given the user interfac=
e constraints described in</div><div>&gt; =A0 =A0 [abarth-rtcweb].</div><di=
v>&gt;=A0</div>

<div>&gt; =A0[BA] I think it&#39;s more than a user intervention/user inter=
face issue.</div><div>&gt; =A0Aside from snooping the signaling to see if t=
he callee includes an</div><div>&gt; =A0&quot;isfocus&quot; tag, how can th=
e browser know if it is calling a conference</div>

<div>&gt; =A0bridge or not? Personally, I&#39;d remove the &quot;in theory&=
quot; sentence.</div><div><br></div><div>I rewrote to describe how this cou=
ld in principle happen with</div><div>a fingerprint or an IdP.</div><div>

<br></div><div><br></div><div>&gt; =A04.2.4. IP Location Privacy</div><div>=
&gt;=A0</div><div>&gt; =A0 =A0 Note that as soon as the callee sends their =
ICE candidates, the</div><div>&gt; =A0 =A0 caller learns the callee&#39;s I=
P addresses. =A0The callee&#39;s server</div>

<div>&gt; =A0 =A0 reflexive address reveals a lot of information about the =
callee&#39;s</div><div>&gt; =A0 =A0 location. =A0In order to avoid tracking=
, implementations may wish to</div><div>&gt; =A0 =A0 suppress the start of =
ICE negotiation until the callee has answered.</div>

<div>&gt; =A0 =A0 In addition, either side may wish to hide their location =
entirely by</div><div>&gt; =A0 =A0 forcing all traffic through a TURN serve=
r.</div><div>&gt;=A0</div><div>&gt; =A0[BA] Might be useful to say explicit=
ly that the concern about location</div>

<div>&gt; =A0privacy is restricted to media; hiding the client&#39;s locati=
on from the Web</div><div>&gt; =A0server is handled by things like ToR, and=
 signaling privacy is constrained</div><div>&gt; =A0by the signaling protoc=
ol (e.g. SIP privacy, etc.).</div>

<div><br></div><div>I added some material here.</div><div><br></div><div><b=
r></div><div><br></div><div>&gt; #11: Security vs. Security-Arch</div><div>=
&gt;=A0</div><div>&gt; =A0At various points within the Security Architectur=
e document, I had a</div>

<div>&gt; =A0feeling of deju vu, encountering material that was also in the=
 Security</div><div>&gt; =A0doc. =A0The overlap made me wonder if we should=
 either move some material</div><div>&gt; =A0from Arch to Security, or whet=
her we couldn&#39;t live with a single doc</div>

<div>&gt; =A0instead of two.</div><div><br></div><div>I agree it&#39;s a bi=
t awkward. I&#39;m taking my guidance on one versus two</div><div>documents=
 from the chairs. If the WG wants me to merge, I&#39;ll do so.</div><div>

<br></div><div><br></div><div>&gt; =A0Abstract</div><div>&gt;=A0</div><div>=
&gt; =A0All but the last sentence of the abstract is identical to that of t=
he</div><div>&gt; =A0Security document. =A0Would it make some sense to shor=
ten the abstract?</div>

<div>&gt; =A0Also, references aren&#39;t allowed in abstracts.</div><div><b=
r></div><div>Shortened it.</div><div><br></div><div><br></div><div>&gt; =A0=
Section 1</div><div>&gt;=A0</div><div>&gt; =A0The first paragraph duplicate=
s material from the abstract as well as</div>

<div>&gt; =A0Section 1 of the Security doc, and figure 1 is also included i=
n both docs.</div><div>&gt; =A0However, the security doc doesn&#39;t discus=
s the multidomain case. =A0Would it</div><div>&gt; =A0make sense to move ma=
terial from Section 1 of this document to the</div>

<div>&gt; =A0security doc? =A0If this were done, would we still need Sectio=
n 1 in this</div><div>&gt; =A0doc?</div><div><br></div><div>As long as thes=
e are still separate documents, I&#39;d like to keep this</div><div>one as-=
is. I tried to keep the security document high-level, so I think</div>

<div>this is probably the right split...</div><div><br></div><div><br></div=
><div><br></div><div>&gt; =A0Section 5.2</div><div>&gt;=A0</div><div>&gt; =
=A0This section, with its requirements on APIs, seemed similar in spirit to=
</div>

<div>&gt; =A0Section 4.1 of the Security doc.</div><div><br></div><div>Agre=
ed. The idea of Section 4.1 is to provide analysis and this to</div><div>pr=
ovide nomative language. Agreed it&#39;s somewhat imperfect.</div><div><br>

</div><div><br></div><div>&gt; =A0Section 5.4</div><div>&gt;=A0</div><div>&=
gt; =A0Material in this section (such as the discussion of ToR) seems like =
it</div><div>&gt; =A0might better belong in Section 4.2.4 of the Security d=
oc.</div>

<div><br></div><div>I added some more high-level material into the security=
 doc.</div><div><br></div><div><br></div><div><br></div><div>&gt; #10: Addi=
tional Threats</div><div>&gt;=A0</div><div>&gt; =A0Aside from the threats d=
escribed in the document, a few others come to</div>

<div>&gt; =A0mind. =A0It might be useful for the document to state explicit=
ly why or why</div><div>&gt; =A0not these are out of scope:</div><div>&gt;=
=A0</div><div>&gt; =A0a. Live versus replayed streams. =A0While you might h=
ave media security,</div>

<div>&gt; =A0this doesn&#39;t tell you whether the stream is actually origi=
nating from a</div><div>&gt; =A0device or is being replayed.</div><div>&gt;=
=A0</div><div>&gt; =A0b. Prank calling. =A0While the document talks about p=
ermission to make</div>

<div>&gt; =A0calls, it doesn&#39;t mention permission to receive or blockin=
g of unwanted</div><div>&gt; =A0calls.</div><div><br></div><div>I added a s=
ection about malicious peers.</div><div><br></div><div><br></div><div><br>
</div>
<div>DRAFT-IETF-RTCWEB-SECURITY-ARCH</div><div>----------------------------=
---</div><div>&gt; WESTERLUND</div><div>&gt; I have made an individual revi=
ew of the security architecture document</div><div>&gt; version 6 and have =
the following comments.</div>

<div>&gt;=A0</div><div>&gt; 1. In title and in other places, should we use =
WebRTC as the handle to</div><div>&gt; the complete solution?</div><div><br=
></div><div>Done.</div><div><br></div><div><br></div><div>&gt; 2. Section 7=
.2 is wrongly labeled, I assume it is -04 version</div>

<div><br></div><div>Unfortunately not.</div><div><br></div><div>&quot;Versi=
on -04 was a version control mistake. =A0Please ignore.&quot;</div><div><br=
></div><div><br></div><div>&gt; 3. Section 4:</div><div>&gt; As is conventi=
onal in the Web, all identities are</div>

<div>&gt; =A0 =A0ultimately rooted that system.</div><div>&gt;</div><div>&g=
t; I guess this should be ... rooted in that system?</div><div><br></div><d=
iv>Fixed.</div><div><br></div><div><br></div><div>&gt; 4. Figure 4. The arr=
ow between Bob&#39;s browser and Bob&#39;s IdP is missalinged.</div>

<div><br></div><div>Fixed.</div><div><br></div><div>&gt; 5. Section 4.1:</d=
iv><div>&gt; Because this is an audio/video call, it creates</div><div>&gt;=
 =A0 =A0two MediaStreams, one connected to an audio input and one connected=
</div>

<div>&gt; =A0 =A0to a video input.</div><div>&gt;=A0</div><div>&gt; I think=
 this normally would be one mediaStream but with two</div><div>&gt; MediaSt=
reamTracks.</div><div><br></div><div>Agreed.</div><div><br></div><div><br><=
/div>

<div>&gt; 6. Section 4.1:</div><div>&gt; If Bob agrees [I am ignoring early=
 media for now], a PeerConnection</div><div>&gt; =A0 =A0is instantiated wit=
h the message from Alice&#39;s side.</div><div>&gt;=A0</div><div>&gt; I thi=
nk you can remove the parenthesis. First of all I don&#39;t think it</div>

<div>&gt; relavant in the paragraph. Secondly, early media doesn&#39;t exis=
t in the</div><div>&gt; context of a single peerConnection.</div><div><br><=
/div><div>Removed.</div><div><br></div><div><br></div><div>&gt; 7. Section =
4.3</div>

<div>&gt;=A0</div><div>&gt; If Alice and Bob authenticated via their IdPs,<=
/div><div>&gt; =A0 =A0then they also know that the signaling service is not=
 mounting a man-</div><div>&gt; =A0 =A0in-the-middle attack on theor traffi=
c.</div>

<div>&gt;=A0</div><div>&gt; theor / their?</div><div><br></div><div>Fixed.<=
/div><div><br></div><div><br></div><div>&gt; 8. Section 5.1:</div><div>&gt;=
=A0</div><div>&gt; It is RECOMMENDED that browsers which allow</div><div>&g=
t; =A0 =A0active mixed content nevertheless disable RTCWEB functionality in=
</div>

<div>&gt; =A0 =A0mixed content settings. [[ OPEN ISSUE: =A0Should this be a=
 2119 MUST?</div><div>&gt; =A0 =A0It&#39;s not clear what set of conditions=
 would make this OK, other than</div><div>&gt; =A0 =A0that browser manufact=
urers have traditionally been permissive here</div>

<div>&gt; =A0 =A0here.]]</div><div>&gt;=A0</div><div>&gt; I am actually qui=
te worried about not making this a MUST. Mixed content</div><div>&gt; clear=
ly causes significant security holes that I think really should be</div><di=
v>

&gt; avoided.</div><div><br></div><div>Agreed. Changed per-wg discussion.</=
div><div><br></div><div><br></div><div>&gt; 9. Section 5.1, similarly I thi=
nk this applies to the second open issue</div><div>&gt; in this section.</d=
iv>

<div><br></div><div>Changed. If people disagree, please objec.t</div><div><=
br></div><div><br></div><div>&gt; 10. Section 5.3:</div><div>&gt;=A0</div><=
div>&gt; [Note:</div><div>&gt; =A0 =A0this document takes no position on th=
e split between ICE in JS and</div>

<div>&gt; =A0 =A0ICE in the browser. =A0The above text is written the way i=
t is for</div><div>&gt; =A0 =A0editorial convenience and will be modified a=
ppropriately if the WG</div><div>&gt; =A0 =A0decides on ICE in the JS.] =A0=
The JS MUST NOT be permitted to control</div>

<div>&gt; =A0 =A0the local ufrag and password, though it of course knows it=
.</div><div>&gt;=A0</div><div>&gt; I think you can clean up this statement =
as there is consensus to leave</div><div>&gt; ICE in the browser.</div><div=
>
<br>
</div><div>Done.</div><div><br></div><div><br></div><div>&gt; 11. Section 5=
.3:</div><div>&gt; A separate document will profile the ICE</div><div>&gt; =
=A0 =A0timers to be used; see [I-D.muthu-behave-consent-freshness].</div><d=
iv>

&gt;=A0</div><div>&gt; We should clarify the WG consensus on this document =
and the scope the</div><div>&gt; document has.</div><div><br></div><div>I&#=
39;m not sure I understand this. Is this a proposed change to the</div><div=
>

document or a WG action item?</div><div><br></div><div><br></div><div>&gt; =
12. Section 5.4:</div><div>&gt;=A0</div><div>&gt; =A0 =A0API Requirement: =
=A0The API MUST provide a mechanism for the calling</div><div>&gt; =A0 =A0 =
=A0 application JS to indicate that only TURN candidates are to be</div>

<div>&gt; =A0 =A0 =A0 used. =A0This prevents the peer from learning one&#39=
;s IP address at</div><div>&gt; =A0 =A0 =A0 all.</div><div>&gt;=A0</div><di=
v>&gt; What about the ICE candidates related address information, that shou=
ld</div><div>

&gt; be mentioned I think also as possible risk of leaking and what to set =
it to.</div><div><br></div><div>Good point!</div><div><br></div><div><br></=
div><div>&gt; 13. I am missing a discussion on how to avoid the linkage iss=
ues. I</div>

<div>&gt; think we need to point out the known sources to linkage that WebR=
TC</div><div>&gt; adds. I am aware of RTCP CNAMEs and as well as the certif=
icate used in</div><div>&gt; DTLS(-SRTP). Their should be recommendations o=
n how to avoid these issues.</div>

<div><br></div><div>See S 5.5 below. Are you looking for something else.</d=
iv><div><br></div><div><br></div><div><br></div><div><br></div><div>&gt; 14=
. Section 5.5</div><div>&gt; =A0 =A0[OPEN ISSUE: =A0What should the setting=
s be here? =A0MUST?]</div>

<div>&gt; =A0 =A0Implementations MAY support SDES for media traffic for bac=
kward</div><div>&gt; =A0 =A0compatibility purposes.</div><div>&gt;=A0</div>=
<div>&gt; This needs to be resolved. I see no issues with retaining the</di=
v><div>

&gt; possibility for SDES and other schemes. I think however, the downgrade=
</div><div>&gt; issues may need more discussion here. This is a somewhat di=
scussed in</div><div>&gt; the security considerations section later.</div>

<div><br></div><div>I plan to hold off on this till after the discussion in=
 Berlin.</div><div><br></div><div><br></div><div>&gt; 15. Section 5.5</div>=
<div>&gt; =A0 =A0API Requirement: =A0The API MUST provide a mechanism to in=
dicate that a</div>

<div>&gt; =A0 =A0 =A0 fresh DTLS key pair is to be generated for a specific=
 call. =A0This</div><div>&gt; =A0 =A0 =A0 is intended to allow for unlinkab=
ility. =A0Note that there are also</div><div>&gt; =A0 =A0 =A0 settings wher=
e it is attractive to use the same keying material</div>

<div>&gt; =A0 =A0 =A0 repeatedly, especially those with key continuity-base=
d</div><div>&gt; =A0 =A0 =A0 authentication.</div><div>&gt;=A0</div><div>&g=
t; I think this has to do with 13. For example should really the same</div>=
<div>&gt; certificate be used with different origins, unless it is a intent=
ionally</div>

<div>&gt; added endpoint certificate that provide verifiable source?</div><=
div><br></div><div>No, it shouldn&#39;t be used with different origins exce=
pt for thjat.</div><div><br></div><div><br></div><div>&gt; 16. Section 5.5<=
/div>

<div>&gt; [largely derived from</div><div>&gt; =A0 =A0 =A0 [I-D.kaufman-rtc=
web-security-ui]</div><div>&gt;=A0</div><div>&gt; Maybe this is more suitab=
le in a contributors section instead of inline</div><div>&gt; in text.</div=
><div>

<br></div><div>Done.</div><div><br></div><div><br></div><div>&gt; 17. Secti=
on 5.5:</div><div>&gt;=A0</div><div>&gt; =A0 =A0 =A0 * =A0The &quot;securit=
y characteristics&quot; MUST indicate the cryptographic</div><div>&gt; =A0 =
=A0 =A0 =A0 =A0algorithms in use (For example: =A0&quot;AES-CBC&quot; or &q=
uot;Null Cipher&quot;.)</div>

<div>&gt;=A0</div><div>&gt; Does there need to be additional discussion or =
clarifications on high</div><div>&gt; level indications when one uses NULL =
cipher that doesn&#39;t provide</div><div>&gt; confidentiality?</div><div>

<br></div><div>Good point. Added a new requirement here.</div><div><br></di=
v><div><br></div><div>&gt; 18. Section 5.6.2:</div><div>&gt;=A0</div><div>&=
gt; The details of the mechanism are described in the W3C API</div><div>
&gt; =A0 =A0specification,</div>
<div>&gt;=A0</div><div>&gt; I think a reference needs to be added here.</di=
v><div>&gt;=A0</div><div>&gt; 19. Section 5.6.3</div><div>&gt; Todo REF!</d=
iv><div><br></div><div>Added.</div><div><br></div><div><br></div><div>&gt; =
20. Section <a href=3D"http://5.6.4.1">5.6.4.1</a>:</div>

<div>&gt;=A0</div><div>&gt; =A0 =A0The &quot;algorithm&quot; and digest val=
ues correspond directly to the</div><div>&gt; =A0 =A0algorithm and digest i=
n the a=3Dfingerprint line of the SDP.</div><div>&gt;=A0</div><div>&gt; Sho=
uld there be a reference to where a=3Dfingerprint is defined here?</div>

<div><br></div><div>Done.</div><div><br></div><div><br></div><div>&gt; Also=
, does it need to be clarified what &quot;algorithm&quot; and digest</div><=
div>&gt; reference, the ABNF contructs or value for those parameters.</div>

<div><br></div><div>I added &quot;values&quot; here. Is that what you were =
looking for?</div><div><br></div><div><br></div><div>&gt; 21. Section <a hr=
ef=3D"http://5.6.4.2">5.6.4.2</a>:</div><div>&gt; This SDP attribute is und=
erspecified. No clear definition of what is</div>

<div>&gt; allowed, no IANA consideration section registering it.</div><div>=
<br></div><div>It&#39;s just a base64-encoded identity attribute.</div><div=
><br></div><div><br></div><div>&gt; Also, should this SDP attribute also ca=
rry the origin of the IdP? I just</div>

<div>&gt; wonder how a non browser can determine which IdP has been providi=
ng the</div><div>&gt; assertion.</div><div><br></div><div>It&#39;s in the a=
ssertion. Do you think it would be better in the</div><div>a=3Dline?</div>

<div><br></div><div><br></div><div>&gt; 22. Section 5.6.5.1 and following.<=
/div><div>&gt;=A0</div><div>&gt; I would like that all these JSON objects w=
ould have a bit more crisp</div><div>&gt; definitions. Sure, the JSON has c=
ertain limiations in what values can be</div>

<div>&gt; used, but it is unclear if the full UTF-8 values can be used in a=
ll</div><div>&gt; cases, or if there is restrictions on the URL provided, o=
r if it should</div><div>&gt; be URI&#39;s in fact?</div><div><br></div>

<div>OK. I will circle around with Ted (as a URL expert) and anyone else</d=
iv><div>who feels expert to try to nail this down.</div><div><br></div><div=
>&gt; 23. I am missing a security discussion around the fact that I can cla=
im</div>

<div>&gt; to have an IdP and load whatever JS code into the proxy. What sec=
urity</div><div>&gt; implications deos this have. Section 5.6.5.2 does prov=
ide some basic</div><div>&gt; restrictions here. Are more needed?</div>

<div><br></div><div>I tried to hit this in 5.7.4. Generally if you are a si=
te you can</div><div>always load new JS into the browser, so this is just p=
art of the</div><div>threat model. Can you give me an example of what you&#=
39;re looking for.</div>

<div><br></div><div><br></div><div>&gt; 24. Section 5.6.5.2.2:</div><div>&g=
t;=A0</div><div>&gt; =A0 =A0idp: =A0A dictionary containing the domain name=
 of the provider and the</div><div>&gt; =A0 =A0 =A0 protocol string</div><d=
iv>&gt;=A0</div>

<div>&gt; What is the definition of dictionary here?</div><div><br></div><d=
iv>I&#39;m not sure enough of the terminology here. Isn&#39;t dictionary</d=
iv><div>what we call JS associative-array type structures.</div><div><br>

</div><div><br></div><div><br></div><div>&gt; 25. Are there any time limits=
 on responding to SIGN or Verify, or can</div><div>&gt; you get application=
s to hand simply by not responding to a IdP message?</div><div><br></div>

<div>I would hope the browser would produce an error at some point,</div><d=
iv>but I&#39;m not sure we need to specify here.</div><div><br></div><div><=
br></div><div>&gt; 26. Section 5.6.5.2.3:</div><div>&gt;=A0</div><div>&gt; =
Are the fields required to be included or not. This is a bit unclear.</div>

<div><br></div><div>Currently it says:</div><div><br></div><div>&quot;MUST =
contain a message field consisting of a dictionary/hash with the following =
fields:&quot;</div><div><br></div><div>Would you like it to say &quot;which=
 MUST contain the following fields&quot;</div>

<div><br></div><div><br></div><div><br></div><div>&gt; Section 27. Section =
5.6.5.2.3:</div><div>&gt; Open issue to resolve. Please try to resolve this=
 issue by talking to</div><div>&gt; some people.</div><div><br></div><div>

Agreed. This is now on my TODO list.</div><div><br></div><div><br></div><di=
v>&gt; 27. Section 5.6.5.2.3.1</div><div>&gt;=A0</div><div>&gt; Open issue =
needing resolution. No opinion.</div><div><br></div><div>Will raise this in=
 Berlin.</div>

<div><br></div><div><br></div><div>&gt; 28. Section 5.7.2:</div><div>&gt;=
=A0</div><div>&gt; Missing the linkability issues discussion here.</div><di=
v><br></div><div>Added some material.</div><div><br></div><div><br></div><d=
iv>

&gt; 29. Section 5.7.2:</div><div>&gt;=A0</div><div>&gt; Combined RTCWEB/To=
r</div><div>&gt; =A0 =A0implementations SHOULD arrange to route the media a=
s well as the</div><div>&gt; =A0 =A0signaling through Tor. [Currently this =
will produce very suboptimal</div>

<div>&gt; =A0 =A0performance.]</div><div><br></div><div>Done.</div><div><br=
></div><div><br></div><div>&gt; I think you can remove the parenthis but ke=
ep the text here.</div><div>&gt;=A0</div><div>&gt; 30. Page 40 SDP example =
looks strange with no a=3Didenity attribute,</div>

<div>&gt; instead just the JSON object.</div><div>Fixed.</div><div><br></di=
v><div><br></div><div><br></div><div>OHLSSON</div><div>&gt; When reading dr=
aft-ietf-rtcweb-security-arch I noticed a couple of things that might be mi=
ssing:</div>

<div>&gt;=A0</div><div>&gt; - According to section 4.3.2.4 in draft-ietf-rt=
cweb-security there shold</div><div>&gt; =A0 be a mechanism to verify that =
the remote browser has engaged a &quot;secure</div><div>&gt; =A0 media mode=
&quot;. This mechanism is not (yet) described in the document.</div>

<div><br></div><div>Good catch. I have to write something here but I haven&=
#39;t yet.</div><div><br></div><div><br></div><div>&gt; - Except for paragr=
aph 6 in section 4.1, there is not much said about</div><div>&gt; =A0 the i=
mplications of the &quot;secure media mode&quot;.</div>

<div><br></div><div>This is referring to isolated streams, which is now in =
gUM. How much</div><div>do you think I should say here?</div><div><br></div=
><div><br></div><div>&gt; - An enterprise network with an HTTP proxy may no=
t want external web sites to</div>

<div>&gt; =A0 learn the IP addresses of its hosts using the PeerConnection =
API. I&#39;m</div><div>&gt; =A0 not sure if this kind of attack is relevant=
 or not. If it is then</div><div>&gt; =A0 it should be mentioned in Section=
 5.4. IP Location Privacy.</div>

<div><br></div><div>Added some text.</div><div>o</div><div>&gt; I also have=
 some minor comments on the rest of the document:</div><div>&gt;=A0</div><d=
iv>&gt; - Section 3, 1st paragraph</div><div>&gt;=A0</div><div>&gt; =A0 =A0=
The basic assumption of this architecture is that network resources</div>

<div>&gt; =A0 =A0exist in a hierarchy of trust, rooted in the browser, whic=
h serves as</div><div>&gt; =A0 =A0the user&#39;s TRUSTED COMPUTING BASE (TC=
B).</div><div>&gt;=A0</div><div>&gt; =A0 The term &quot;hierarchy of trust&=
quot; is unclear in this context. Are there other</div>

<div>&gt; =A0 nodes in this hierarchy except for the browser and web server=
?</div><div><br></div><div>I would claim browser, web server, idp (if any),=
 other network elements.</div><div><br></div><div><br></div><div><br></div>

<div>&gt; - Section 3, 2nd paragraph</div><div>&gt;=A0</div><div>&gt; =A0 =
=A0 This is a natural extension of the end-to-end principle.</div><div>&gt;=
=A0</div><div>&gt; =A0 Perhaps it&#39;s just me but I don&#39;t understand =
what this end-to-end principle is.</div>

<div><br></div><div>I just removed it since it doesn&#39;t add value.</div>=
<div><br></div><div><br></div><div>&gt; - Section 4.3, 1st paragraph</div><=
div>&gt;=A0</div><div>&gt; =A0 =A0 The total number of channels depends on =
the amount of</div>

<div>&gt; =A0 =A0 muxing; in the most likely case we are using both RTP/RTC=
P mux and</div><div>&gt; =A0 =A0 muxing multiple media streams on the same =
channel, in which case</div><div>&gt; =A0 =A0 there is only one DTLS handsh=
ake.</div>

<div>&gt;=A0</div><div>&gt; =A0 =A0Won&#39;t there be separate handshake fo=
r the data channel?</div><div><br></div><div>My understanding was that we c=
ould mux the datachannel onto</div><div>the same 5-tuple.</div><div><br></d=
iv>

<div><br></div><div>&gt; - Section 5.4, 1st paragraph</div><div>&gt;=A0</di=
v><div>&gt; =A0 =A0 [...] which leaks large amounts of location information=
,</div><div>&gt; =A0 =A0 especially for mobile devices.</div><div>&gt;=A0</=
div><div>

&gt; =A0 =A0Why are mobile devices different in this aspect?</div><div><br>=
</div><div>They&#39;re not. Removed.</div><div><br></div><div><br></div><di=
v>&gt; - Section 5.6.4</div><div>&gt;=A0</div><div>&gt; =A0 The SDP example=
 contains RTP/AVP but I thought SRTP was made mandatory?</div>

<div><br></div><div>Bit rot. Fixed.</div><div><br></div><div><br></div><div=
>&gt; - Section 5.6.5.1, 3rd paragraph</div><div>&gt;=A0</div><div>&gt; =A0=
 =A0All requests from the PeerConnection object MUST contain an &quot;id&qu=
ot;</div>

<div>&gt; =A0 =A0field which MUST be unique for that PeerConnection object.=
 =A0Any</div><div>&gt; =A0 =A0responses from the IdP proxy MUST contain the=
 same id in response,</div><div>&gt; =A0 =A0which allows the PeerConnection=
 to correlate requests and responses.</div>

<div>&gt;=A0</div><div>&gt; =A0 Couldn&#39;t the browser implementation han=
dle the message routing</div><div>&gt; =A0 without the id field?</div><div>=
<br></div><div>This isn&#39;t about routing, it&#39;s about what happens if=
 there are</div>

<div>multiple outstanding queries.</div><div><br></div><div><br></div><div>=
&gt; - Section 5.6.5.1.1</div><div>&gt;=A0</div><div>&gt; =A0 The mandatory=
 id field is missing in the example</div><div><br></div><div>Fixed.</div><d=
iv>

<br></div><div><br></div><div>&gt; - Section 5.6.5.2.2</div><div>&gt;=A0</d=
iv><div>&gt; =A0 How is the assertion field transformed into the a=3Didenti=
ty attribute?</div><div>&gt; =A0 I guess you=B4re using some form of B64 en=
coding but this is not expained</div>

<div>&gt; =A0 anywhere.</div><div><br></div><div>It&#39;s earlier, but I ad=
ded it.</div><div><br></div><div><br></div><div>&gt; - Section 5.6.5.2.3</d=
iv><div>&gt;=A0</div><div>&gt; =A0 I think the reason for including request=
_origin should be explained here</div>

<div>&gt; =A0 instead of in the end of the document.</div><div><br></div><d=
iv>I added a forward reference instead to avoid breaking up the document</d=
iv><div>flow.</div><div><br></div><div><br></div><div>&gt; - Section 5.7.1,=
 3rd paragraph</div>

<div>&gt;=A0</div><div>&gt; =A0 WebCrypto API lacks a reference.</div><div>=
&gt;=A0</div><div>&gt; - Section 5.7.4.5.2</div><div>&gt;=A0</div><div>&gt;=
 =A0 =A0 Any IdP which uses cookies to persist logins will be broken</div><=
div>&gt; =A0 =A0 by third-party cookie blocking.</div>

<div>&gt;=A0</div><div>&gt; =A0 Shouldn&#39;t it say &quot;Any IdP which us=
es third-party cookies&quot;?</div><div><br></div><div>IdPs are inherently =
third-party in this context.</div><div><br></div><div><br></div><div>&gt; -=
 Appendix A and B</div>

<div>&gt;=A0</div><div>&gt; =A0 Both appendices appear unfinished. For exam=
ple, ROAP is still mentioned</div><div>&gt; =A0 in othe BrowserID example a=
nd it&#39;s unclear what client credentials</div><div>&gt; =A0 Bob uses in =
the OAuth example.</div>

<div><br></div><div>Agreed. I have a TODO to clean these up.</div><div><br>=
</div><div><br></div><div><br></div><div><br></div><div>JOHNSTON</div><div>=
--------</div><div>&gt; Again, lots of RTCWEB, RTCWeb, etc that should be W=
ebRTC.</div>

<div><br></div><div>Fixed.</div><div><br></div><div><br></div><div>o&gt; Th=
e document introduction seems to be specific to audio and video</div><div>=
=A0 WebRTC sessions. =A0Later, it becomes clear that the analysis applies</=
div>

<div>=A0 to the data channel as well. =A0This should be stated clearly up f=
ront</div><div>=A0 and any differences between the two should be explained =
(i.e. DTLS</div><div>=A0 vs SRTP encryption).</div><div><br></div><div>I ad=
ded a sentence in S 4.3. This is actually the first significant</div>

<div>discussion of SRTP, so I think this works best.</div><div><br></div><d=
iv><br></div><div><br></div><div>&gt; &quot;SIP or XMPP&quot; - XMPP isn&#3=
9;t a signaling protocol but Jingle is, so that is probably what should be =
mentioned, but Jingle doesn&#39;t use SDP.</div>

<div><br></div><div>I just left it as SIP.</div><div><br></div><div><br></d=
iv><div>&gt; &quot;Web sites whose origin we can verify (optimally via HTTP=
S, but in some cases because we are on a topologically restricted network, =
such as behind a firewall)&quot; =A0- what is the 2nd case - no verificatio=
n? =A0Verification using something other than HTTPS?</div>

<div><br></div><div>I added &quot;and can infer authentication from firewal=
l behavior&quot;</div><div><br></div><div>&gt; 3.1 =A0middle para graph is =
basically saying that security policy can be applied after authentication -=
 might be worth stating this after the Dr. Evil analogy.</div>

<div><br></div><div>Added some text.</div><div><br></div><div>&gt; 4. &quot=
;Specifically, Alice and Bob have relationships with</div><div>&gt; =A0 =A0=
some Identity Provider (IdP) that supports a protocol such as OpenID</div>
<div>
&gt; =A0 =A0or BrowserID) that can be used to demonstrate their identity to=
 other</div><div>&gt; =A0 =A0parties.&quot; - needs one more ( to parse cor=
rectly.</div><div>&gt;=A0</div><div>&gt; Figures 3 &amp; 4. =A0Would be bet=
ter to have IdP1 and IdP2 to make clear that this can be 2 different provid=
ers. Also, Secure WebSockets should be shown with HTTPS. =A0=A0</div>

<div><br></div><div>Done.</div><div><br></div><div><br></div><div>&gt; In F=
igures 3 &amp; 4, media is shown as DTLS-SRTP when the media is in fact SRT=
P which is keyed with DTLS-SRTP. =A0In fact, RFC 3711 is not even reference=
d, when it should be a normative reference, which is potentially very confu=
sing and misleading for implementors. =A0There are many places where DTLS-S=
RTP described as if it is more than a key agreement protocol for SRTP.</div=
>

<div><br></div><div>I changed this to be &quot;DTLS+SRTP&quot;, added a ref=
erence to RFC 3711, and</div><div>went through each use of DTLS-SRTP for ap=
propriateness, modifying</div><div>where I thought necessary. Please let me=
 know if there are places I</div>

<div>missed.</div><div><br></div><div><br></div><div>&gt; Figure 4 is the T=
rapezoid of the overview and should be referenced as such.</div><div><br></=
div><div>Added a back reference to figure 2.</div><div><br></div><div>
&gt; JS is used but not defined. =A0</div>
<div><br></div><div>Defined it back in the intro.</div><div><br></div><div>=
<br></div><div>&gt; 4.1 =A0The first example in this document of microphone=
 and webcam permissions is that of a permanent grant. =A0Is this the model =
we want to encourage? =A0Shouldn&#39;t the first mention be of the much saf=
er per call grant? =A0Later the per call grant is mentioned, but deep in th=
e document.</div>

<div><br></div><div>Rewritten.</div><div><br></div><div><br></div><div>&gt;=
 4.1 &quot;This message is sent to the signaling server, e.g., by XMLHttpRe=
quest[XmlHttpRequest] or by WebSockets [RFC6455] &quot; =A0Sentence is miss=
ing a period. =A0Do we really mean HTTPS and Secure WebSockets here?</div>

<div><br></div><div>I added &quot;preferably over TLS&quot;.</div><div><br>=
</div><div><br></div><div>&gt; 4.1 &quot;This allows the browser to display=
 a trusted element in the browser chrome indicating that a call is coming i=
n from Alice.&quot; =A0Can this assertion be made now, prior to the DTLS-SR=
TP handshake completing? =A0If this identity assertion is made at this stag=
e, then a different fingerprint shows up in the DTLS-SRTP handshake, does t=
he browser chrome then indicate &#39;sorry, I made a mistake, it isn&#39;t =
Alice calling&#39;?</div>

<div><br></div><div>I think at that point you just show a network error. Th=
e identity</div><div>assertion indicates that Alice was trying to call and =
if someone</div><div>is attacking, then that&#39;s just a failure.</div>

<div><br></div><div><br></div><div>&gt; There are many cases of using [] in=
stead of () for parentheical text.</div><div><br></div><div>I try to use []=
 for Notes or open issues.</div><div><br></div><div><br></div><div>&gt; 4.2=
 ICE is not defined or referenced in its first occurance.</div>

<div><br></div><div>Fixed in 4.1.</div><div><br></div><div>&gt; typo &quot;=
theor&quot;</div><div>&gt;=A0</div><div>&gt; 5.1 &quot;[[ OPEN ISSUE: =A0Sh=
ould this be a 2119 MUST?</div><div>&gt; =A0 =A0It&#39;s not clear what set=
 of conditions would make this OK, other than</div>

<div>&gt; =A0 =A0that browser manufacturers have traditionally been permiss=
ive here</div><div>&gt; =A0 =A0here.]] &quot;=A0</div><div>&gt;=A0</div><di=
v>&gt; =A0 =A0[[ OPEN ISSUE:: =A0Should we be more aggressive about this?]]=
</div><div>&gt;=A0</div>

<div>&gt; Do these issues need to be discussed in a W3C document where brow=
ser expertise might help us do the right thing?</div><div><br></div><div>I =
think we resolved this in Orlando and the document reflects what I think</d=
iv>

<div>the consensus was.</div><div><br></div><div><br></div><div>&gt; 5.2 =
=A0Do we really expect browsers to have X.509 certificates? =A0Does anyone =
do this with DTLS-SRTP today? =A0Do browsers have UI to manage these certs?=
</div>

<div><br></div><div>No, we don&#39;t. This was a request from Martin becaus=
e he was expecting</div><div>servers might. I added some text to try to mak=
e that clear.</div><div><br></div><div><br></div><div><br></div><div>&gt; 5=
.2 The API and UI requirements and elsewhere - should these be in a W3C doc=
ument instead of/in addition to this document?</div>

<div><br></div><div>My sense is that IETF was responsible for the security =
analysis so they</div><div>belong here.</div><div><br></div><div><br></div>=
<div>&gt; 5.2 &quot;Implementations which support some form of direct user =
authentication</div>

<div>&gt; =A0 =A0SHOULD also provide a policy by which a user can authorize=
 calls only</div><div>&gt; =A0 =A0to specific counterparties.&quot; =A0What=
 is a counterparty?</div><div><br></div><div>Changed to &quot;communicating=
 peers.&quot;</div>

<div><br></div><div><br></div><div>&gt; 5.3 =A0 ICE-Lite is not defined or =
referenced. =A0Since this is a MUST, we need a normative reference - is it =
what is described as ICE Lite in RFC 5245 or draft-rescorla-mmusic-ice-lite=
?</div>

<div><br></div><div>It&#39;s 5245 again. I added a cite.</div><div><br></di=
v><div><br></div><div>&gt; Question: =A0If a browser has multiple public/pr=
ivate keys, including an X.509 one, can the JS suggest which one to use for=
 a particular PeerConnection?</div>

<div><br></div><div>I think the answer needs to be yes here, but we probabl=
y need to clean</div><div>up the W3C API a bit to match.</div><div><br></di=
v><div><br></div><div>&gt; 5.5 Again, the media channel is secured using SR=
TP, not DTLS-SRTP,</div>

<div>&gt; which is one way to key SRTP. =A0In the future, new and better ke=
y</div><div>&gt; agreements will be developed, and if we write all our spec=
s assuming</div><div>&gt; one particular keying method, then they will beco=
me obsolete.</div>

<div><br></div><div>Good point. Here is my new text which I hope is clearer=
:</div><div><br></div><div>=A0 =A0Implementations MUST implement SRTP [RFC3=
711]. =A0Implementations MUST</div><div>=A0 =A0implement DTLS [RFC4347] and=
 DTLS-SRTP [RFC5763][RFC5764] for SRTP</div>

<div>=A0 =A0keying. =A0Implementations MUST implement</div><div>=A0 =A0[I-D=
.ietf-tsvwg-sctp-dtls-encaps].</div><div><br></div><div>=A0 =A0All media ch=
annels MUST be secured via SRTP. =A0Media traffic MUST NOT</div><div>=A0 =
=A0be sent over plain (unencrypted) RTP. =A0DTLS-SRTP MUST be offered for</=
div>

<div>=A0 =A0every media channel and MUST be the default; i.e., if an</div><=
div>=A0 =A0implementation receives an offer for DTLS-SRTP and SDES, DTLS-SR=
TP</div><div>=A0 =A0MUST be selected.</div><div><br></div><div>=A0 =A0All d=
ata channels MUST be secured via DTLS.</div>

<div><br></div><div><br></div><div>&gt; Section 5.6 with Appendix A reads l=
ike a completely separate</div><div>&gt; document. =A0It is also very hard =
reading as there are lots of new</div><div>&gt; concepts and ideas. =A0Shou=
ld perhaps this be in a separate document?</div>

<div>&gt; The basic security architecture for WebRTC is unlikely to change,=
</div><div>&gt; but given that we have no experience with this IdP system, =
it seems</div><div>&gt; likely that it will evolve and perhaps change signi=
ficantly, which</div>

<div>&gt; is an argument for a separate document.</div><div><br></div><div>=
I agree the writing is a bit awkward. I&#39;m rewriting App A to make it</d=
iv><div>clearer and I&#39;ll see what I can do to about making S 5.6 merge =
better</div>

<div>as well.</div><div><br></div><div><br></div><div>&gt; 5.6.4.2 =A0Is th=
is document defining a new SDP attribute? =A0Shouldn&#39;t this be done in =
MMUSIC?</div><div><br></div><div>I have a TODO here. Will consult with the =
chairs.</div>

<div><br></div><div><br></div><div>&gt; typos &quot;implemementations&quot;=
 =A0&quot;termnating&quot; &quot;restrcitions&quot;=A0</div><div><br></div>=
<div>Fixed.</div><div><br></div><div><br></div><div>&gt; WebIntents mention=
ed without reference or explanation.</div>

<div><br></div><div>WebIntents is OBE. Removed.</div><div><br></div><div><b=
r></div><div>&gt; Missing normative reference: RFC 3711.</div><div><br></di=
v><div>Fixed.</div><div><br></div><div><br></div><div>-Ekr</div></div>
<div>
<br></div><div><br></div><div><br></div><div><br></div></div>

--047d7b6da78246f54a04e18ce9ab--

From bo.burman@ericsson.com  Mon Jul 15 08:15:26 2013
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From: Bo Burman <bo.burman@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Some thoughts on optional audio codecs
Thread-Index: Ac6BN11IugiIgzm2TXug3PomYB8N3A==
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Subject: [rtcweb] Some thoughts on optional audio codecs
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Regarding the previous discussion on optional audio codecs in the (currentl=
y expired) draft on RTCWEB audio codecs (https://datatracker.ietf.org/doc/d=
raft-ietf-rtcweb-audio/):

I think most parties involved in WebRTC work, myself included, hope and bel=
ieve that it will be ubiquitous and easy to include real-time media convers=
ation functionality in nearly any web context. Since it will be that easy, =
it can be expected that most web developers need not be, and thus will not =
be, media specialists or very knowledgeable about codecs.

The definition of RTCWEB MTI codecs ensures that communication is possible =
since at least one codec will always be found, but it is not possible to cl=
aim the resulting communication to be optimum for every possible context.

Even if WebRTC will be close to ubiquitous, there will for quite some time =
likely be a desire to reach real-time media domains and devices that were n=
ot originally designed for and thus are not optimized for use with WebRTC. =
A communication device that is not designed solely for WebRTC use will like=
ly include functionality and codecs also for its "native" domain.

Any added cost of not being able to use existing "native" codecs will vary =
both in amount and where the cost has to be taken. Eliminating it is indeed=
 an optimization, but the total cost savings may still be significant.

With the current design and to my understanding, it will be the browser ven=
dor's choice to add optional codecs, including any "native" domain codecs. =
 The choice may possibly be delegated to individual web developers making u=
se of WebRTC functionality. A browser vendor will arguably have to know eac=
h target platform to some extent, but it can hardly be assumed that a web d=
eveloper knows the capabilities of all devices that will use the WebRTC-ena=
bled site unless the browser can provide the needed information. There is a=
 risk that "native" codecs in devices are not well handled, unless the moti=
vations and methods to make use of them are better specified.

While any audio codecs besides the MTI ones are clearly optional, I believe=
 the suggested text addition on optional audio codecs to the RTCWEB audio d=
raft in Ticket #12 (http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/12#) t=
o be too brief considering the above.

In that draft, I would prefer something more in line with:

"If other suitable audio codecs are available to the browser to use, it is =
recommended that they are also included in the offer in order to maximize t=
he possibility to establish the session without the need for audio transcod=
ing".

Assuming that the browser vendor (or web developer) is sufficiently concern=
ed with codecs to read the audio codecs draft (or the corresponding RFC to-=
be), the above text may, as a start, give some added guidance why non-MTI c=
odecs may be desirable to consider in addition to the MTI ones.

Cheers,
Bo

Multimedia Technologies
Ericsson Research
F=E4r=F6gatan 6
SE-164 80, Kista, Sweden
www.ericsson.com

From suhasietf@gmail.com  Mon Jul 15 08:43:34 2013
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--001a11c3505ea86bbb04e18eb8be
Content-Type: text/plain; charset=ISO-8859-1

Hello All

   An updated version of the SDP Attributes Mux draft has been submitted
with reviews for close to 50% of the attributes completed. Thanks every one
for their valuable inputs. We are half-way there :)


Cheers
Suhas Nandakumar

A new version of I-D, draft-nandakumar-mmusic-sdp-mux-attributes-03.txt
has been successfully submitted by Suhas Nandakumar and posted to the
IETF repository.

Filename:        draft-nandakumar-mmusic-sdp-mux-attributes
Revision:        03
Title:           A Framework for SDP Attributes when Multiplexing
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 56
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-sdp-mux-attributes-03.txt
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-sdp-mux-attributes
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-03
Diff:
http://www.ietf.org/rfcdiff?url2=draft-nandakumar-mmusic-sdp-mux-attributes-03

Abstract:
   The Session Description Protocol (SDP) provides mechanisms to
   describe attributes of multimedia sessions and of individual media
   streams (e.g., Real-time Transport Protocol (RTP) sessions) within a
   multimedia session.  In the RTCWeb WG, there is a need to use a
   single 5-tuple for sending and receiving media associated with
   multiple media descriptions ("m=" lines).  Such a requirement has
   raised concerns over the semantic implications of the SDP attributes
   associated with the RTP Sessions multiplexed over a single transport
   layer flow.

   The scope of this specification is to provide a framework for
   analyzing the multiplexing characteristics of SDP attributes.  The
   specification also categorizes existing attributes based on the
   framework described herein.

--001a11c3505ea86bbb04e18eb8be
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello All<br><div class=3D"gmail_quote"><div><br></div><div>=A0 =A0An updat=
ed version of the SDP Attributes Mux draft has been submitted with reviews =
for close to 50% of the attributes completed. Thanks every one for their va=
luable inputs. We are half-way there :)</div>

<div><br></div><div><br></div><div>Cheers</div><div>Suhas Nandakumar</div><=
div><br></div><div>A new version of I-D, draft-nandakumar-mmusic-sdp-mux-at=
tributes-03.txt</div><div>has been successfully submitted by Suhas Nandakum=
ar and posted to the</div>

<div>IETF repository.</div><div><br></div><div>Filename: =A0 =A0 =A0 =A0dra=
ft-nandakumar-mmusic-sdp-mux-attributes</div><div>Revision: =A0 =A0 =A0 =A0=
03</div><div>Title: =A0 =A0 =A0 =A0 =A0 A Framework for SDP Attributes when=
 Multiplexing</div><div>

Creation date: =A0 2013-07-15</div><div>Group: =A0 =A0 =A0 =A0 =A0 Individu=
al Submission</div><div>Number of pages: 56</div><div>URL: =A0 =A0 =A0 =A0 =
=A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts/draft-nandakumar-mmu=
sic-sdp-mux-attributes-03.txt" target=3D"_blank">http://www.ietf.org/intern=
et-drafts/draft-nandakumar-mmusic-sdp-mux-attributes-03.txt</a></div>

<div>Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/=
draft-nandakumar-mmusic-sdp-mux-attributes" target=3D"_blank">http://datatr=
acker.ietf.org/doc/draft-nandakumar-mmusic-sdp-mux-attributes</a></div><div=
>Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-nanda=
kumar-mmusic-sdp-mux-attributes-03" target=3D"_blank">http://tools.ietf.org=
/html/draft-nandakumar-mmusic-sdp-mux-attributes-03</a></div>

<div>Diff: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/rfcdiff?ur=
l2=3Ddraft-nandakumar-mmusic-sdp-mux-attributes-03" target=3D"_blank">http:=
//www.ietf.org/rfcdiff?url2=3Ddraft-nandakumar-mmusic-sdp-mux-attributes-03=
</a></div><div>
<br></div><div>Abstract:</div>
<div>=A0 =A0The Session Description Protocol (SDP) provides mechanisms to</=
div><div>=A0 =A0describe attributes of multimedia sessions and of individua=
l media</div><div>=A0 =A0streams (e.g., Real-time Transport Protocol (RTP) =
sessions) within a</div>

<div>=A0 =A0multimedia session. =A0In the RTCWeb WG, there is a need to use=
 a</div><div>=A0 =A0single 5-tuple for sending and receiving media associat=
ed with</div><div>=A0 =A0multiple media descriptions (&quot;m=3D&quot; line=
s). =A0Such a requirement has</div>

<div>=A0 =A0raised concerns over the semantic implications of the SDP attri=
butes</div><div>=A0 =A0associated with the RTP Sessions multiplexed over a =
single transport</div><div>=A0 =A0layer flow.</div><div><br></div><div>=A0 =
=A0The scope of this specification is to provide a framework for</div>

<div>=A0 =A0analyzing the multiplexing characteristics of SDP attributes. =
=A0The</div><div>=A0 =A0specification also categorizes existing attributes =
based on the</div><div>=A0 =A0framework described herein.</div>
</div><br>

--001a11c3505ea86bbb04e18eb8be--

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From: ken carlberg <carlberg@g11.org.uk>
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To: Toerless Eckert (eckert) <eckert@cisco.com>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [rtcweb] [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

some food for thought...

about 10 years ago, I put together a draft that added an optional TLV =
prioritization header to RTP to accomplish something similar in terms of =
having the app decide which was the better RTP packet to drop during =
times of congestion.  My aim was to have these decisions made at =
forwarding application-layer gateways that did not act as a RTP/RTCP =
endpoint.  The strength of the approach (in my opinion :-) was that it =
could span pockets of non-diff-serv regions and still carry significance =
downstream.  The weakness was the need for DPI if one wanted to make a =
more informed decision along the path about dropping a packet.

I wonder if such a dual-layer design might be helpful in your efforts.  =
In any case, I have an interest in the normalizer draft.

-ken


On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> =
wrote:

> When dealing with congestion for real-time datagram (UDP/RTP) based =
video flows, one idea
> that we think has shown to provide good value is to intelligently drop =
in the network
> preferentially the least important packets within a video flow, for =
example non-referenced
> P-frames, because their loss creates the lowest impact to video =
quality.
>=20
> One of the main concerns raised against such schemes was one of =
fairness. If flows=20
> would start to indicate different priorities for packets in them, how =
would we
> be able to avoid that flows would indicate all their packets to be of =
high priority
> to shift packet drops to other flows. Or even more importantly: how =
could we motivate
> flows to indicate that some of their packets are low priority without =
them having to
> fear that they would experience more loss than than flows that don't =
do this.
>=20
> So, there is one draft that we would like to present (presumably in =
TSVWG)  explaining
> how we think this problem can be solved: =
draft-lai-tsvwg-normalizer-00.txt
>=20
> (Actually, this is how we did solve the problem in our implementatin =
of intelligent dropping,
> so this is not theoretical).
>=20
> Now granted, the problem of fairness is but one element of building a =
complete ssytem
> around the idea of intelligent dropping, so if folks here on the lists =
would like to
> see for example=20
>=20
>  a) how well can it work in routers to shift packet drops to low-prio =
packets
>  b) whats the evidence that it's good for video quality
>  c) How should we do the marking best going forward
>=20
> Then please let us know. We'd be very interested to provide insight =
into the parts of
> those pieces that we have also worked out or discuss the ones we have =
not (primarily c ;-)
>=20
> Thanks!
>    Toerless
>=20
> In-Reply-To: =
<A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
>=20
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) =
wrote:
>> Hi all,
>>=20
>> We have submitted a new Internet draft to describe a Normalization =
Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The =
goal of NM deployment is to create a new incentive for video encoders to =
generate more discardable packets when using AF PHB in DIffServ. We are =
looking forward to your review comments.
>>=20
>> Best regards,
>> CJ
>>=20
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert =
(eckert); Fred Yip (fyip)
>> Subject: New Version Notification for =
draft-lai-tsvwg-normalizer-00.txt
>>=20
>>=20
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-lai-tsvwg-normalizer
>> Revision:	 00
>> Title:		 Normalization Marker for AF PHB Group in =
DiffServ
>> Creation date:	 2013-02-08
>> WG ID:		 Individual Submission
>> Number of pages: 15
>> URL:             =
http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
>> Status:          =
http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
>> Htmlized:        =
http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
>>=20
>>=20
>> Abstract:
>>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>   nodes to normalize the distribution of DS codepoint (DSCP) markings
>>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>   useful for traffic conditioning with Active Queue Management (AQM)
>>   when the AF PHB group is deployed for multimedia service classes =
that
>>   carry video packets with advanced codec technologies.  Using NM can
>>   offer an incentive that is however missing in today's ecosystem of
>>   practical deployment, i.e., when congestion occurs in the AF PHB
>>   group, the network should allow a video codec to benefit from its
>>   effort of generating finer layers of intra-flow precedence (IFP) =
with
>>   discardable packets in a more balanced distribution.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


From carlberg@g11.org.uk  Mon Jul 15 06:41:33 2013
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From: ken carlberg <carlberg@g11.org.uk>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [rtcweb] [rmcat] [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

sorry for the tardy reply.  I'll dig through my archives and send you a =
copy.  my draft proposed adding a "classifier" field, but I only did one =
rev (which was missing some better rationale after a long talk I had =
with Dave Oran) and i suspect that it would need a fair amount of =
scrubbing before re-issuing another version.  the first order of =
business, though, will be to gauge the feedback/comments to your draft.

-ken

ps, CJ thanks for your clarification in your previous response


On Jul 11, 2013, at 8:30 PM, Toerless Eckert <eckert@cisco.com> wrote:

> Thanks, Ken
>=20
> Indeed yes, that's exactly the line of thought that we would like to
> see proceed as well. Any URL still existing for that draft ?
>=20
> "application layer gateway" could specifically mean switching MCU =
using
> intelligent dropping in conjunction with outbound congestion control.
>=20
> I definitely would like to see RTP headers as the ultimatele marking
> option even for routers to do intelligent dropping (or other packet
> processing "beyond DSCP)). Intra-flow (per-packet) DSCP is just a big =
pain=20
> (aka: impossible) for applications. Many OSs don't have any APIs to =
support
> this. And DSCP management in networks is already difficult enough as =
it
> stands.
>=20
> Would certainly like to see if there is interest to write up r present
> more pieces of this puzzle, especially an RTP extension. We just felt =
that the
> fairness in intelligent dropping was the biggest concern our video
> application folks had, that's why we wanted to explain how that piece
> could work first.=20
>=20
> Cheers
>    Toerless
>=20
>=20
> On Thu, Jul 11, 2013 at 12:04:58PM -0400, ken carlberg wrote:
>> Toerless,
>>=20
>> some food for thought...
>>=20
>> about 10 years ago, I put together a draft that added an optional TLV =
prioritization header to RTP to accomplish something similar in terms of =
having the app decide which was the better RTP packet to drop during =
times of congestion.  My aim was to have these decisions made at =
forwarding application-layer gateways that did not act as a RTP/RTCP =
endpoint.  The strength of the approach (in my opinion :-) was that it =
could span pockets of non-diff-serv regions and still carry significance =
downstream.  The weakness was the need for DPI if one wanted to make a =
more informed decision along the path about dropping a packet.
>>=20
>> I wonder if such a dual-layer design might be helpful in your =
efforts.  In any case, I have an interest in the normalizer draft.
>>=20
>> -ken
>>=20
>>=20
>> On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) =
<eckert@cisco.com> wrote:
>>=20
>>> When dealing with congestion for real-time datagram (UDP/RTP) based =
video flows, one idea
>>> that we think has shown to provide good value is to intelligently =
drop in the network
>>> preferentially the least important packets within a video flow, for =
example non-referenced
>>> P-frames, because their loss creates the lowest impact to video =
quality.
>>>=20
>>> One of the main concerns raised against such schemes was one of =
fairness. If flows=20
>>> would start to indicate different priorities for packets in them, =
how would we
>>> be able to avoid that flows would indicate all their packets to be =
of high priority
>>> to shift packet drops to other flows. Or even more importantly: how =
could we motivate
>>> flows to indicate that some of their packets are low priority =
without them having to
>>> fear that they would experience more loss than than flows that don't =
do this.
>>>=20
>>> So, there is one draft that we would like to present (presumably in =
TSVWG)  explaining
>>> how we think this problem can be solved: =
draft-lai-tsvwg-normalizer-00.txt
>>>=20
>>> (Actually, this is how we did solve the problem in our implementatin =
of intelligent dropping,
>>> so this is not theoretical).
>>>=20
>>> Now granted, the problem of fairness is but one element of building =
a complete ssytem
>>> around the idea of intelligent dropping, so if folks here on the =
lists would like to
>>> see for example=20
>>>=20
>>> a) how well can it work in routers to shift packet drops to low-prio =
packets
>>> b) whats the evidence that it's good for video quality
>>> c) How should we do the marking best going forward
>>>=20
>>> Then please let us know. We'd be very interested to provide insight =
into the parts of
>>> those pieces that we have also worked out or discuss the ones we =
have not (primarily c ;-)
>>>=20
>>> Thanks!
>>>   Toerless
>>>=20
>>> In-Reply-To: =
<A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
>>>=20
>>> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) =
wrote:
>>>> Hi all,
>>>>=20
>>>> We have submitted a new Internet draft to describe a Normalization =
Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The =
goal of NM deployment is to create a new incentive for video encoders to =
generate more discardable packets when using AF PHB in DIffServ. We are =
looking forward to your review comments.
>>>>=20
>>>> Best regards,
>>>> CJ
>>>>=20
>>>> -----Original Message-----
>>>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>>>> Sent: Friday, February 08, 2013 6:25 PM
>>>> To: Cheng-Jia Lai (chelai)
>>>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert =
(eckert); Fred Yip (fyip)
>>>> Subject: New Version Notification for =
draft-lai-tsvwg-normalizer-00.txt
>>>>=20
>>>>=20
>>>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>>>> has been successfully submitted by Cheng-Jia Lai and posted to the
>>>> IETF repository.
>>>>=20
>>>> Filename:	 draft-lai-tsvwg-normalizer
>>>> Revision:	 00
>>>> Title:		 Normalization Marker for AF PHB Group in =
DiffServ
>>>> Creation date:	 2013-02-08
>>>> WG ID:		 Individual Submission
>>>> Number of pages: 15
>>>> URL:             =
http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
>>>> Status:          =
http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
>>>> Htmlized:        =
http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
>>>>=20
>>>>=20
>>>> Abstract:
>>>>  This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>>>  nodes to normalize the distribution of DS codepoint (DSCP) =
markings
>>>>  for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM =
is
>>>>  useful for traffic conditioning with Active Queue Management (AQM)
>>>>  when the AF PHB group is deployed for multimedia service classes =
that
>>>>  carry video packets with advanced codec technologies.  Using NM =
can
>>>>  offer an incentive that is however missing in today's ecosystem of
>>>>  practical deployment, i.e., when congestion occurs in the AF PHB
>>>>  group, the network should allow a video codec to benefit from its
>>>>  effort of generating finer layers of intra-flow precedence (IFP) =
with
>>>>  discardable packets in a more balanced distribution.
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> The IETF Secretariat
>>>>=20
>>>=20
>>> --=20
>>> ---
>>> Toerless Eckert, eckert@cisco.com
>>> Cisco NSSTG Systems & Technology Architecture
>>> SDN: Let me play with the network, mommy!
>>>=20
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


From suhasietf@gmail.com  Mon Jul 15 10:18:28 2013
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: mmusic WG <mmusic@ietf.org>, rtcweb@ietf.org
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Subject: [rtcweb] New Draft: draft-nandakumar-mmusic-rtcweb-grouping-00.txt
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--089e013d0f806648e504e1900c34
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Hello All

   A new draft has been submitted to describe Simulcast streams within SDP
based on RFC5576 ssrc-grouping semantics.

Please have a read and let me know your thoughts

Cheers
Suhas

A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-00.txt
has been successfully submitted by Suhas Nandakumar and posted to the
IETF repository.

Filename:        draft-nandakumar-mmusic-rtcweb-grouping
Revision:        00
Title:           SSRC Group Based Simulcast Signaling
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 8
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping=
-00.txt<https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4c=
q8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fwww.=
ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-grouping-00.txt=
>
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping<htt=
ps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowN=
DR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fdatatracker.iet=
f.org%2fdoc%2fdraft-nandakumar-mmusic-rtcweb-grouping>
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<https=
://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR=
3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2ftools.ietf.org%2f=
html%2fdraft-nandakumar-mmusic-rtcweb-grouping-00>


Abstract:
   In some applications it may be necessary to send multiple media
   encodings corresponding to a media source with in independent RTP
   media streams.  This is called Simulcast.  This document discusses a
   framework for describing simulcast media streams in SDP and also
   defines semantics to express relationship amongst them.  The
   semantics defined in this document are to be used with the source
   specific grouping framework defined in the [RFC5576]

--089e013d0f806648e504e1900c34
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><span style=3D"font-family:monospace;font-size:16px">Hello All</span><=
/div><div><span style=3D"font-family:monospace;font-size:16px"><br></span><=
/div><div><span style=3D"font-family:monospace;font-size:16px">=A0 =A0A new=
 draft has been submitted to describe Simulcast streams within SDP based on=
 RFC5576 ssrc-grouping semantics.</span></div>
<div><span style=3D"font-family:monospace;font-size:16px"><br></span></div>=
<div><span style=3D"font-family:monospace;font-size:16px">Please have a rea=
d and let me know your thoughts</span></div><div><span style=3D"font-family=
:monospace;font-size:16px"><br>
</span></div><div><span style=3D"font-family:monospace;font-size:16px">Chee=
rs</span></div><div><span style=3D"font-family:monospace;font-size:16px">Su=
has</span></div><span style=3D"font-family:monospace;font-size:16px"><div><=
span style=3D"font-family:monospace;font-size:16px"><br>
</span></div>A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</span><br style=3D"font-family:monospace;font-size:16px"><span style=
=3D"font-family:monospace;font-size:16px">has been successfully submitted b=
y Suhas Nandakumar and posted to the</span><br style=3D"font-family:monospa=
ce;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">IETF repository.</span=
><br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-famil=
y:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:=
16px">Filename:=A0=A0=A0=A0=A0=A0=A0 draft-nandakumar-mmusic-rtcweb-groupin=
g</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Revision:=A0=A0=A0=A0=
=A0=A0=A0 00</span><br style=3D"font-family:monospace;font-size:16px"><span=
 style=3D"font-family:monospace;font-size:16px">Title:=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 SSRC Group Based Simulcast Signaling</span><br style=3D"font-fami=
ly:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Creation date:=A0=A0 2=
013-07-15</span><br style=3D"font-family:monospace;font-size:16px"><span st=
yle=3D"font-family:monospace;font-size:16px">Group:=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 Individual Submission</span><br style=3D"font-family:monospace;font-=
size:16px">
<span style=3D"font-family:monospace;font-size:16px">Number of pages: 8</sp=
an><br style=3D"font-family:monospace;font-size:16px"><span style=3D"font-f=
amily:monospace;font-size:16px">URL:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiM=
HIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&amp;URL=3Dhtt=
p%3a%2f%2fwww.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-g=
rouping-00.txt" target=3D"_blank" style=3D"font-family:monospace;font-size:=
16px">http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-gr=
ouping-00.txt</a><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Status:=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3D=
KlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY=
.&amp;URL=3Dhttp%3a%2f%2fdatatracker.ietf.org%2fdoc%2fdraft-nandakumar-mmus=
ic-rtcweb-grouping" target=3D"_blank" style=3D"font-family:monospace;font-s=
ize:16px">http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-gr=
ouping</a><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Htmlized:=A0=A0=A0=A0=
=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlI=
Q10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&a=
mp;URL=3Dhttp%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-nandakumar-mmusic-rtcwe=
b-grouping-00" target=3D"_blank" style=3D"font-family:monospace;font-size:1=
6px">http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<=
/a><br style=3D"font-family:monospace;font-size:16px">
<br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-family=
:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:1=
6px">Abstract:</span><br style=3D"font-family:monospace;font-size:16px"><sp=
an style=3D"font-family:monospace;font-size:16px">=A0=A0 In some applicatio=
ns it may be necessary to send multiple media</span><br style=3D"font-famil=
y:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 encodings corre=
sponding to a media source with in independent RTP</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 media streams.=A0 This is called Simulcast.=A0 This docum=
ent discusses a</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 framework for d=
escribing simulcast media streams in SDP and also</span><br style=3D"font-f=
amily:monospace;font-size:16px"><span style=3D"font-family:monospace;font-s=
ize:16px">=A0=A0 defines semantics to express relationship amongst them.=A0=
 The</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 semantics defin=
ed in this document are to be used with the source</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 specific grouping framework defined in the [RFC5576]</spa=
n>

--089e013d0f806648e504e1900c34--

From suhasietf@gmail.com  Mon Jul 15 11:34:49 2013
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Date: Mon, 15 Jul 2013 11:34:46 -0700
Message-ID: <CAMRcRGRwMKyrkYc1=vhx3DxQXvAboo8h1TpL2fDd0Xfjndbhvg@mail.gmail.com>
From: Suhas Nandakumar <suhasietf@gmail.com>
To: mmusic WG <mmusic@ietf.org>, rtcweb@ietf.org
Content-Type: multipart/alternative; boundary=047d7bf10aea6b3ecd04e1911dd1
Subject: Re: [rtcweb] New Draft: draft-nandakumar-mmusic-rtcweb-grouping-00.txt
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--047d7bf10aea6b3ecd04e1911dd1
Content-Type: text/plain; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

The links in my earlier email seems to have mail.cisco.com redirects and
hence I am including the direct links for the same below

Filename:        draft-nandakumar-mmusic-rtcweb-grouping
Revision:         00
Title:              SSRC Group Based Simulcast Signaling
Creation date:   2013-07-15
Group:            Individual Submission
Number of pages: 8
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping=
-00.txt
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00

Sorry for the inconvenience

Cheers
Suhas
PS: Thanks Martin and Ethan for catching this.


On Mon, Jul 15, 2013 at 10:18 AM, Suhas Nandakumar <suhasietf@gmail.com>wro=
te:

> Hello All
>
>    A new draft has been submitted to describe Simulcast streams within SD=
P
> based on RFC5576 ssrc-grouping semantics.
>
> Please have a read and let me know your thoughts
>
> Cheers
> Suhas
>
> A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-00.txt
> has been successfully submitted by Suhas Nandakumar and posted to the
> IETF repository.
>
> Filename:        draft-nandakumar-mmusic-rtcweb-grouping
> Revision:        00
> Title:           SSRC Group Based Simulcast Signaling
> Creation date:   2013-07-15
> Group:           Individual Submission
> Number of pages: 8
> URL:
> http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-groupi=
ng-00.txt<https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX=
4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fww=
w.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-grouping-00.t=
xt>
> Status:
> http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping<h=
ttps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUo=
wNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fdatatracker.i=
etf.org%2fdoc%2fdraft-nandakumar-mmusic-rtcweb-grouping>
> Htmlized:
> http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<htt=
ps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowN=
DR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2ftools.ietf.org%=
2fhtml%2fdraft-nandakumar-mmusic-rtcweb-grouping-00>
>
>
> Abstract:
>    In some applications it may be necessary to send multiple media
>    encodings corresponding to a media source with in independent RTP
>    media streams.  This is called Simulcast.  This document discusses a
>    framework for describing simulcast media streams in SDP and also
>    defines semantics to express relationship amongst them.  The
>    semantics defined in this document are to be used with the source
>    specific grouping framework defined in the [RFC5576]

--047d7bf10aea6b3ecd04e1911dd1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><br></div><div><br></div>The links in my earlier email seems to have <=
a href=3D"http://mail.cisco.com" target=3D"_blank">mail.cisco.com</a> redir=
ects and hence I am including the direct links for the same below<div><br><=
/div>
<div><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-s=
ize:13px;background-color:rgb(255,255,255)">Filename: =A0 =A0 =A0 =A0draft-=
nandakumar-mmusic-</span><span style=3D"color:rgb(34,34,34);font-family:ari=
al,sans-serif;font-size:13px;background-color:rgb(255,255,255)">rtcweb-grou=
ping</span><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fo=
nt-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Revision: =A0 =A0 =A0 =A0 00</span><=
br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px=
;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Title: =A0 =A0 =A0 =A0 =A0 =A0 =A0SS=
RC Group Based Simulcast Signaling</span><br style=3D"color:rgb(34,34,34);f=
ont-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255=
)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Creation date: =A0=A0</span><span cl=
ass=3D"aBn" tabindex=3D"0" style=3D"border-bottom-width:1px;border-bottom-s=
tyle:dashed;border-bottom-color:rgb(204,204,204);color:rgb(34,34,34);font-f=
amily:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><s=
pan class=3D"aQJ" style>2013-07-15</span></span><br style=3D"color:rgb(34,3=
4,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,=
255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Group: =A0 =A0 =A0 =A0 =A0 =A0Indivi=
dual Submission</span><br style=3D"color:rgb(34,34,34);font-family:arial,sa=
ns-serif;font-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Number of pages: 8</span><br style=
=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;backgro=
und-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">URL: =A0 =A0 =A0 =A0 =A0 =A0=A0</spa=
n><a href=3D"http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rt=
cweb-grouping-00.txt" target=3D"_blank" style=3D"color:rgb(17,85,204);font-=
family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">h=
ttp://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</a><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fon=
t-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Status: =A0 =A0 =A0 =A0 =A0</span><a=
 href=3D"http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-gro=
uping" target=3D"_blank" style=3D"color:rgb(17,85,204);font-family:arial,sa=
ns-serif;font-size:13px;background-color:rgb(255,255,255)">http://datatrack=
er.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping</a><br style=3D"col=
or:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;background-col=
or:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Htmlized: =A0 =A0 =A0 =A0</span><a h=
ref=3D"http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-0=
0" target=3D"_blank" style=3D"color:rgb(17,85,204);font-family:arial,sans-s=
erif;font-size:13px;background-color:rgb(255,255,255)">http://tools.ietf.or=
g/html/draft-nandakumar-mmusic-rtcweb-grouping-00</a><br style=3D"color:rgb=
(34,34,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb=
(255,255,255)">
</div>
<div><br></div><div>Sorry for the inconvenience</div><div><br></div><div>Ch=
eers</div><div>Suhas<br>PS: Thanks Martin and Ethan for catching this.</div=
><div><br></div><div><br><div class=3D"gmail_quote">On Mon, Jul 15, 2013 at=
 10:18 AM, Suhas Nandakumar <span dir=3D"ltr">&lt;<a href=3D"mailto:suhasie=
tf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>&gt;</span> wrote:<b=
r>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><span style=3D"font-family:monospace;fo=
nt-size:16px">Hello All</span></div><div><span style=3D"font-family:monospa=
ce;font-size:16px"><br>

</span></div><div><span style=3D"font-family:monospace;font-size:16px">=A0 =
=A0A new draft has been submitted to describe Simulcast streams within SDP =
based on RFC5576 ssrc-grouping semantics.</span></div>
<div><span style=3D"font-family:monospace;font-size:16px"><br></span></div>=
<div><span style=3D"font-family:monospace;font-size:16px">Please have a rea=
d and let me know your thoughts</span></div><div><span style=3D"font-family=
:monospace;font-size:16px"><br>


</span></div><div><span style=3D"font-family:monospace;font-size:16px">Chee=
rs</span></div><div><span style=3D"font-family:monospace;font-size:16px">Su=
has</span></div><span style=3D"font-family:monospace;font-size:16px"><div><=
span style=3D"font-family:monospace;font-size:16px"><br>


</span></div>A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</span><br style=3D"font-family:monospace;font-size:16px"><span style=
=3D"font-family:monospace;font-size:16px">has been successfully submitted b=
y Suhas Nandakumar and posted to the</span><br style=3D"font-family:monospa=
ce;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">IETF repository.</span=
><br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-famil=
y:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:=
16px">Filename:=A0=A0=A0=A0=A0=A0=A0 draft-nandakumar-mmusic-rtcweb-groupin=
g</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Revision:=A0=A0=A0=A0=
=A0=A0=A0 00</span><br style=3D"font-family:monospace;font-size:16px"><span=
 style=3D"font-family:monospace;font-size:16px">Title:=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 SSRC Group Based Simulcast Signaling</span><br style=3D"font-fami=
ly:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Creation date:=A0=A0 2=
013-07-15</span><br style=3D"font-family:monospace;font-size:16px"><span st=
yle=3D"font-family:monospace;font-size:16px">Group:=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 Individual Submission</span><br style=3D"font-family:monospace;font-=
size:16px">


<span style=3D"font-family:monospace;font-size:16px">Number of pages: 8</sp=
an><br style=3D"font-family:monospace;font-size:16px"><span style=3D"font-f=
amily:monospace;font-size:16px">URL:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiM=
HIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&amp;URL=3Dhtt=
p%3a%2f%2fwww.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-g=
rouping-00.txt" style=3D"font-family:monospace;font-size:16px" target=3D"_b=
lank">http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-gr=
ouping-00.txt</a><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Status:=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3D=
KlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY=
.&amp;URL=3Dhttp%3a%2f%2fdatatracker.ietf.org%2fdoc%2fdraft-nandakumar-mmus=
ic-rtcweb-grouping" style=3D"font-family:monospace;font-size:16px" target=
=3D"_blank">http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-=
grouping</a><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Htmlized:=A0=A0=A0=A0=
=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlI=
Q10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&a=
mp;URL=3Dhttp%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-nandakumar-mmusic-rtcwe=
b-grouping-00" style=3D"font-family:monospace;font-size:16px" target=3D"_bl=
ank">http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<=
/a><br style=3D"font-family:monospace;font-size:16px">


<br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-family=
:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:1=
6px">Abstract:</span><br style=3D"font-family:monospace;font-size:16px"><sp=
an style=3D"font-family:monospace;font-size:16px">=A0=A0 In some applicatio=
ns it may be necessary to send multiple media</span><br style=3D"font-famil=
y:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 encodings corre=
sponding to a media source with in independent RTP</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 media streams.=A0 This is called Simulcast.=A0 This docum=
ent discusses a</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 framework for d=
escribing simulcast media streams in SDP and also</span><br style=3D"font-f=
amily:monospace;font-size:16px"><span style=3D"font-family:monospace;font-s=
ize:16px">=A0=A0 defines semantics to express relationship amongst them.=A0=
 The</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 semantics defin=
ed in this document are to be used with the source</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 specific grouping framework defined in the [RFC5576]</spa=
n>
</blockquote></div><br></div>

--047d7bf10aea6b3ecd04e1911dd1--

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Date: Mon, 15 Jul 2013 18:44:51 +0000
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Subject: [rtcweb] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Forwarding it to rtcweb..

-----Original Message-----
From: Muthu Arul Mozhi Perumal (mperumal)=20
Sent: Tuesday, July 16, 2013 12:13 AM
To: behave@ietf.org
Subject: FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt

The text and the algorithm in the draft are significantly simplified in thi=
s updated version.=20

Comments welcome..

- Authors

-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of internet-drafts@ietf.org
Sent: Monday, July 15, 2013 11:08 PM
To: i-d-announce@ietf.org
Subject: I-D Action: draft-muthu-behave-consent-freshness-04.txt

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.


	Title           : STUN Usage for Consent Freshness
	Author(s)       : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
	Filename        : draft-muthu-behave-consent-freshness-04.txt
	Pages           : 7
	Date            : 2013-07-15

Abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.  A related problem
   is session liveness.  WebRTC application may want to detect
   connection failure and take appropriate action.

   This document describes how a WebRTC browser can verify peer consent
   to continue sending traffic and detect connection failure.

The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-muthu-behave-consent-freshness-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-muthu-behave-consent-freshness-04


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From adam@nostrum.com  Mon Jul 15 12:05:35 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] Unified Plan for SDP Handling
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[Cross-posting to RTCWEB; follow-ups to MMUSIC, please]

After significant work, Justin, Martin and I have managed to produce a 
compromise plan that provides a high degree of interoperability with 
existing devices (and future non-WebRTC devices) while not being 
excessively onerous for WebRTC implementations or applications that use 
them. It's been a tricky balancing act, but I think we've found a good 
mix between the two that can form a solid basis for the working group to 
move forward.

Rather than summarize the key points of the document in this email, I 
direct interested parties to section 2 of the document, which summarizes 
the key aspects of the plan in eight relatively concise bullet points.

I apologize for the late publication date of this document -- there's 
actually been a lot more work put into coming up with a unified draft 
than I originally anticipated, and the production of this document took 
at least two weeks longer than I expected it to.

Note that this document is intended to be a plan for the work to be done 
in this area, and not a specification in itself. The intention is that 
its contents are used as the basis for work in several other drafts -- 
some new, some not -- that form the corpus of work necessary for RTCWEB 
(and potentially CLUE) to move forward. Except in rare cases, the 
document does not attempt to explicitly call out venues or documents for 
such work, as we (or, at the very least, I) anticipate guidance from the 
various working group chairs to assist in such decisions.

Comments prior to Berlin would be very helpful, although this will 
clearly be a point of significant discussion at the face-to-face meeting.

Document link:

http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt

/a

From internet-drafts@ietf.org  Mon Jul 15 12:24:03 2013
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : RTCWeb Data Channels
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-05.txt
	Pages           : 13
	Date            : 2013-07-15

Abstract:
   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document specifies the non-media data transport aspects of the WebRTC
   framework.  It provides an architectural overview of how the Stream
   Control Transmission Protocol (SCTP) is used in the WebRTC context as
   a generic transport service allowing Web Browser to exchange generic
   data from peer to peer.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-channel-05


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : WebRTC Data Channel Protocol
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-protocol-00.txt
	Pages           : 11
	Date            : 2013-07-15

Abstract:
   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocols to support for direct interactive rich
   communication using audio, video, and data between two peers' web-
   browsers.  This document specifies an actual (minor) protocol for how
   the JS-layer DataChannel objects provide the data channels between
   the peers.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Date: Mon, 15 Jul 2013 13:46:21 -0700
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Ted Hardie <ted.ietf@gmail.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Draft agenda for IETF 87
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Hello Chairs

  I was wondering if we could add "RTCWeb SDP Examples" draft to the Agenda
that shows several SDP examples for the most common WebRTC Use-cases.

The draft location is :
http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp-02


Thanks
Suhas

On Thu, Jul 11, 2013 at 9:51 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Greetings,
>
> Below is an initial draft agenda for the upcoming meeting.   Since we have
> not yet reached the draft deadline (which is the 15th), there may be new
> drafts or updates that result in changes.  We did already receive requests
> for NAT/Firewall traversal discussion, and the chairs will be working with
> the document authors to get them considered in the appropriate groups.
>
> As folks have probably noticed, we are meeting Thursday and Friday, after
> the MMUSIC sessions are complete (they are Tuesday and Wednesday). This
> should allow us to discuss the results on our first day.
>
> Please send feedback or change proposals to the list.
>
> thanks,
>
> Ted and Cullen
>
> Day 1:
>
> Should SDES be part of  WebRTC security practice and, if so, how?
> Presentations: 30 minutes
> Discussion:  40 minutes
>
> Post-Plan A/Plan B MMUSIC discussion of impact to RTCWEB documents
> Presentation: 30 minutes
> Discussion: 30 minutes
>
> Security document updates
> Presentation: 10 minutes
> Discussion: 10 minutes
>
> Day 2:
>
> Chair Discussion:  10 minutes
>
> Use Case Requirements updates:
> Issues list presentation: 20 minutes
> Discussion: 20 minutes
>
> Data channel:
> Issues list presentation:  45 minutes
> Discussion: 45 minutes
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c3505efde2a204e192f327
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Hello Chairs<div><br></div><div>=A0 I was wondering if we could add &quot;R=
TCWeb SDP Examples&quot; draft to the Agenda that shows several SDP example=
s for the most common WebRTC Use-cases.</div><div><br></div><div>The draft =
location is :</div>
<div><a href=3D"http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp-02">=
http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp-02</a></div><div><br=
></div><div><br></div><div>Thanks</div><div>Suhas</div><div><br><div class=
=3D"gmail_quote">
On Thu, Jul 11, 2013 at 9:51 AM, Ted Hardie <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex">
Greetings,<br><br>Below is an initial draft agenda for the upcoming meeting=
.=A0=A0 Since we have not yet reached the draft deadline (which is the 15th=
), there may be new drafts or updates that result in changes.=A0 We did alr=
eady receive requests for NAT/Firewall traversal discussion, and the chairs=
 will be working with the document authors to get them considered in the ap=
propriate groups.<br>

<br>As folks have probably noticed, we are meeting Thursday and Friday,=20
after the MMUSIC sessions are complete (they are Tuesday and Wednesday). Th=
is should allow us to discuss the results on our first day. <br><br>Please =
send feedback or change proposals to the list.<br><br>thanks,<br><br>Ted an=
d Cullen<br>

<br>Day 1:<br><br>
Should SDES be part of=A0 WebRTC security practice and, if so, how?<br>Pres=
entations: 30 minutes<br>Discussion:=A0 40 minutes<br><br>Post-Plan A/Plan =
B MMUSIC discussion of impact to RTCWEB documents<br>Presentation: 30 minut=
es<br>


Discussion: 30 minutes<br><br>Security document updates<br>Presentation: 10=
 minutes<br>Discussion: 10 minutes<br><br>Day 2:<br><br>Chair Discussion:=
=A0 10 minutes<br><br>Use Case Requirements updates:<br>Issues list present=
ation: 20 minutes<br>

Discussion: 20 minutes<br>
<br>Data channel:<br>Issues list presentation:=A0 45 minutes<br>Discussion:=
 45 minutes<br><br><br>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c3505efde2a204e192f327--

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--089e013d0f800b62fb04e19302da
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Hello All

  An updated version of the draft showcasing SDP examples for the most
common WebRTC use-cases has be submitted.

I would recommend PDF version since that shows embedded images.

Please let us know your thoughts.

Cheers
Suhas

Filename:        draft-nandakumar-rtcweb-sdp
Revision:        02
Title:           SDP for the WebRTC
Creation date:   2013-07-12
Group:           Individual Submission
Number of pages: 64
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-rtcweb-sdp-02.txt
Status:          http://datatracker.ietf.org/doc/draft-nandakumar-rtcweb-sdp
Htmlized:        http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp-02
Diff:
http://www.ietf.org/rfcdiff?url2=draft-nandakumar-rtcweb-sdp-02

Abstract:
   The Web Real-Time Communication (WebRTC) [WEBRTC] working group is
   charged to provide protocol support for direct interactive rich
   communication using audio,video and data between two peers' web
   browsers.  With in the WebRTC framework, Session Description protocol
   (SDP) [RFC4566] is used for negotiating session capabilities between
   the peers.  Such a negotiataion happens based on the SDP Offer/Answer
   exchange mechanism described in the RFC 3264 [RFC3264].

   This document serves a introductory purpose in describing the role of
   SDP for the most common WebRTC use-cases.

   This SDP examples provided in this document is still a work in
   progress, but aims to align closest to the evolving standards.




The IETF Secretariat

--089e013d0f800b62fb04e19302da
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Hello All<div><br></div><div>=A0 An updated version of the draft showcasing=
 SDP examples for the most common WebRTC use-cases has be submitted.</div><=
div><br></div><div>I would recommend PDF version since that shows embedded =
images.</div>
<div><br></div><div>Please let us know your thoughts.</div><div><br></div><=
div>Cheers</div><div>Suhas<br><div class=3D"gmail_quote"><br>Filename: =A0 =
=A0 =A0 =A0draft-nandakumar-rtcweb-sdp<br>
Revision: =A0 =A0 =A0 =A002<br>
Title: =A0 =A0 =A0 =A0 =A0 SDP for the WebRTC<br>
Creation date: =A0 2013-07-12<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 64<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-nandakumar-rtcweb-sdp-02.txt" target=3D"_blank">http://www.ietf.org/=
internet-drafts/draft-nandakumar-rtcweb-sdp-02.txt</a><br>
Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-nandakumar-rtcweb-sdp" target=3D"_blank">http://datatracker.ietf.org/doc/d=
raft-nandakumar-rtcweb-sdp</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-nandak=
umar-rtcweb-sdp-02" target=3D"_blank">http://tools.ietf.org/html/draft-nand=
akumar-rtcweb-sdp-02</a><br>
Diff: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/rfcdiff?url2=3D=
draft-nandakumar-rtcweb-sdp-02" target=3D"_blank">http://www.ietf.org/rfcdi=
ff?url2=3Ddraft-nandakumar-rtcweb-sdp-02</a><br>
<br>
Abstract:<br>
=A0 =A0The Web Real-Time Communication (WebRTC) [WEBRTC] working group is<b=
r>
=A0 =A0charged to provide protocol support for direct interactive rich<br>
=A0 =A0communication using audio,video and data between two peers&#39; web<=
br>
=A0 =A0browsers. =A0With in the WebRTC framework, Session Description proto=
col<br>
=A0 =A0(SDP) [RFC4566] is used for negotiating session capabilities between=
<br>
=A0 =A0the peers. =A0Such a negotiataion happens based on the SDP Offer/Ans=
wer<br>
=A0 =A0exchange mechanism described in the RFC 3264 [RFC3264].<br>
<br>
=A0 =A0This document serves a introductory purpose in describing the role o=
f<br>
=A0 =A0SDP for the most common WebRTC use-cases.<br>
<br>
=A0 =A0This SDP examples provided in this document is still a work in<br>
=A0 =A0progress, but aims to align closest to the evolving standards.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--089e013d0f800b62fb04e19302da--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 15 Jul 2013 17:52:50 -0400
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Subject: [rtcweb] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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I have changed the WG for this draft from RTCWEB to BEHAVE. Many, but not
all of the comments I received on the RTCWEB mailing list have been
addressed.

BEHAVE chairs, I would like 10 minutes of agenda time to discuss this draft.

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Jul 15, 2013 at 5:49 PM
Subject: New Version Notification for draft-uberti-behave-turn-rest-00.txt
To: Justin Uberti <justin@uberti.name>



A new version of I-D, draft-uberti-behave-turn-rest-00.txt
has been successfully submitted by Justin Uberti and posted to the
IETF repository.

Filename:        draft-uberti-behave-turn-rest
Revision:        00
Title:           A REST API For Access To TURN Services
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 8
URL:
http://www.ietf.org/internet-drafts/draft-uberti-behave-turn-rest-00.txt
Status:
http://datatracker.ietf.org/doc/draft-uberti-behave-turn-rest
Htmlized:        http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00


Abstract:
   This document describes a proposed standard REST API for obtaining
   access to TURN services via ephemeral (i.e. time-limited)
   credentials.  These credentials are vended by a web service over
   HTTP, and then supplied to and checked by a TURN server using the
   standard TURN protocol.  The usage of ephemeral credentials ensures
   that access to the TURN server can be controlled even if the
   credentials can be discovered by the user, as is the case in WebRTC
   where TURN credentials must be specified in Javascript.




The IETF Secretariat

--047d7ba97b94f39ea604e193e2dd
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<div dir=3D"ltr">I have changed the WG for this draft from RTCWEB to BEHAVE=
. Many, but not all of the comments I received on the RTCWEB mailing list h=
ave been addressed.<br><div class=3D"gmail_quote"><br><div dir=3D"ltr">BEHA=
VE chairs, I would like 10 minutes of agenda time to discuss this draft.<br=
>

<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</=
a>&gt;</span><br>



Date: Mon, Jul 15, 2013 at 5:49 PM<br>Subject: New Version Notification for=
 draft-uberti-behave-turn-rest-00.txt<br>To: Justin Uberti &lt;<a href=3D"m=
ailto:justin@uberti.name" target=3D"_blank">justin@uberti.name</a>&gt;<br>

<br><br><br>
A new version of I-D, draft-uberti-behave-turn-rest-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Filename: =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-uberti-behave-turn-rest<br>
Revision: =C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 A REST API For Access To TURN Ser=
vices<br>
Creation date: =C2=A0 2013-07-15<br>
Group: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Number of pages: 8<br>
URL: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.o=
rg/internet-drafts/draft-uberti-behave-turn-rest-00.txt" target=3D"_blank">=
http://www.ietf.org/internet-drafts/draft-uberti-behave-turn-rest-00.txt</a=
><br>
Status: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://datatracker.iet=
f.org/doc/draft-uberti-behave-turn-rest" target=3D"_blank">http://datatrack=
er.ietf.org/doc/draft-uberti-behave-turn-rest</a><br>
Htmlized: =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/=
draft-uberti-behave-turn-rest-00" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-uberti-behave-turn-rest-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes a proposed standard REST API for obtai=
ning<br>
=C2=A0 =C2=A0access to TURN services via ephemeral (i.e. time-limited)<br>
=C2=A0 =C2=A0credentials. =C2=A0These credentials are vended by a web servi=
ce over<br>
=C2=A0 =C2=A0HTTP, and then supplied to and checked by a TURN server using =
the<br>
=C2=A0 =C2=A0standard TURN protocol. =C2=A0The usage of ephemeral credentia=
ls ensures<br>
=C2=A0 =C2=A0that access to the TURN server can be controlled even if the<b=
r>
=C2=A0 =C2=A0credentials can be discovered by the user, as is the case in W=
ebRTC<br>
=C2=A0 =C2=A0where TURN credentials must be specified in Javascript.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>
</div><br></div>

--047d7ba97b94f39ea604e193e2dd--

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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-07.txt
	Pages           : 61
	Date            : 2013-07-15

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-07


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 15 Jul 2013 17:43:38 -0700
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Hey Randell, can you give a short summary of the changes, if any, that I
would or other implementors would need to know about?


On Mon, Jul 15, 2013 at 1:31 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Data Channel Protocol
>         Author(s)       : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-00.txt
>         Pages           : 11
>         Date            : 2013-07-15
>
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies an actual (minor) protocol for how
>    the JS-layer DataChannel objects provide the data channels between
>    the peers.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--e89a8ff2488104dbb904e196472b
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<div dir=3D"ltr">Hey Randell, can you give a short summary of the changes, =
if any, that I would or other implementors would need to know about?</div><=
div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Jul 15=
, 2013 at 1:31 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts=
@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:=
<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : WebR=
TC Data Channel Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author(s) =C2=A0 =C2=A0 =C2=A0 : Randell Jesup<=
br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Salvatore Loreto<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Michael Tuexen<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-iet=
f-rtcweb-data-protocol-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 11<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2013-07-15<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0The Web Real-Time Communication (WebRTC) working group is char=
ged to<br>
=C2=A0 =C2=A0provide protocols to support for direct interactive rich<br>
=C2=A0 =C2=A0communication using audio, video, and data between two peers&#=
39; web-<br>
=C2=A0 =C2=A0browsers. =C2=A0This document specifies an actual (minor) prot=
ocol for how<br>
=C2=A0 =C2=A0the JS-layer DataChannel objects provide the data channels bet=
ween<br>
=C2=A0 =C2=A0the peers.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol=
" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data=
-protocol</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00" t=
arget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol=
-00</a><br>
<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--e89a8ff2488104dbb904e196472b--

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Hey Randell, can you give a short summary of the changes, if any, that I
would or other implementors would need to know about?


On Mon, Jul 15, 2013 at 12:23 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : RTCWeb Data Channels
>         Author(s)       : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-channel-05.txt
>         Pages           : 13
>         Date            : 2013-07-15
>
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged to
>    provide protocol support for direct interactive rich communication
>    using audio, video, and data between two peers' web-browsers.  This
>    document specifies the non-media data transport aspects of the WebRTC
>    framework.  It provides an architectural overview of how the Stream
>    Control Transmission Protocol (SCTP) is used in the WebRTC context as
>    a generic transport service allowing Web Browser to exchange generic
>    data from peer to peer.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-channel-05
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Hey Randell, can you give a short summary of the changes, =
if any, that I would or other implementors would need to know about?<br></d=
iv><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Ju=
l 15, 2013 at 12:23 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-d=
rafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : RTCW=
eb Data Channels<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author(s) =C2=A0 =C2=A0 =C2=A0 : Randell Jesup<=
br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Salvatore Loreto<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Michael Tuexen<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-iet=
f-rtcweb-data-channel-05.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 13<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2013-07-15<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0The Web Real-Time Communication (WebRTC) working group is char=
ged to<br>
=C2=A0 =C2=A0provide protocol support for direct interactive rich communica=
tion<br>
=C2=A0 =C2=A0using audio, video, and data between two peers&#39; web-browse=
rs. =C2=A0This<br>
=C2=A0 =C2=A0document specifies the non-media data transport aspects of the=
 WebRTC<br>
=C2=A0 =C2=A0framework. =C2=A0It provides an architectural overview of how =
the Stream<br>
=C2=A0 =C2=A0Control Transmission Protocol (SCTP) is used in the WebRTC con=
text as<br>
=C2=A0 =C2=A0a generic transport service allowing Web Browser to exchange g=
eneric<br>
=C2=A0 =C2=A0data from peer to peer.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel"=
 target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-=
channel</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05" ta=
rget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-0=
5</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-channe=
l-05" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcwe=
b-data-channel-05</a><br>
<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--001a11c39d185caa7004e19647c7--

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Simon Perreault <simon.perreault@viagenie.ca>, "behave@ietf.org" <behave@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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[Added rtcweb since I am not sure if everyone involved there are following =
this discussion in behave]

Thanks for the review. See inline..=20

|-----Original Message-----
|From: behave-bounces@ietf.org [mailto:behave-bounces@ietf.org] On Behalf O=
f Simon Perreault
|Sent: Tuesday, July 16, 2013 3:05 PM
|To: behave@ietf.org
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt
|
|Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|> The text and the algorithm in the draft are significantly simplified in =
this updated version.
|>
|> Comments welcome..
|
|MUCH better introduction. Now I feel like I understand the need exactly.
|
|The "Design Considerations" section is still very confusing to me.
|
|>    Though ICE specifies STUN Binding indications to be used for
|>    keepalives, it requires that an agent be prepared to receive
|>    connectivity check as well.  If a connectivity check is received, a
|>    response is generated, but there is no impact on ICE processing, as
|>    described in section 10 of [RFC5245].
|
|...so? Why is "an impact on ICE processing" necessary?

Meant to stress these Binding request/response doesn't trigger an ICE resta=
rt..

|
|>    While a WebRTC browser could verify whether the peer continues to
|>    send SRTCP reports before sending traffic to the peer, the usage of
|>    SRTCP together with Security Descriptions [RFC4568] requires exposing
|>    the media keys to the JavaScript and renders SRTCP unsuitable for
|>    consent freshness.
|
|Why does it "require exposing the media keys to the JavaScript"? Is this
|because of a law of nature, or is it because of the way the JavaScript
|API is being designed? Could the JS API be changed to accommodate
|SRTCP+SDES?

That's how the API construct looks like today -- the JavaScript would get a=
n SDP blob from the browser containing the crypto keys used for SDES-SRTP. =
Of course, the browser could hide those keys by putting a "****" in SDP -:)=
. SRTP itself is still being debated in RTCWEB, so nothing is final, yet.

|
|>    For consent, calculating the SHA1 HMAC is necessary for MESSAGE-
|>    INTEGRITY which is computationally expensive.
|
|You need to qualify "expensive". It's certainly not expensive in all cases=
.
|
|>    Security analysis
|>    concluded that the STUN 96 bits transaction ID is sufficient for
|>    consent, without needing MESSAGE-INTEGRITY.
|
|What analysis? You would need to explain it in detail. But if that's not
|part of your suggested solution, just remove the sentence.

Agree it could be elaborated. The intention was to say the foll:

   STUN requires the 96 bits transaction ID to be uniformly and randomly
   chosen from the interval 0 .. 2**96-1, and be cryptographically
   random. This should suffice to prevent off path attacks on consent=20
   freshness, so MESSAGE-INTEGRITY is not necessary from a security
   perspective.

However, MESSAGE-INTEGRITY is still used to maintain backward compatibility=
=20
with legacy ICE/ICE-lite implementations.

|
|Now on to section "4. Solution Overview"...
|
|>    Every Tc seconds, the WebRTC browser sends a STUN Binding Request to
|>    the peer.  This request MUST use a new, cryptographically random
|>    Transaction ID [RFC4086], and is formatted as for an ICE connectivity
|>    check [RFC5245].  A valid STUN Binding Response is also formatted as
|>    for an ICE connectivity check [RFC5245].  The STUN Binding Request
|>    and STUN Binding Response are validated as for an ICE connectivity
|>    check [RFC5245].
|
|Couldn't this whole paragraph be simplified to "Every Tc seconds, the
|WebRTC browser sends an ICE connectivity check."? Is there anything new
|here besides the "every Tc" thing?

The difference from the ICE connectivity check is that there is no exponent=
ial back off for retransmissions.
=20
|
|>    Liveness timer: If no packets have been received on the local port in
|>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
|>    connectivity has been lost.  The JavaScript application will use this
|>    notification however it desires (e.g., cease transmitting to the
|>    remote peer, provide a notification to the user, etc.).
|
|This seems to me like it will not fulfill the goal set in the abstract:
|"to ensure that a malicious JavaScript cannot use the browser as a
|platform for launching attacks". If the JavaScript is free to ignore the
|notification from the browser, then it has no security benefits. If you
|want to reach that goal, the browser needs to forcefully stop transmitting=
.

That goal is fulfilled by the consent checks -- the browser would stop tran=
smitting everything on that candidate pair, including liveness checks, if t=
here is a consent failure.

|
|>    When not actively sending traffic on a nominated candidate pair,
|>    performing consent freshness does not serve any purpose from a
|>    security perspective.
|
|I don't understand what this means. Why is the "security perspective"
|important here? Aren't we concerned about keepalives?

You mean one could use keepalives (Binding indications) for launching attac=
ks, so consent freshness would be required for sending them as well?

|
|>    In ICE [RFC5245] the STUN request/response are protected with
|>    MESSAGE-INTEGRITY, using an ephemeral username and password exchanged
|>    in the SDP ice-ufrag and ice-pwd attributes.  This prevents ICE from
|>    accidentally connecting to an in-intended peer, in that ICE will only
|>    connect to a peer that also knows the same username and password
|>    (exchanged in call signaling).  Once that connection to the remote
|>    peer has been established with ICE, the consent to continue sending
|>    traffic does not benefit from re-asserting that same username and
|>    password, so long as the senders and receiver's IP addresses remain
|>    the same (as they usually do).
|
|I had not understood from the text that there was no auth stuff in the
|request. The text said "This request [...] is formatted as for an ICE
|connectivity check [RFC5245]." I thought that included auth stuff.
|
|The explanation why no auth stuff in the request is acceptable should be
|moved to section 3.

Agree, it could be moved under design considerations.

Muthu=20

|
|Simon
|_______________________________________________
|Behave mailing list
|Behave@ietf.org
|https://www.ietf.org/mailman/listinfo/behave

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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Le 2013-07-16 12:49, Muthu Arul Mozhi Perumal (mperumal) a écrit :
> |>    Though ICE specifies STUN Binding indications to be used for
> |>    keepalives, it requires that an agent be prepared to receive
> |>    connectivity check as well.  If a connectivity check is received, a
> |>    response is generated, but there is no impact on ICE processing, as
> |>    described in section 10 of [RFC5245].
> |
> |...so? Why is "an impact on ICE processing" necessary?
> 
> Meant to stress these Binding request/response doesn't trigger an ICE restart..

Ok, then s/but/and/.

> |>    While a WebRTC browser could verify whether the peer continues to
> |>    send SRTCP reports before sending traffic to the peer, the usage of
> |>    SRTCP together with Security Descriptions [RFC4568] requires exposing
> |>    the media keys to the JavaScript and renders SRTCP unsuitable for
> |>    consent freshness.
> |
> |Why does it "require exposing the media keys to the JavaScript"? Is this
> |because of a law of nature, or is it because of the way the JavaScript
> |API is being designed? Could the JS API be changed to accommodate
> |SRTCP+SDES?
> 
> That's how the API construct looks like today -- the JavaScript would get an SDP blob from the browser containing the crypto keys used for SDES-SRTP. Of course, the browser could hide those keys by putting a "****" in SDP -:). SRTP itself is still being debated in RTCWEB, so nothing is final, yet.

Ok, then say so in the draft.

> |>    Security analysis
> |>    concluded that the STUN 96 bits transaction ID is sufficient for
> |>    consent, without needing MESSAGE-INTEGRITY.
> |
> |What analysis? You would need to explain it in detail. But if that's not
> |part of your suggested solution, just remove the sentence.
> 
> Agree it could be elaborated. The intention was to say the foll:
> 
>    STUN requires the 96 bits transaction ID to be uniformly and randomly
>    chosen from the interval 0 .. 2**96-1, and be cryptographically
>    random. This should suffice to prevent off path attacks on consent 
>    freshness, so MESSAGE-INTEGRITY is not necessary from a security
>    perspective.
> 
> However, MESSAGE-INTEGRITY is still used to maintain backward compatibility 
> with legacy ICE/ICE-lite implementations.

Ok, then say so in the draft.

> |Now on to section "4. Solution Overview"...
> |
> |>    Every Tc seconds, the WebRTC browser sends a STUN Binding Request to
> |>    the peer.  This request MUST use a new, cryptographically random
> |>    Transaction ID [RFC4086], and is formatted as for an ICE connectivity
> |>    check [RFC5245].  A valid STUN Binding Response is also formatted as
> |>    for an ICE connectivity check [RFC5245].  The STUN Binding Request
> |>    and STUN Binding Response are validated as for an ICE connectivity
> |>    check [RFC5245].
> |
> |Couldn't this whole paragraph be simplified to "Every Tc seconds, the
> |WebRTC browser sends an ICE connectivity check."? Is there anything new
> |here besides the "every Tc" thing?
> 
> The difference from the ICE connectivity check is that there is no exponential back off for retransmissions.

Ok, then say so in the draft.

> |>    Liveness timer: If no packets have been received on the local port in
> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
> |>    connectivity has been lost.  The JavaScript application will use this
> |>    notification however it desires (e.g., cease transmitting to the
> |>    remote peer, provide a notification to the user, etc.).
> |
> |This seems to me like it will not fulfill the goal set in the abstract:
> |"to ensure that a malicious JavaScript cannot use the browser as a
> |platform for launching attacks". If the JavaScript is free to ignore the
> |notification from the browser, then it has no security benefits. If you
> |want to reach that goal, the browser needs to forcefully stop transmitting.
> 
> That goal is fulfilled by the consent checks -- the browser would stop transmitting everything on that candidate pair, including liveness checks, if there is a consent failure.

That's not what the draft says. It says that the browser "notifies" the
JS app. It needs to say that the browser MUST stop sending.

> |>    When not actively sending traffic on a nominated candidate pair,
> |>    performing consent freshness does not serve any purpose from a
> |>    security perspective.
> |
> |I don't understand what this means. Why is the "security perspective"
> |important here? Aren't we concerned about keepalives?
> 
> You mean one could use keepalives (Binding indications) for launching attacks, so consent freshness would be required for sending them as well?

No.

This is a section about keepalives. I just don't understand this
sentence, and I don't understand why it talks about security.

Simon

From mperumal@cisco.com  Tue Jul 16 05:43:32 2013
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Simon Perreault <simon.perreault@viagenie.ca>
Thread-Topic: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Inline..

|-----Original Message-----
|From: Simon Perreault [mailto:simon.perreault@viagenie.ca]
|Sent: Tuesday, July 16, 2013 5:34 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: behave@ietf.org; rtcweb@ietf.org
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt
|
|Le 2013-07-16 12:49, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|> |>    Though ICE specifies STUN Binding indications to be used for
|> |>    keepalives, it requires that an agent be prepared to receive
|> |>    connectivity check as well.  If a connectivity check is received, =
a
|> |>    response is generated, but there is no impact on ICE processing, a=
s
|> |>    described in section 10 of [RFC5245].
|> |
|> |...so? Why is "an impact on ICE processing" necessary?
|>
|> Meant to stress these Binding request/response doesn't trigger an ICE re=
start..
|
|Ok, then s/but/and/.
|
|> |>    While a WebRTC browser could verify whether the peer continues to
|> |>    send SRTCP reports before sending traffic to the peer, the usage o=
f
|> |>    SRTCP together with Security Descriptions [RFC4568] requires expos=
ing
|> |>    the media keys to the JavaScript and renders SRTCP unsuitable for
|> |>    consent freshness.
|> |
|> |Why does it "require exposing the media keys to the JavaScript"? Is thi=
s
|> |because of a law of nature, or is it because of the way the JavaScript
|> |API is being designed? Could the JS API be changed to accommodate
|> |SRTCP+SDES?
|>
|> That's how the API construct looks like today -- the JavaScript would ge=
t an SDP blob from the
|browser containing the crypto keys used for SDES-SRTP. Of course, the brow=
ser could hide those keys by
|putting a "****" in SDP -:). SRTP itself is still being debated in RTCWEB,=
 so nothing is final, yet.
|
|Ok, then say so in the draft.
|
|> |>    Security analysis
|> |>    concluded that the STUN 96 bits transaction ID is sufficient for
|> |>    consent, without needing MESSAGE-INTEGRITY.
|> |
|> |What analysis? You would need to explain it in detail. But if that's no=
t
|> |part of your suggested solution, just remove the sentence.
|>
|> Agree it could be elaborated. The intention was to say the foll:
|>
|>    STUN requires the 96 bits transaction ID to be uniformly and randomly
|>    chosen from the interval 0 .. 2**96-1, and be cryptographically
|>    random. This should suffice to prevent off path attacks on consent
|>    freshness, so MESSAGE-INTEGRITY is not necessary from a security
|>    perspective.
|>
|> However, MESSAGE-INTEGRITY is still used to maintain backward compatibil=
ity
|> with legacy ICE/ICE-lite implementations.
|
|Ok, then say so in the draft.
|
|> |Now on to section "4. Solution Overview"...
|> |
|> |>    Every Tc seconds, the WebRTC browser sends a STUN Binding Request =
to
|> |>    the peer.  This request MUST use a new, cryptographically random
|> |>    Transaction ID [RFC4086], and is formatted as for an ICE connectiv=
ity
|> |>    check [RFC5245].  A valid STUN Binding Response is also formatted =
as
|> |>    for an ICE connectivity check [RFC5245].  The STUN Binding Request
|> |>    and STUN Binding Response are validated as for an ICE connectivity
|> |>    check [RFC5245].
|> |
|> |Couldn't this whole paragraph be simplified to "Every Tc seconds, the
|> |WebRTC browser sends an ICE connectivity check."? Is there anything new
|> |here besides the "every Tc" thing?
|>
|> The difference from the ICE connectivity check is that there is no expon=
ential back off for
|retransmissions.
|
|Ok, then say so in the draft.
|
|> |>    Liveness timer: If no packets have been received on the local port=
 in
|> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
|> |>    connectivity has been lost.  The JavaScript application will use t=
his
|> |>    notification however it desires (e.g., cease transmitting to the
|> |>    remote peer, provide a notification to the user, etc.).
|> |
|> |This seems to me like it will not fulfill the goal set in the abstract:
|> |"to ensure that a malicious JavaScript cannot use the browser as a
|> |platform for launching attacks". If the JavaScript is free to ignore th=
e
|> |notification from the browser, then it has no security benefits. If you
|> |want to reach that goal, the browser needs to forcefully stop transmitt=
ing.
|>
|> That goal is fulfilled by the consent checks -- the browser would stop t=
ransmitting everything on
|that candidate pair, including liveness checks, if there is a consent fail=
ure.
|
|That's not what the draft says. It says that the browser "notifies" the
|JS app. It needs to say that the browser MUST stop sending.

No. That section is about liveness check and its intention is just notify t=
he JavaScript of a potential connectivity loss. It is when the consent chec=
k fails the browser actually stops sending everything. Does the draft need =
more text on the distinction between consent and liveness tests?

|
|> |>    When not actively sending traffic on a nominated candidate pair,
|> |>    performing consent freshness does not serve any purpose from a
|> |>    security perspective.
|> |
|> |I don't understand what this means. Why is the "security perspective"
|> |important here? Aren't we concerned about keepalives?
|>
|> You mean one could use keepalives (Binding indications) for launching at=
tacks, so consent freshness
|would be required for sending them as well?
|
|No.
|
|This is a section about keepalives. I just don't understand this
|sentence, and I don't understand why it talks about security.

Ok, let me elaborate:
- Consent freshness is not necessary when the browser is not sending any=20
  traffic on a candidate pair.
- If the browser is not performing consent freshness on a candidate pair=20
  for the above reason, it performs ICE keepalives (or RTP keepalives) to
  refresh NAT bindings.

Of course, the browser could continue performing consent freshness even whe=
n it is not sending any other traffic on that candidate pair and skip ICE k=
eepalives.

Muthu

|
|Simon

From simon.perreault@viagenie.ca  Tue Jul 16 06:04:00 2013
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Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a écrit :
> |> |>    Liveness timer: If no packets have been received on the local port in
> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
> |> |>    connectivity has been lost.  The JavaScript application will use this
> |> |>    notification however it desires (e.g., cease transmitting to the
> |> |>    remote peer, provide a notification to the user, etc.).
> |> |
> |> |This seems to me like it will not fulfill the goal set in the abstract:
> |> |"to ensure that a malicious JavaScript cannot use the browser as a
> |> |platform for launching attacks". If the JavaScript is free to ignore the
> |> |notification from the browser, then it has no security benefits. If you
> |> |want to reach that goal, the browser needs to forcefully stop transmitting.
> |>
> |> That goal is fulfilled by the consent checks -- the browser would stop transmitting everything on
> |that candidate pair, including liveness checks, if there is a consent failure.
> |
> |That's not what the draft says. It says that the browser "notifies" the
> |JS app. It needs to say that the browser MUST stop sending.
> 
> No. That section is about liveness check and its intention is just notify the JavaScript of a potential connectivity loss. It is when the consent check fails the browser actually stops sending everything. Does the draft need more text on the distinction between consent and liveness tests?

Ah! No, you're right, and the text is already perfectly clear about
this. No need to change. I was just confused.

> |> |>    When not actively sending traffic on a nominated candidate pair,
> |> |>    performing consent freshness does not serve any purpose from a
> |> |>    security perspective.
> |> |
> |> |I don't understand what this means. Why is the "security perspective"
> |> |important here? Aren't we concerned about keepalives?
> |>
> |> You mean one could use keepalives (Binding indications) for launching attacks, so consent freshness
> |would be required for sending them as well?
> |
> |No.
> |
> |This is a section about keepalives. I just don't understand this
> |sentence, and I don't understand why it talks about security.
> 
> Ok, let me elaborate:
> - Consent freshness is not necessary when the browser is not sending any 
>   traffic on a candidate pair.
> - If the browser is not performing consent freshness on a candidate pair 
>   for the above reason, it performs ICE keepalives (or RTP keepalives) to
>   refresh NAT bindings.
> 
> Of course, the browser could continue performing consent freshness even when it is not sending any other traffic on that candidate pair and skip ICE keepalives.

Ah, ok I understand with your explanation. It makes sense. There should
be a way to reformulate the text to make it clearer.

Thanks,
Simon

From andrew.hutton@siemens-enterprise.com  Tue Jul 16 06:22:43 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A compromise for SDES
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Subject: Re: [rtcweb] A compromise for SDES
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> On: 13 July 2013 19:07 Hadriel Kaplan Wrote:
> To: rtcweb@ietf.org
> Subject: [rtcweb] A compromise for SDES
>=20
> Howdy,
> Someone's asked me to explain the compromise I plan on presenting in
> Berlin now, so we have more time to argue over it or chew on it.  The
> reason I was going to wait is because I wanted to explain how I got to
> the point of thinking such a compromise would make sense.  So I'll try
> to do that in this email, which will make this email long.  Nothin'
> much I can do about that.
>=20
> Obviously this is all my personal opinion, not that of my employer, and
> not a statement claiming to be objective fact, etc.
>=20
> Background:
> I think many people would agree that SDES has some benefits compared to
> DTLS for those of us wishing to interface to the SIP world.  Some of
> those benefits are commercial/cost factors, some are more tangible like
> reducing early media clipping.  Whether you agree with such benefits or
> not, so long as WebRTC could support SDES in a sufficiently-secure
> manner, there'd be less concern over having it.  Most of the arguments
> against SDES have been directed at whether it could in fact be
> "sufficiently secure".  Personally I find most of the arguments against
> it being secure enough to be essentially FUD, and/or also apply to
> DTLS-SRTP.
>=20
> On the flip-side, I think the commercial/cost factor for SDES is not a
> strong cause, because I believe the market will bear whatever the extra
> cost may be. (I wasn't sure of this 2 years ago when this all started,
> but I'm fairly confident of it now)  And I think having only DTLS-SRTP
> as MTI would have a marketing benefit for WebRTC as a technology, which
> I value highly.

There is a significant cost factor in scenarios where a large percentage of=
 sessions have to be interworked and whether the market will bear this or n=
ot it is certainly a cost that it would be good to avoid.=20


>=20
> Given those personal feelings, I didn't much care either way, so long
> as a decision was made - although I still preferred having SDES to
> avoid the early media clipping problem.  So when I was going to present
> on this topic back in the Vancouver meeting, my last slide basically
> said: "If we support SDES and it turns out later we were wrong, we can
> always rescind it".  This was based on the notion that browsers get
> updated all the time, especially for security issues.
>=20
> But later it occurred to me that's actually the Achilles' heel of
> supporting SDES in WebRTC, for those of us wanting to gateway this
> stuff to SIP.  Imagine if it turns out we were wrong, and some
> expected-or-unexpected vulnerability is exploited against some large
> WebRTC service provider, and makes the news/slashdot.  Browser vendors
> would instantly have new code removing SDES support, and browsers in
> devices would be updated virtually overnight - some users may take a
> long time to upgrade their specific browsers on their specific devices
> - but many of them would do so within days.  That would be untenable
> for a WebRTC service provider.  Even the smaller Enterprises take a
> long time to upgrade code on their servers, so even for them changing
> their gateways to suddenly support DTLS-SRTP would be unrealistic.
> Imagine what it would be for larger providers, who might even have to
> deploy more servers to handle the sudden additional overhead of DTLS-
> SRTP.  Meanwhile a growing perc
>  entage of their users can no longer use the service.  That's bad mojo.
>=20
> The things that a WebRTC-to-SIP service provider can control, and
> arguably the much larger use-case for this stuff to begin with, are not
> really true "browsers" anyway - they're web-based-framework device
> Apps, provided by the service provider to begin with.  I.e., the apps
> they build using web application framework things like
> PhoneGap/Cordova, Appcelerator Titanium, etc.  I mean using a real
> Browser is nice for ad-hoc stuff, like being able to click on a "talk
> to a rep" button on a website, or a webex/meetecho type thing; but for
> real communication "service", whether it be as a subscriber of a
> traditional carrier, or Skype, or an employee of an Enterprise, or a
> call-center attendant, or whatever - for those most people would never
> want to have to open a browser just to receive/make calls.  They'd give
> you an app to use instead.
>=20
> So it's the app use-case that would benefit the most from SDES,
> especially for issues like media clipping.  And the app use-case
> doesn't have the security concerns nor concerns for unforeseen
> overnight updates.
>=20
> The Compromise:
> So given that background, I was planning to propose that the security
> doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
> statement that web-based application frameworks SHOULD also support
> SDES. (with text about why and how, etc.)

I don't think we can or should say that SDES SHOULD be implemented by one t=
ype of framework but not another we just need to say something consistent f=
or all. I agree with most of what is stated in the reasoning above and give=
n that there is a strong commercial incentive for SDES support I think keep=
ing DTLS-SRTP as the MTI and making SDES a SHOULD would be a good compromis=
e.=20


>=20
> I don't think this is too controversial, because web-based frameworks
> are never beholden to following browser behavior anyway - they're used
> to build a native application, and native applications have a very
> different security/threat model in practice.  They're written for a
> specific purpose, and installed by users from known sources for that
> known purpose.  Technically, afaik, nothing we do in RTCWEB WG or W3C's
> WEBRTC group have any requirements/mandates for native applications
> anyway - an app maker would just ignore something they don't think
> applies to them - but I think web-based frameworks do generally try to
> follow W3C models for Javascript APIs, and will likely read the IETF
> RFCs for the media-layer stuff too.  So I think having this SHOULD
> statement would be beneficial.
>=20
> Or if the WG prefers, we could even write a separate doc on what a web-
> based framework maker should consider supporting/not-supporting. (I can
> imagine a few other things they might want to offer that a browser
> can't)
>=20
> -hadriel
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/8/13 10:43 PM, I=F1aki Baz Castillo wrote:=0A=
> 2013/7/8 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com>:=0A=
>>> So, was there ever a consensus that "developers MUST NOT=0A=
>>> touch SDP"?=0A=
>>=0A=
>> No, developers are free to touch the SDP (and probably must in certain=
=0A=
>> cases).=0A=
>=0A=
>=0A=
> Do you feel how much painful is "touching a browser generated SDP"? To be=
 clear:=0A=
>=0A=
> - JS requests "A" to browser.=0A=
> - Browser returns "A" (in the form of unmanageable blob).=0A=
> - JS modifies "A" to send "A-bis" in the wire.=0A=
> - Remote peer receives "A-bis" and replies "B-bis".=0A=
> - Both JS app are lying to their respective  browsers to get the=0A=
> desired behavior.=0A=
>=0A=
> This is really painful, really.=0A=
=0A=
I've been doing:=0A=
=0A=
- JS requests "A" to browser.=0A=
- Browser returns "A" (in the form of unmanageable blob).=0A=
- JS modifies "A" to send "A-bis" in the wire to a system not =0A=
understanding "A" as is=0A=
- Remote end receives "A-bis" and replies "B-bis".=0A=
- JS modifies "B-bis" (using parts of "A" in the process) and applies =0A=
"B" to browser=0A=
=0A=
Not nice and clean for sure, but quite doable. JS is nice for text parsing.=
=0A=
=0A=
But the big problems I experienced were:=0A=
* "A" was different between different browsers (in a way that the JS had =
=0A=
to be different)=0A=
* "A" would change in ways breaking my code between versions=0A=
* What would accepted for "B" would also vary.=0A=
=0A=
Most of this would presumable go away once we settle on exactly how SDP =0A=
should be used (assuming SDP stays).=0A=
=0A=
But all that said, I still think we need API surface that makes the need =
=0A=
to mangle the SDP an exception (OTOH, perhaps interop with legacy - as I =
=0A=
did - is an exception? Depends on where you come from I guess.).=0A=
=0A=
>=0A=
> In the other side, as I explained in other thread (may be in this=0A=
> one?) adding some methods to the current API (to avoid playing with=0A=
> the SDP) is not the way to go. A specification in which the JS app=0A=
> does not know what he is sending in-the-wire (a blob generated by the=0A=
> browser) is doomed to failure.=0A=
>=0A=
> Please remember that mandating SDP (plain SDP) means that no other=0A=
> media signaling protocol can be implemented in future WebRTC apps.=0A=
=0A=
I think Peter Thatcher had an idea of an API extension where you could =0A=
do away with SDP if you want to.=0A=
=0A=
>=0A=
> SIP phones have *fixed* code and *fixed* logic. This is no longer true=0A=
> in the Web.=0A=
>=0A=
> --=0A=
> I=F1aki Baz Castillo=0A=
> <ibc@aliax.net>=0A=
>=0A=
=0A=

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Subject: Re: [rtcweb] A compromise for SDES
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On 7/16/13 3:22 PM, Hutton, Andrew wrote:
>> The Compromise:
>> >So given that background, I was planning to propose that the security
>> >doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
>> >statement that web-based application frameworks SHOULD also support
>> >SDES. (with text about why and how, etc.)
> I don't think we can or should say that SDES SHOULD be implemented by one type of framework but not another we just need to say something consistent for all. I agree with most of what is stated in the reasoning above and given that there is a strong commercial incentive for SDES support I think keeping DTLS-SRTP as the MTI and making SDES a SHOULD would be a good compromise.
>
I also agree with most of what Hadriel stated in his long mail;
however I don't like the idea we start to make difference between what 
kind of webrtc you can do
with a browser and what you can do on a web-based application framework
(btw I also share the Andrew concern about the fact that we can/should 
say what to do or not to do
in each type of framework)

/Salvatore


From enrico.marocco@telecomitalia.it  Tue Jul 16 07:35:34 2013
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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On 7/15/13 5:15 PM, Bo Burman wrote:
> In that draft, I would prefer something more in line with:
>=20
> "If other suitable audio codecs are available to the browser to use,
> it is recommended that they are also included in the offer in order
> to maximize the possibility to establish the session without the need
> for audio transcoding".

Yes, in fact this is happening already:
https://hacks.mozilla.org/2013/01/firefox-development-highlights-h-264-mp=
3-support-on-windows-scoped-stylesheets-more/

If people feel SHOULD/RECOMMENDED to be too strong, I'm sure we can find
the right keyword in RFC 6919..

Enrico


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From keith.drage@alcatel-lucent.com  Tue Jul 16 07:58:33 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "clue@ietf.org" <clue@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>, "avt@ietf.org" <avt@ietf.org>
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Date: Tue, 16 Jul 2013 14:58:24 +0000
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Subject: [rtcweb] FW: Taxonomy
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Extensive cross-post do not reply to this email.

For information. We do expect participation input from all the listed worki=
ng groups (and possibly more).

Keith

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf Of=
 DRAGE, Keith (Keith)
Sent: 16 July 2013 15:53
To: avtext@ietf.org
Subject: [avtext] Taxonomy

(As AVTEXT WG cochair)

The latest version of the RTP taxonomy draft has been submitted.

https://datatracker.ietf.org/doc/draft-lennox-raiarea-rtp-grouping-taxonomy=
/=20

This document was allocated to AVTEXT by DISPATCH, and we have created a mi=
lestone for it in AVTEXT.

This draft is not yet a working group document.

We do expect to spend some significant meeting time discussing the contents=
 in the AVTEXT meeting in Berlin, so please start raising your issues on th=
e AVTEXT list.

Additionally, if there are counter proposals existing that the AVTEXT chair=
s would not otherwise be aware of, then please identify them to the AVTEXT =
chairs / list.

Regards

Keith
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From Kalyani.Bogineni@VerizonWireless.com  Tue Jul 16 10:02:27 2013
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From: "Bogineni, Kalyani" <Kalyani.Bogineni@VerizonWireless.com>
To: 'Bo Burman' <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 16 Jul 2013 13:02:07 -0400
Thread-Topic: Some thoughts on optional audio codecs
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Cc: "Bogineni, Kalyani" <Kalyani.Bogineni@VerizonWireless.com>
Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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We support the following wording proposal from Bo Burman.

"If other suitable audio codecs are available to the browser to use, it is =
recommended that they are also included in the offer in order to maximize t=
he possibility to establish the session without the need for audio transcod=
ing".

Regards,
Kalyani Bogineni
Verizon

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bo Burman
Sent: Monday, July 15, 2013 11:15 AM
To: rtcweb@ietf.org
Subject: [rtcweb] Some thoughts on optional audio codecs

Regarding the previous discussion on optional audio codecs in the (currentl=
y expired) draft on RTCWEB audio codecs (https://datatracker.ietf.org/doc/d=
raft-ietf-rtcweb-audio/):

I think most parties involved in WebRTC work, myself included, hope and bel=
ieve that it will be ubiquitous and easy to include real-time media convers=
ation functionality in nearly any web context. Since it will be that easy, =
it can be expected that most web developers need not be, and thus will not =
be, media specialists or very knowledgeable about codecs.

The definition of RTCWEB MTI codecs ensures that communication is possible =
since at least one codec will always be found, but it is not possible to cl=
aim the resulting communication to be optimum for every possible context.

Even if WebRTC will be close to ubiquitous, there will for quite some time =
likely be a desire to reach real-time media domains and devices that were n=
ot originally designed for and thus are not optimized for use with WebRTC. =
A communication device that is not designed solely for WebRTC use will like=
ly include functionality and codecs also for its "native" domain.

Any added cost of not being able to use existing "native" codecs will vary =
both in amount and where the cost has to be taken. Eliminating it is indeed=
 an optimization, but the total cost savings may still be significant.

With the current design and to my understanding, it will be the browser ven=
dor's choice to add optional codecs, including any "native" domain codecs. =
 The choice may possibly be delegated to individual web developers making u=
se of WebRTC functionality. A browser vendor will arguably have to know eac=
h target platform to some extent, but it can hardly be assumed that a web d=
eveloper knows the capabilities of all devices that will use the WebRTC-ena=
bled site unless the browser can provide the needed information. There is a=
 risk that "native" codecs in devices are not well handled, unless the moti=
vations and methods to make use of them are better specified.

While any audio codecs besides the MTI ones are clearly optional, I believe=
 the suggested text addition on optional audio codecs to the RTCWEB audio d=
raft in Ticket #12 (http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/12#) t=
o be too brief considering the above.

In that draft, I would prefer something more in line with:

"If other suitable audio codecs are available to the browser to use, it is =
recommended that they are also included in the offer in order to maximize t=
he possibility to establish the session without the need for audio transcod=
ing".

Assuming that the browser vendor (or web developer) is sufficiently concern=
ed with codecs to read the audio codecs draft (or the corresponding RFC to-=
be), the above text may, as a start, give some added guidance why non-MTI c=
odecs may be desirable to consider in addition to the MTI ones.

Cheers,
Bo

Multimedia Technologies
Ericsson Research
F=E4r=F6gatan 6
SE-164 80, Kista, Sweden
www.ericsson.com
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From adam@nostrum.com  Tue Jul 16 13:07:19 2013
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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On 7/16/13 09:35, Enrico Marocco wrote:
> On 7/15/13 5:15 PM, Bo Burman wrote:
>> In that draft, I would prefer something more in line with:
>>
>> "If other suitable audio codecs are available to the browser to use,
>> it is recommended that they are also included in the offer in order
>> to maximize the possibility to establish the session without the need
>> for audio transcoding".
> Yes, in fact this is happening already:
> https://hacks.mozilla.org/2013/01/firefox-development-highlights-h-264-mp3-support-on-windows-scoped-stylesheets-more/
>

You have misread that article. What that article says is that Mozilla is 
adding select platform-supplied codecs to Firefox for non-WebRTC uses.

Our implementation of audio codecs for WebRTC continues to support PCMU, 
PCMA, Opus, and nothing else; our implementation use VP8 exclusively for 
video.

/a

From bernard_aboba@hotmail.com  Tue Jul 16 17:15:51 2013
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From silviapfeiffer1@gmail.com  Tue Jul 16 17:27:08 2013
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Wed, 17 Jul 2013 10:26:45 +1000
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The whole point of standardisation at the W3C is for browser to agree
on the knobs. So, I do think it's a valuable effort.
Cheers,
Silvia.


On Wed, Jul 17, 2013 at 10:15 AM, Bernard Aboba
<bernard_aboba@hotmail.com> wrote:
> The problem was that we never defined what mangling browsers had to suppo=
rt. Given all the SDP specs that is actually a huge work item. And creating=
 APIs to avoid mangling isn't actually that much simpler. What knobs do you=
 provide and which ones do you ignore? Browsers might not agree on those ei=
ther.
>
> On Jul 8, 2013, at 1:37 PM, "Stefan H=E5kansson LK" <stefan.lk.hakansson@=
ericsson.com> wrote:
>
>> On 7/8/13 5:22 PM, Roman Shpount wrote:
>>
>>> So, was there ever a consensus that "developers MUST NOT
>>> touch SDP"?
>>
>> No, developers are free to touch the SDP (and probably must in certain
>> cases).
>>
>>>
>>> _____________
>>> Roman Shpount
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From ranjit.avasarala@nsn.com  Tue Jul 16 22:50:58 2013
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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Cc: "behave@ietf.org" <behave@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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Hi Muthu

Though using STUN request/response may be good for querying about consent, =
I feel it is overloading the functionality of STUN. Many publicly available=
 STUN servers or those that are part of Call servers may not support this.

Can't there be some other neutral mechanism to query peer's consent - throu=
gh signaling?=20


Regards
Ranjit



-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 ext Simon Perreault
Sent: Tuesday, July 16, 2013 6:34 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: rtcweb@ietf.org; behave@ietf.org
Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt

Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
> |> |>    Liveness timer: If no packets have been received on the local po=
rt in
> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
> |> |>    connectivity has been lost.  The JavaScript application will use=
 this
> |> |>    notification however it desires (e.g., cease transmitting to the
> |> |>    remote peer, provide a notification to the user, etc.).
> |> |
> |> |This seems to me like it will not fulfill the goal set in the abstrac=
t:
> |> |"to ensure that a malicious JavaScript cannot use the browser as a
> |> |platform for launching attacks". If the JavaScript is free to ignore =
the
> |> |notification from the browser, then it has no security benefits. If y=
ou
> |> |want to reach that goal, the browser needs to forcefully stop transmi=
tting.
> |>
> |> That goal is fulfilled by the consent checks -- the browser would stop=
 transmitting everything on
> |that candidate pair, including liveness checks, if there is a consent fa=
ilure.
> |
> |That's not what the draft says. It says that the browser "notifies" the
> |JS app. It needs to say that the browser MUST stop sending.
>=20
> No. That section is about liveness check and its intention is just notify=
 the JavaScript of a potential connectivity loss. It is when the consent ch=
eck fails the browser actually stops sending everything. Does the draft nee=
d more text on the distinction between consent and liveness tests?

Ah! No, you're right, and the text is already perfectly clear about
this. No need to change. I was just confused.

> |> |>    When not actively sending traffic on a nominated candidate pair,
> |> |>    performing consent freshness does not serve any purpose from a
> |> |>    security perspective.
> |> |
> |> |I don't understand what this means. Why is the "security perspective"
> |> |important here? Aren't we concerned about keepalives?
> |>
> |> You mean one could use keepalives (Binding indications) for launching =
attacks, so consent freshness
> |would be required for sending them as well?
> |
> |No.
> |
> |This is a section about keepalives. I just don't understand this
> |sentence, and I don't understand why it talks about security.
>=20
> Ok, let me elaborate:
> - Consent freshness is not necessary when the browser is not sending any=
=20
>   traffic on a candidate pair.
> - If the browser is not performing consent freshness on a candidate pair=
=20
>   for the above reason, it performs ICE keepalives (or RTP keepalives) to
>   refresh NAT bindings.
>=20
> Of course, the browser could continue performing consent freshness even w=
hen it is not sending any other traffic on that candidate pair and skip ICE=
 keepalives.

Ah, ok I understand with your explanation. It makes sense. There should
be a way to reformulate the text to make it clearer.

Thanks,
Simon
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From mperumal@cisco.com  Tue Jul 16 23:37:22 2013
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Ranjit,

|Many publicly available STUN servers or those that are part of Call server=
s
|may not support this.

Public STUN servers and call servers are not expected to support this.

|Can't there be some other neutral mechanism to query peer's consent - thro=
ugh=20
|signaling?

No. That signaling between the JS and the web server could be anything (SIP=
 or XMPP or proprietary or whatever) and the browser has no control over it=
.

Muthu

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 11:19 AM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Hi Muthu
|
|Though using STUN request/response may be good for querying about consent,=
 I feel it is overloading
|the functionality of STUN. Many publicly available STUN servers or those t=
hat are part of Call servers
|may not support this.
|
|Can't there be some other neutral mechanism to query peer's consent - thro=
ugh signaling?
|
|
|Regards
|Ranjit
|
|
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f ext Simon Perreault
|Sent: Tuesday, July 16, 2013 6:34 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|> |> |>    Liveness timer: If no packets have been received on the local p=
ort in
|> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
|> |> |>    connectivity has been lost.  The JavaScript application will us=
e this
|> |> |>    notification however it desires (e.g., cease transmitting to th=
e
|> |> |>    remote peer, provide a notification to the user, etc.).
|> |> |
|> |> |This seems to me like it will not fulfill the goal set in the abstra=
ct:
|> |> |"to ensure that a malicious JavaScript cannot use the browser as a
|> |> |platform for launching attacks". If the JavaScript is free to ignore=
 the
|> |> |notification from the browser, then it has no security benefits. If =
you
|> |> |want to reach that goal, the browser needs to forcefully stop transm=
itting.
|> |>
|> |> That goal is fulfilled by the consent checks -- the browser would sto=
p transmitting everything on
|> |that candidate pair, including liveness checks, if there is a consent f=
ailure.
|> |
|> |That's not what the draft says. It says that the browser "notifies" the
|> |JS app. It needs to say that the browser MUST stop sending.
|>
|> No. That section is about liveness check and its intention is just notif=
y the JavaScript of a
|potential connectivity loss. It is when the consent check fails the browse=
r actually stops sending
|everything. Does the draft need more text on the distinction between conse=
nt and liveness tests?
|
|Ah! No, you're right, and the text is already perfectly clear about
|this. No need to change. I was just confused.
|
|> |> |>    When not actively sending traffic on a nominated candidate pair=
,
|> |> |>    performing consent freshness does not serve any purpose from a
|> |> |>    security perspective.
|> |> |
|> |> |I don't understand what this means. Why is the "security perspective=
"
|> |> |important here? Aren't we concerned about keepalives?
|> |>
|> |> You mean one could use keepalives (Binding indications) for launching=
 attacks, so consent
|freshness
|> |would be required for sending them as well?
|> |
|> |No.
|> |
|> |This is a section about keepalives. I just don't understand this
|> |sentence, and I don't understand why it talks about security.
|>
|> Ok, let me elaborate:
|> - Consent freshness is not necessary when the browser is not sending any
|>   traffic on a candidate pair.
|> - If the browser is not performing consent freshness on a candidate pair
|>   for the above reason, it performs ICE keepalives (or RTP keepalives) t=
o
|>   refresh NAT bindings.
|>
|> Of course, the browser could continue performing consent freshness even =
when it is not sending any
|other traffic on that candidate pair and skip ICE keepalives.
|
|Ah, ok I understand with your explanation. It makes sense. There should
|be a way to reformulate the text to make it clearer.
|
|Thanks,
|Simon
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From ranjit.avasarala@nsn.com  Tue Jul 16 23:47:08 2013
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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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Hi Muthu

If many public STUN servers and call servers that implement STUN or ICElite=
 functionalities, are not expected to support, then who would have interest=
 to support this functionality? Ideally getting consent for receiving traff=
ic or call should be part of signaling. I do not think using STUN is approp=
riate.

Regards
Ranjit




-----Original Message-----
From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Wednesday, July 17, 2013 12:07 PM
To: Avasarala, Ranjit (NSN - IN/Bangalore)
Cc: rtcweb@ietf.org; behave@ietf.org
Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt

Ranjit,

|Many publicly available STUN servers or those that are part of Call server=
s
|may not support this.

Public STUN servers and call servers are not expected to support this.

|Can't there be some other neutral mechanism to query peer's consent - thro=
ugh=20
|signaling?

No. That signaling between the JS and the web server could be anything (SIP=
 or XMPP or proprietary or whatever) and the browser has no control over it=
.

Muthu

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 11:19 AM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Hi Muthu
|
|Though using STUN request/response may be good for querying about consent,=
 I feel it is overloading
|the functionality of STUN. Many publicly available STUN servers or those t=
hat are part of Call servers
|may not support this.
|
|Can't there be some other neutral mechanism to query peer's consent - thro=
ugh signaling?
|
|
|Regards
|Ranjit
|
|
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f ext Simon Perreault
|Sent: Tuesday, July 16, 2013 6:34 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|> |> |>    Liveness timer: If no packets have been received on the local p=
ort in
|> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
|> |> |>    connectivity has been lost.  The JavaScript application will us=
e this
|> |> |>    notification however it desires (e.g., cease transmitting to th=
e
|> |> |>    remote peer, provide a notification to the user, etc.).
|> |> |
|> |> |This seems to me like it will not fulfill the goal set in the abstra=
ct:
|> |> |"to ensure that a malicious JavaScript cannot use the browser as a
|> |> |platform for launching attacks". If the JavaScript is free to ignore=
 the
|> |> |notification from the browser, then it has no security benefits. If =
you
|> |> |want to reach that goal, the browser needs to forcefully stop transm=
itting.
|> |>
|> |> That goal is fulfilled by the consent checks -- the browser would sto=
p transmitting everything on
|> |that candidate pair, including liveness checks, if there is a consent f=
ailure.
|> |
|> |That's not what the draft says. It says that the browser "notifies" the
|> |JS app. It needs to say that the browser MUST stop sending.
|>
|> No. That section is about liveness check and its intention is just notif=
y the JavaScript of a
|potential connectivity loss. It is when the consent check fails the browse=
r actually stops sending
|everything. Does the draft need more text on the distinction between conse=
nt and liveness tests?
|
|Ah! No, you're right, and the text is already perfectly clear about
|this. No need to change. I was just confused.
|
|> |> |>    When not actively sending traffic on a nominated candidate pair=
,
|> |> |>    performing consent freshness does not serve any purpose from a
|> |> |>    security perspective.
|> |> |
|> |> |I don't understand what this means. Why is the "security perspective=
"
|> |> |important here? Aren't we concerned about keepalives?
|> |>
|> |> You mean one could use keepalives (Binding indications) for launching=
 attacks, so consent
|freshness
|> |would be required for sending them as well?
|> |
|> |No.
|> |
|> |This is a section about keepalives. I just don't understand this
|> |sentence, and I don't understand why it talks about security.
|>
|> Ok, let me elaborate:
|> - Consent freshness is not necessary when the browser is not sending any
|>   traffic on a candidate pair.
|> - If the browser is not performing consent freshness on a candidate pair
|>   for the above reason, it performs ICE keepalives (or RTP keepalives) t=
o
|>   refresh NAT bindings.
|>
|> Of course, the browser could continue performing consent freshness even =
when it is not sending any
|other traffic on that candidate pair and skip ICE keepalives.
|
|Ah, ok I understand with your explanation. It makes sense. There should
|be a way to reformulate the text to make it clearer.
|
|Thanks,
|Simon
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From enrico.marocco@telecomitalia.it  Wed Jul 17 00:44:54 2013
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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--------------ms070901030702060503060000
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On 7/16/13 10:07 PM, Adam Roach wrote:
>>> In that draft, I would prefer something more in line with:
>>>
>>> "If other suitable audio codecs are available to the browser to use,
>>> it is recommended that they are also included in the offer in order
>>> to maximize the possibility to establish the session without the need=

>>> for audio transcoding".
>> Yes, in fact this is happening already:
>> https://hacks.mozilla.org/2013/01/firefox-development-highlights-h-264=
-mp3-support-on-windows-scoped-stylesheets-more/
>=20
> You have misread that article. What that article says is that Mozilla i=
s=20
> adding select platform-supplied codecs to Firefox for non-WebRTC uses.

Sure, didn't mean to imply that it is happening for WebRTC.

> Our implementation of audio codecs for WebRTC continues to support PCMU=
,=20
> PCMA, Opus, and nothing else; our implementation use VP8 exclusively fo=
r=20
> video.

And that's pretty visible also from the outside. What's not visible is
whether at some point in time Mozilla will consider taking advantage of
OS/platform-provided encoding/decoding capabilities also for WebRTC. One
may be led to assume yes, as you're doing it for static content already.

But maybe different conditions apply here, and I don't expect them to be
discussed on this mailing list (even if it would be very, very
interesting!).

Enrico



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From andrew.hutton@siemens-enterprise.com  Wed Jul 17 01:46:01 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Bogineni, Kalyani" <Kalyani.Bogineni@VerizonWireless.com>, 'Bo Burman' <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Some thoughts on optional audio codecs
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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We appear to have been around this loop a number of times the text suggeste=
d here is exactly what was suggested by Andrew Allen back in January and I =
for one supported it them and still do - See http://www.ietf.org/mail-archi=
ve/web/rtcweb/current/msg06121.html.

Not sure there was a definitive conclusion to that particular consensus cal=
l.

Andy


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bogineni, Kalyani
> Sent: 16 July 2013 18:02
> To: 'Bo Burman'; rtcweb@ietf.org
> Cc: Bogineni, Kalyani
> Subject: Re: [rtcweb] Some thoughts on optional audio codecs
>=20
> We support the following wording proposal from Bo Burman.
>=20
> "If other suitable audio codecs are available to the browser to use, it
> is recommended that they are also included in the offer in order to
> maximize the possibility to establish the session without the need for
> audio transcoding".
>=20
> Regards,
> Kalyani Bogineni
> Verizon
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bo Burman
> Sent: Monday, July 15, 2013 11:15 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Some thoughts on optional audio codecs
>=20
> Regarding the previous discussion on optional audio codecs in the
> (currently expired) draft on RTCWEB audio codecs
> (https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/):
>=20
> I think most parties involved in WebRTC work, myself included, hope and
> believe that it will be ubiquitous and easy to include real-time media
> conversation functionality in nearly any web context. Since it will be
> that easy, it can be expected that most web developers need not be, and
> thus will not be, media specialists or very knowledgeable about codecs.
>=20
> The definition of RTCWEB MTI codecs ensures that communication is
> possible since at least one codec will always be found, but it is not
> possible to claim the resulting communication to be optimum for every
> possible context.
>=20
> Even if WebRTC will be close to ubiquitous, there will for quite some
> time likely be a desire to reach real-time media domains and devices
> that were not originally designed for and thus are not optimized for
> use with WebRTC. A communication device that is not designed solely for
> WebRTC use will likely include functionality and codecs also for its
> "native" domain.
>=20
> Any added cost of not being able to use existing "native" codecs will
> vary both in amount and where the cost has to be taken. Eliminating it
> is indeed an optimization, but the total cost savings may still be
> significant.
>=20
> With the current design and to my understanding, it will be the browser
> vendor's choice to add optional codecs, including any "native" domain
> codecs.  The choice may possibly be delegated to individual web
> developers making use of WebRTC functionality. A browser vendor will
> arguably have to know each target platform to some extent, but it can
> hardly be assumed that a web developer knows the capabilities of all
> devices that will use the WebRTC-enabled site unless the browser can
> provide the needed information. There is a risk that "native" codecs in
> devices are not well handled, unless the motivations and methods to
> make use of them are better specified.
>=20
> While any audio codecs besides the MTI ones are clearly optional, I
> believe the suggested text addition on optional audio codecs to the
> RTCWEB audio draft in Ticket #12
> (http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/12#) to be too brief
> considering the above.
>=20
> In that draft, I would prefer something more in line with:
>=20
> "If other suitable audio codecs are available to the browser to use, it
> is recommended that they are also included in the offer in order to
> maximize the possibility to establish the session without the need for
> audio transcoding".
>=20
> Assuming that the browser vendor (or web developer) is sufficiently
> concerned with codecs to read the audio codecs draft (or the
> corresponding RFC to-be), the above text may, as a start, give some
> added guidance why non-MTI codecs may be desirable to consider in
> addition to the MTI ones.
>=20
> Cheers,
> Bo
>=20
> Multimedia Technologies
> Ericsson Research
> F=E4r=F6gatan 6
> SE-164 80, Kista, Sweden
> www.ericsson.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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|If many public STUN servers and call servers that implement STUN or ICElit=
e=20
|functionalities, are not expected to support

I should have said they have no role to play. It is similar to peer-to-peer=
 ICE connectivity checks -- those checks don't go through public STUN or ca=
ll servers at all.=20

The difference between ICE connectivity checks and consent freshness checks=
 is that the former is performed before session establishment and the later=
 is performed after session establishment.

|then who would have interest to support this functionality?

RTCWEB browsers and browser-like platforms.

|Ideally getting consent for receiving traffic or call should be part of si=
gnaling.=20
|I do not think using STUN is appropriate.

ICE already does that. You think it is inappropriate?

Muthu

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 12:16 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Hi Muthu
|
|If many public STUN servers and call servers that implement STUN or ICElit=
e functionalities, are not
|expected to support, then who would have interest to support this function=
ality? Ideally getting
|consent for receiving traffic or call should be part of signaling. I do no=
t think using STUN is
|appropriate.
|
|Regards
|Ranjit
|
|
|
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 12:07 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Ranjit,
|
||Many publicly available STUN servers or those that are part of Call serve=
rs
||may not support this.
|
|Public STUN servers and call servers are not expected to support this.
|
||Can't there be some other neutral mechanism to query peer's consent - thr=
ough
||signaling?
|
|No. That signaling between the JS and the web server could be anything (SI=
P or XMPP or proprietary or
|whatever) and the browser has no control over it.
|
|Muthu
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 11:19 AM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Hi Muthu
||
||Though using STUN request/response may be good for querying about consent=
, I feel it is overloading
||the functionality of STUN. Many publicly available STUN servers or those =
that are part of Call
|servers
||may not support this.
||
||Can't there be some other neutral mechanism to query peer's consent - thr=
ough signaling?
||
||
||Regards
||Ranjit
||
||
||
||-----Original Message-----
||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of ext Simon Perreault
||Sent: Tuesday, July 16, 2013 6:34 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
||> |> |>    Liveness timer: If no packets have been received on the local =
port in
||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
||> |> |>    connectivity has been lost.  The JavaScript application will u=
se this
||> |> |>    notification however it desires (e.g., cease transmitting to t=
he
||> |> |>    remote peer, provide a notification to the user, etc.).
||> |> |
||> |> |This seems to me like it will not fulfill the goal set in the abstr=
act:
||> |> |"to ensure that a malicious JavaScript cannot use the browser as a
||> |> |platform for launching attacks". If the JavaScript is free to ignor=
e the
||> |> |notification from the browser, then it has no security benefits. If=
 you
||> |> |want to reach that goal, the browser needs to forcefully stop trans=
mitting.
||> |>
||> |> That goal is fulfilled by the consent checks -- the browser would st=
op transmitting everything
|on
||> |that candidate pair, including liveness checks, if there is a consent =
failure.
||> |
||> |That's not what the draft says. It says that the browser "notifies" th=
e
||> |JS app. It needs to say that the browser MUST stop sending.
||>
||> No. That section is about liveness check and its intention is just noti=
fy the JavaScript of a
||potential connectivity loss. It is when the consent check fails the brows=
er actually stops sending
||everything. Does the draft need more text on the distinction between cons=
ent and liveness tests?
||
||Ah! No, you're right, and the text is already perfectly clear about
||this. No need to change. I was just confused.
||
||> |> |>    When not actively sending traffic on a nominated candidate pai=
r,
||> |> |>    performing consent freshness does not serve any purpose from a
||> |> |>    security perspective.
||> |> |
||> |> |I don't understand what this means. Why is the "security perspectiv=
e"
||> |> |important here? Aren't we concerned about keepalives?
||> |>
||> |> You mean one could use keepalives (Binding indications) for launchin=
g attacks, so consent
||freshness
||> |would be required for sending them as well?
||> |
||> |No.
||> |
||> |This is a section about keepalives. I just don't understand this
||> |sentence, and I don't understand why it talks about security.
||>
||> Ok, let me elaborate:
||> - Consent freshness is not necessary when the browser is not sending an=
y
||>   traffic on a candidate pair.
||> - If the browser is not performing consent freshness on a candidate pai=
r
||>   for the above reason, it performs ICE keepalives (or RTP keepalives) =
to
||>   refresh NAT bindings.
||>
||> Of course, the browser could continue performing consent freshness even=
 when it is not sending any
||other traffic on that candidate pair and skip ICE keepalives.
||
||Ah, ok I understand with your explanation. It makes sense. There should
||be a way to reformulate the text to make it clearer.
||
||Thanks,
||Simon
||_______________________________________________
||rtcweb mailing list
||rtcweb@ietf.org
||https://www.ietf.org/mailman/listinfo/rtcweb

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--001a11c20dfe7dc3c104e1b15bda
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On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Perumal (mperumal) <
mperumal@cisco.com> wrote:

> [Added rtcweb since I am not sure if everyone involved there are followin=
g
> this discussion in behave]
>
> Thanks for the review. See inline..
>
> |-----Original Message-----
> |From: behave-bounces@ietf.org [mailto:behave-bounces@ietf.org] On Behalf
> Of Simon Perreault
> |Sent: Tuesday, July 16, 2013 3:05 PM
> |To: behave@ietf.org
> |Subject: Re: [BEHAVE] FW: I-D Action:
> draft-muthu-behave-consent-freshness-04.txt
> |
> |Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
> |> The text and the algorithm in the draft are significantly simplified i=
n
> this updated version.
> |>
> |> Comments welcome..
> |
> |MUCH better introduction. Now I feel like I understand the need exactly.
> |
> |The "Design Considerations" section is still very confusing to me.
> |
> |>    Though ICE specifies STUN Binding indications to be used for
> |>    keepalives, it requires that an agent be prepared to receive
> |>    connectivity check as well.  If a connectivity check is received, a
> |>    response is generated, but there is no impact on ICE processing, as
> |>    described in section 10 of [RFC5245].
> |
> |...so? Why is "an impact on ICE processing" necessary?
>
> Meant to stress these Binding request/response doesn't trigger an ICE
> restart..
>
> |
> |>    While a WebRTC browser could verify whether the peer continues to
> |>    send SRTCP reports before sending traffic to the peer, the usage of
> |>    SRTCP together with Security Descriptions [RFC4568] requires exposi=
ng
> |>    the media keys to the JavaScript and renders SRTCP unsuitable for
> |>    consent freshness.
> |
> |Why does it "require exposing the media keys to the JavaScript"? Is this
> |because of a law of nature, or is it because of the way the JavaScript
> |API is being designed? Could the JS API be changed to accommodate
> |SRTCP+SDES?
>
> That's how the API construct looks like today -- the JavaScript would get
> an SDP blob from the browser containing the crypto keys used for SDES-SRT=
P.
> Of course, the browser could hide those keys by putting a "****" in SDP -=
:).



Uh, then how would they be sent to the other side?

As far as I can tell, with any plausible API SDES requires exposing the
keys to JS.

-Ekr

--001a11c20dfe7dc3c104e1b15bda
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Perumal (mperumal=
) <span dir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_bl=
ank">mperumal@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">[Added rtcweb since I am not sure if everyon=
e involved there are following this discussion in behave]<br>
<br>
Thanks for the review. See inline..<br>
<br>
|-----Original Message-----<br>
|From: <a href=3D"mailto:behave-bounces@ietf.org">behave-bounces@ietf.org</=
a> [mailto:<a href=3D"mailto:behave-bounces@ietf.org">behave-bounces@ietf.o=
rg</a>] On Behalf Of Simon Perreault<br>
|Sent: Tuesday, July 16, 2013 3:05 PM<br>
|To: <a href=3D"mailto:behave@ietf.org">behave@ietf.org</a><br>
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt<br>
|<br>
|Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :<br>
<div class=3D"im">|&gt; The text and the algorithm in the draft are signifi=
cantly simplified in this updated version.<br>
|&gt;<br>
|&gt; Comments welcome..<br>
|<br>
</div>|MUCH better introduction. Now I feel like I understand the need exac=
tly.<br>
|<br>
|The &quot;Design Considerations&quot; section is still very confusing to m=
e.<br>
|<br>
|&gt; =A0 =A0Though ICE specifies STUN Binding indications to be used for<b=
r>
|&gt; =A0 =A0keepalives, it requires that an agent be prepared to receive<b=
r>
|&gt; =A0 =A0connectivity check as well. =A0If a connectivity check is rece=
ived, a<br>
|&gt; =A0 =A0response is generated, but there is no impact on ICE processin=
g, as<br>
|&gt; =A0 =A0described in section 10 of [RFC5245].<br>
|<br>
|...so? Why is &quot;an impact on ICE processing&quot; necessary?<br>
<br>
Meant to stress these Binding request/response doesn&#39;t trigger an ICE r=
estart..<br>
<br>
|<br>
|&gt; =A0 =A0While a WebRTC browser could verify whether the peer continues=
 to<br>
|&gt; =A0 =A0send SRTCP reports before sending traffic to the peer, the usa=
ge of<br>
|&gt; =A0 =A0SRTCP together with Security Descriptions [RFC4568] requires e=
xposing<br>
|&gt; =A0 =A0the media keys to the JavaScript and renders SRTCP unsuitable =
for<br>
|&gt; =A0 =A0consent freshness.<br>
|<br>
|Why does it &quot;require exposing the media keys to the JavaScript&quot;?=
 Is this<br>
|because of a law of nature, or is it because of the way the JavaScript<br>
|API is being designed? Could the JS API be changed to accommodate<br>
|SRTCP+SDES?<br>
<br>
That&#39;s how the API construct looks like today -- the JavaScript would g=
et an SDP blob from the browser containing the crypto keys used for SDES-SR=
TP. Of course, the browser could hide those keys by putting a &quot;****&qu=
ot; in SDP -:).</blockquote>

<div><br></div><div><br></div><div>Uh, then how would they be sent to the o=
ther side?</div><div><br></div><div>As far as I can tell, with any plausibl=
e API SDES requires exposing the keys to JS.</div><div><br></div><div>
-Ekr</div>
<div><br></div></div></div></div>

--001a11c20dfe7dc3c104e1b15bda--

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My comments inline <Ranjit>


Regards
Ranjit




-----Original Message-----
From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Wednesday, July 17, 2013 2:16 PM
To: Avasarala, Ranjit (NSN - IN/Bangalore)
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt

|If many public STUN servers and call servers that implement STUN or ICElit=
e=20
|functionalities, are not expected to support

I should have said they have no role to play. It is similar to peer-to-peer=
 ICE connectivity checks -- those checks don't go through public STUN or ca=
ll servers at all.=20
[Ranjit] But there can be scenarios where RTCWebBrowsers do talk to WebRTC =
Gateways which could be co-located with call handling servers. Then it won'=
t make sense to have this feature.

The difference between ICE connectivity checks and consent freshness checks=
 is that the former is performed before session establishment and the later=
 is performed after session establishment.
[Ranjit] Why do you need to check consent of the peer after session is esta=
blished? Ideally I should check consent before session establishment. If th=
e peer does not want to receive traffic, then session itself should not be =
established.=20

|then who would have interest to support this functionality?

RTCWEB browsers and browser-like platforms.
[Ranjit] browser may implement, but it would be very customized one. I do n=
ot think the intermediate entities would want to support.=20

|Ideally getting consent for receiving traffic or call should be part of si=
gnaling.=20
|I do not think using STUN is appropriate.

ICE already does that. You think it is inappropriate?
[Ranjit] I am saying that using STUN for consent check is not appropriate a=
s STUN has other dedicated functionality to do - like connectivity checks, =
keep alives, etc. Also in cases of Symmetric NATs where STUN fails, TURN is=
 needed. Then the proposed feature would anyway fail.

Muthu

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 12:16 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Hi Muthu
|
|If many public STUN servers and call servers that implement STUN or ICElit=
e functionalities, are not
|expected to support, then who would have interest to support this function=
ality? Ideally getting
|consent for receiving traffic or call should be part of signaling. I do no=
t think using STUN is
|appropriate.
|
|Regards
|Ranjit
|
|
|
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 12:07 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org; behave@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|Ranjit,
|
||Many publicly available STUN servers or those that are part of Call serve=
rs
||may not support this.
|
|Public STUN servers and call servers are not expected to support this.
|
||Can't there be some other neutral mechanism to query peer's consent - thr=
ough
||signaling?
|
|No. That signaling between the JS and the web server could be anything (SI=
P or XMPP or proprietary or
|whatever) and the browser has no control over it.
|
|Muthu
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 11:19 AM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Hi Muthu
||
||Though using STUN request/response may be good for querying about consent=
, I feel it is overloading
||the functionality of STUN. Many publicly available STUN servers or those =
that are part of Call
|servers
||may not support this.
||
||Can't there be some other neutral mechanism to query peer's consent - thr=
ough signaling?
||
||
||Regards
||Ranjit
||
||
||
||-----Original Message-----
||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of ext Simon Perreault
||Sent: Tuesday, July 16, 2013 6:34 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
||> |> |>    Liveness timer: If no packets have been received on the local =
port in
||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript that
||> |> |>    connectivity has been lost.  The JavaScript application will u=
se this
||> |> |>    notification however it desires (e.g., cease transmitting to t=
he
||> |> |>    remote peer, provide a notification to the user, etc.).
||> |> |
||> |> |This seems to me like it will not fulfill the goal set in the abstr=
act:
||> |> |"to ensure that a malicious JavaScript cannot use the browser as a
||> |> |platform for launching attacks". If the JavaScript is free to ignor=
e the
||> |> |notification from the browser, then it has no security benefits. If=
 you
||> |> |want to reach that goal, the browser needs to forcefully stop trans=
mitting.
||> |>
||> |> That goal is fulfilled by the consent checks -- the browser would st=
op transmitting everything
|on
||> |that candidate pair, including liveness checks, if there is a consent =
failure.
||> |
||> |That's not what the draft says. It says that the browser "notifies" th=
e
||> |JS app. It needs to say that the browser MUST stop sending.
||>
||> No. That section is about liveness check and its intention is just noti=
fy the JavaScript of a
||potential connectivity loss. It is when the consent check fails the brows=
er actually stops sending
||everything. Does the draft need more text on the distinction between cons=
ent and liveness tests?
||
||Ah! No, you're right, and the text is already perfectly clear about
||this. No need to change. I was just confused.
||
||> |> |>    When not actively sending traffic on a nominated candidate pai=
r,
||> |> |>    performing consent freshness does not serve any purpose from a
||> |> |>    security perspective.
||> |> |
||> |> |I don't understand what this means. Why is the "security perspectiv=
e"
||> |> |important here? Aren't we concerned about keepalives?
||> |>
||> |> You mean one could use keepalives (Binding indications) for launchin=
g attacks, so consent
||freshness
||> |would be required for sending them as well?
||> |
||> |No.
||> |
||> |This is a section about keepalives. I just don't understand this
||> |sentence, and I don't understand why it talks about security.
||>
||> Ok, let me elaborate:
||> - Consent freshness is not necessary when the browser is not sending an=
y
||>   traffic on a candidate pair.
||> - If the browser is not performing consent freshness on a candidate pai=
r
||>   for the above reason, it performs ICE keepalives (or RTP keepalives) =
to
||>   refresh NAT bindings.
||>
||> Of course, the browser could continue performing consent freshness even=
 when it is not sending any
||other traffic on that candidate pair and skip ICE keepalives.
||
||Ah, ok I understand with your explanation. It makes sense. There should
||be a way to reformulate the text to make it clearer.
||
||Thanks,
||Simon
||_______________________________________________
||rtcweb mailing list
||rtcweb@ietf.org
||https://www.ietf.org/mailman/listinfo/rtcweb

From keith.drage@alcatel-lucent.com  Wed Jul 17 04:07:01 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "clue@ietf.org" <clue@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>, "avt@ietf.org" <avt@ietf.org>
Thread-Topic: Taxonomy
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Date: Wed, 17 Jul 2013 11:06:48 +0000
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Breaking my own rule about cross posting...

Just wanted to add is that this document would not be retrospective, i.e. w=
e would not start rewriting the existing published RFCs to take into accoun=
t the new terminology.

The expectation however is that documents going forward from other working =
groups would use the new terminology rather than other terms.=20

Obviously how this is handled is up to the individual working groups, but u=
sing the new terminology will be the only good way of making your results u=
nderstandable outside your specific field.

So that is why your review and input is important.

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith)
> Sent: 16 July 2013 15:58
> To: mmusic@ietf.org; rtcweb@ietf.org; clue@ietf.org; dispatch@ietf.org;
> avt@ietf.org
> Subject: FW: Taxonomy
>=20
> Extensive cross-post do not reply to this email.
>=20
> For information. We do expect participation input from all the listed
> working groups (and possibly more).
>=20
> Keith
>=20
> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of DRAGE, Keith (Keith)
> Sent: 16 July 2013 15:53
> To: avtext@ietf.org
> Subject: [avtext] Taxonomy
>=20
> (As AVTEXT WG cochair)
>=20
> The latest version of the RTP taxonomy draft has been submitted.
>=20
> https://datatracker.ietf.org/doc/draft-lennox-raiarea-rtp-grouping-
> taxonomy/
>=20
> This document was allocated to AVTEXT by DISPATCH, and we have created a
> milestone for it in AVTEXT.
>=20
> This draft is not yet a working group document.
>=20
> We do expect to spend some significant meeting time discussing the
> contents in the AVTEXT meeting in Berlin, so please start raising your
> issues on the AVTEXT list.
>=20
> Additionally, if there are counter proposals existing that the AVTEXT
> chairs would not otherwise be aware of, then please identify them to the
> AVTEXT chairs / list.
>=20
> Regards
>=20
> Keith
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

From ibc@aliax.net  Wed Jul 17 04:27:12 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2013/7/17 DRAGE, Keith (Keith) <keith.drage@alcatel-lucent.com>:
> The expectation however is that documents going forward from other workin=
g groups would use the new terminology rather than other terms.
>
> Obviously how this is handled is up to the individual working groups, but=
 using the new terminology will be the only good way of making your results=
 understandable outside your specific field.
>
> So that is why your review and input is important.

Hi Drage,

First of I'm sorry but I'm not subscribed to MMUSIC nor CLUE, so let
me please reply here (it is just a single topic):

I'm really annoying after reading the draft section 2.4 "Media Stream":

  http://tools.ietf.org/html/draft-lennox-raiarea-rtp-grouping-taxonomy-01#=
section-2.4

------------------------
      Each Media Stream is identified by a unique Synchronization source
      (SSRC) [RFC3550] that is carried in every RTP and Real-time
      Transport Control Protocol (RTCP) packet header.
------------------------

In W3C we have "MediaStream" and "MediaStreamTrack", and
"MediaStreamTrack" is exactly what the draft above defines as "Media
Stream".

So let me say that this is the most confusing terminology it can be.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Le 2013-07-17 11:05, Avasarala, Ranjit (NSN - IN/Bangalore) a écrit :
> Why do you need to check consent of the peer after session is established? [...]
> I am saying that using STUN for consent check is not appropriate [...]

Ranjit,

I would expect one to fully understand the need for consent checks
*before* declaring the proposed solution inappropriate.

Simon

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Date: Wed, 17 Jul 2013 12:47:02 +0000
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|Uh, then how would they be sent to the other side?

Ah, the browser would have to modify the SDP when it is sent on the wire..

|As far as I can tell, with any plausible API SDES requires exposing the ke=
ys to JS.

That helps..thanks.

Muthu

From: Eric Rescorla [mailto:ekr@rtfm.com]
Sent: Wednesday, July 17, 2013 2:32 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: Simon Perreault; behave@ietf.org; rtcweb@ietf.org
Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt



On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Perumal (mperumal) <mperu=
mal@cisco.com<mailto:mperumal@cisco.com>> wrote:
[Added rtcweb since I am not sure if everyone involved there are following =
this discussion in behave]

Thanks for the review. See inline..

|-----Original Message-----
|From: behave-bounces@ietf.org<mailto:behave-bounces@ietf.org> [mailto:beha=
ve-bounces@ietf.org<mailto:behave-bounces@ietf.org>] On Behalf Of Simon Per=
reault
|Sent: Tuesday, July 16, 2013 3:05 PM
|To: behave@ietf.org<mailto:behave@ietf.org>
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt
|
|Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|> The text and the algorithm in the draft are significantly simplified in =
this updated version.
|>
|> Comments welcome..
|
|MUCH better introduction. Now I feel like I understand the need exactly.
|
|The "Design Considerations" section is still very confusing to me.
|
|>    Though ICE specifies STUN Binding indications to be used for
|>    keepalives, it requires that an agent be prepared to receive
|>    connectivity check as well.  If a connectivity check is received, a
|>    response is generated, but there is no impact on ICE processing, as
|>    described in section 10 of [RFC5245].
|
|...so? Why is "an impact on ICE processing" necessary?

Meant to stress these Binding request/response doesn't trigger an ICE resta=
rt..

|
|>    While a WebRTC browser could verify whether the peer continues to
|>    send SRTCP reports before sending traffic to the peer, the usage of
|>    SRTCP together with Security Descriptions [RFC4568] requires exposing
|>    the media keys to the JavaScript and renders SRTCP unsuitable for
|>    consent freshness.
|
|Why does it "require exposing the media keys to the JavaScript"? Is this
|because of a law of nature, or is it because of the way the JavaScript
|API is being designed? Could the JS API be changed to accommodate
|SRTCP+SDES?

That's how the API construct looks like today -- the JavaScript would get a=
n SDP blob from the browser containing the crypto keys used for SDES-SRTP. =
Of course, the browser could hide those keys by putting a "****" in SDP -:)=
.


Uh, then how would they be sent to the other side?

As far as I can tell, with any plausible API SDES requires exposing the key=
s to JS.

-Ekr


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<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|Uh, then how would they be sent to the other side?<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Ah, the browser would have to modify the SDP when it is se=
nt on the wire..<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|As far as I can tell, with any plausible API SDES require=
s exposing the keys to JS.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">That helps..thanks.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Eric Res=
corla [mailto:ekr@rtfm.com]
<br>
<b>Sent:</b> Wednesday, July 17, 2013 2:32 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> Simon Perreault; behave@ietf.org; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-co=
nsent-freshness-04.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Pe=
rumal (mperumal) &lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_blank=
">mperumal@cisco.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">[Added rtcweb since I am not sure if everyone involv=
ed there are following this discussion in behave]<br>
<br>
Thanks for the review. See inline..<br>
<br>
|-----Original Message-----<br>
|From: <a href=3D"mailto:behave-bounces@ietf.org">behave-bounces@ietf.org</=
a> [mailto:<a href=3D"mailto:behave-bounces@ietf.org">behave-bounces@ietf.o=
rg</a>] On Behalf Of Simon Perreault<br>
|Sent: Tuesday, July 16, 2013 3:05 PM<br>
|To: <a href=3D"mailto:behave@ietf.org">behave@ietf.org</a><br>
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt<br>
|<br>
|Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :<o:p><=
/o:p></p>
<div>
<p class=3D"MsoNormal">|&gt; The text and the algorithm in the draft are si=
gnificantly simplified in this updated version.<br>
|&gt;<br>
|&gt; Comments welcome..<br>
|<o:p></o:p></p>
</div>
<p class=3D"MsoNormal">|MUCH better introduction. Now I feel like I underst=
and the need exactly.<br>
|<br>
|The &quot;Design Considerations&quot; section is still very confusing to m=
e.<br>
|<br>
|&gt; &nbsp; &nbsp;Though ICE specifies STUN Binding indications to be used=
 for<br>
|&gt; &nbsp; &nbsp;keepalives, it requires that an agent be prepared to rec=
eive<br>
|&gt; &nbsp; &nbsp;connectivity check as well. &nbsp;If a connectivity chec=
k is received, a<br>
|&gt; &nbsp; &nbsp;response is generated, but there is no impact on ICE pro=
cessing, as<br>
|&gt; &nbsp; &nbsp;described in section 10 of [RFC5245].<br>
|<br>
|...so? Why is &quot;an impact on ICE processing&quot; necessary?<br>
<br>
Meant to stress these Binding request/response doesn't trigger an ICE resta=
rt..<br>
<br>
|<br>
|&gt; &nbsp; &nbsp;While a WebRTC browser could verify whether the peer con=
tinues to<br>
|&gt; &nbsp; &nbsp;send SRTCP reports before sending traffic to the peer, t=
he usage of<br>
|&gt; &nbsp; &nbsp;SRTCP together with Security Descriptions [RFC4568] requ=
ires exposing<br>
|&gt; &nbsp; &nbsp;the media keys to the JavaScript and renders SRTCP unsui=
table for<br>
|&gt; &nbsp; &nbsp;consent freshness.<br>
|<br>
|Why does it &quot;require exposing the media keys to the JavaScript&quot;?=
 Is this<br>
|because of a law of nature, or is it because of the way the JavaScript<br>
|API is being designed? Could the JS API be changed to accommodate<br>
|SRTCP&#43;SDES?<br>
<br>
That's how the API construct looks like today -- the JavaScript would get a=
n SDP blob from the browser containing the crypto keys used for SDES-SRTP. =
Of course, the browser could hide those keys by putting a &quot;****&quot; =
in SDP -:).<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Uh, then how would they be sent to the other side?<o=
:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">As far as I can tell, with any plausible API SDES re=
quires exposing the keys to JS.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">-Ekr<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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References: <20130715173816.18605.12504.idtracker@ietfa.amsl.com> <E721D8C6A2E1544DB2DEBC313AF54DE224198182@xmb-rcd-x02.cisco.com> <51E513BF.2040405@viagenie.ca> <E721D8C6A2E1544DB2DEBC313AF54DE224199A15@xmb-rcd-x02.cisco.com> <51E536C1.1080507@viagenie.ca> <E721D8C6A2E1544DB2DEBC313AF54DE224199D69@xmb-rcd-x02.cisco.com> <51E544BC.6000608@viagenie.ca> <E54AEADE791D51469F45E7FBB9643915090BC6@SGSIMBX001.nsn-intra.net> <E721D8C6A2E1544DB2DEBC313AF54DE22419B715@xmb-rcd-x02.cisco.com> <E54AEADE791D51469F45E7FBB9643915090BF4@SGSIMBX001.nsn-intra.net> <E721D8C6A2E1544DB2DEBC313AF54DE22419BC0C@xmb-rcd-x02.cisco.com> <E54AEADE791D51469F45E7FBB9643915090C97@SGSIMBX001.nsn-intra.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 2:36 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|My comments inline <Ranjit>
|
|Regards
|Ranjit
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 2:16 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
||If many public STUN servers and call servers that implement STUN or ICEli=
te
||functionalities, are not expected to support
|
|I should have said they have no role to play. It is similar to peer-to-pee=
r ICE connectivity checks --
|those checks don't go through public STUN or call servers at all.
|[Ranjit] But there can be scenarios where RTCWebBrowsers do talk to WebRTC=
 Gateways which could be co-
|located with call handling servers. Then it won't make sense to have this =
feature.

An RTCWeb gateway wouldn't have to do anything fancier than what it already=
 does for ICE connectivity checks.=20

|
|The difference between ICE connectivity checks and consent freshness check=
s is that the former is
|performed before session establishment and the later is performed after se=
ssion establishment.
|[Ranjit] Why do you need to check consent of the peer after session is est=
ablished? Ideally I should
|check consent before session establishment. If the peer does not want to r=
eceive traffic, then session
|itself should not be established.

See draft-ietf-rtcweb-security-arch:

<snip>
4.4.  Communications and Consent Freshness

   From a security perspective, everything from here on in is a little
   anticlimactic:  Alice and Bob exchange data protected by the keys
   negotiated by DTLS.  Because of the security guarantees discussed in
   the previous sections, they know that the communications are
   encrypted and authenticated.

   The one remaining security property we need to establish is "consent
   freshness", i.e., allowing Alice to verify that Bob is still prepared
   to receive her communications so that Alice does not continue to send
   large traffic volumes to entities which went abruptly offline.  ICE
   specifies periodic STUN keepalizes but only if media is not flowing.
   Because the consent issue is more difficult here, we require RTCWeb
   implementations to periodically send keepalives.  As described in
   Section 5.3, these keepalives MUST be based on the consent freshness
   mechanism specified in [I-D.muthu-behave-consent-freshness].  If a
   keepalive fails and no new ICE channels can be established, then the
   session is terminated.
</snip>

|
||then who would have interest to support this functionality?
|
|RTCWEB browsers and browser-like platforms.
|[Ranjit] browser may implement, but it would be very customized one. I do =
not think the intermediate
|entities would want to support.

You seem to be completely missing the very purpose of consent.

|
||Ideally getting consent for receiving traffic or call should be part of s=
ignaling.
||I do not think using STUN is appropriate.
|
|ICE already does that. You think it is inappropriate?
|[Ranjit] I am saying that using STUN for consent check is not appropriate =
as STUN has other dedicated
|functionality to do - like connectivity checks, keep alives, etc.=20

See draft-ietf-rtcweb-security and draft-ietf-rtcweb-security.

|Also in cases of Symmetric NATs where STUN fails, TURN is needed. Then the=
 proposed feature would anyway fail.

The solution is built on top of ICE, so should work across these.

Muthu

|
|Muthu
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 12:16 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Hi Muthu
||
||If many public STUN servers and call servers that implement STUN or ICEli=
te functionalities, are not
||expected to support, then who would have interest to support this functio=
nality? Ideally getting
||consent for receiving traffic or call should be part of signaling. I do n=
ot think using STUN is
||appropriate.
||
||Regards
||Ranjit
||
||
||
||
||-----Original Message-----
||From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
||Sent: Wednesday, July 17, 2013 12:07 PM
||To: Avasarala, Ranjit (NSN - IN/Bangalore)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Ranjit,
||
|||Many publicly available STUN servers or those that are part of Call serv=
ers
|||may not support this.
||
||Public STUN servers and call servers are not expected to support this.
||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough
|||signaling?
||
||No. That signaling between the JS and the web server could be anything (S=
IP or XMPP or proprietary or
||whatever) and the browser has no control over it.
||
||Muthu
||
|||-----Original Message-----
|||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@ns=
n.com]
|||Sent: Wednesday, July 17, 2013 11:19 AM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Hi Muthu
|||
|||Though using STUN request/response may be good for querying about consen=
t, I feel it is overloading
|||the functionality of STUN. Many publicly available STUN servers or those=
 that are part of Call
||servers
|||may not support this.
|||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough signaling?
|||
|||
|||Regards
|||Ranjit
|||
|||
|||
|||-----Original Message-----
|||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf=
 Of ext Simon Perreault
|||Sent: Tuesday, July 16, 2013 6:34 PM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|||> |> |>    Liveness timer: If no packets have been received on the local=
 port in
|||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript tha=
t
|||> |> |>    connectivity has been lost.  The JavaScript application will =
use this
|||> |> |>    notification however it desires (e.g., cease transmitting to =
the
|||> |> |>    remote peer, provide a notification to the user, etc.).
|||> |> |
|||> |> |This seems to me like it will not fulfill the goal set in the abst=
ract:
|||> |> |"to ensure that a malicious JavaScript cannot use the browser as a
|||> |> |platform for launching attacks". If the JavaScript is free to igno=
re the
|||> |> |notification from the browser, then it has no security benefits. I=
f you
|||> |> |want to reach that goal, the browser needs to forcefully stop tran=
smitting.
|||> |>
|||> |> That goal is fulfilled by the consent checks -- the browser would s=
top transmitting everything
||on
|||> |that candidate pair, including liveness checks, if there is a consent=
 failure.
|||> |
|||> |That's not what the draft says. It says that the browser "notifies" t=
he
|||> |JS app. It needs to say that the browser MUST stop sending.
|||>
|||> No. That section is about liveness check and its intention is just not=
ify the JavaScript of a
|||potential connectivity loss. It is when the consent check fails the brow=
ser actually stops sending
|||everything. Does the draft need more text on the distinction between con=
sent and liveness tests?
|||
|||Ah! No, you're right, and the text is already perfectly clear about
|||this. No need to change. I was just confused.
|||
|||> |> |>    When not actively sending traffic on a nominated candidate pa=
ir,
|||> |> |>    performing consent freshness does not serve any purpose from =
a
|||> |> |>    security perspective.
|||> |> |
|||> |> |I don't understand what this means. Why is the "security perspecti=
ve"
|||> |> |important here? Aren't we concerned about keepalives?
|||> |>
|||> |> You mean one could use keepalives (Binding indications) for launchi=
ng attacks, so consent
|||freshness
|||> |would be required for sending them as well?
|||> |
|||> |No.
|||> |
|||> |This is a section about keepalives. I just don't understand this
|||> |sentence, and I don't understand why it talks about security.
|||>
|||> Ok, let me elaborate:
|||> - Consent freshness is not necessary when the browser is not sending a=
ny
|||>   traffic on a candidate pair.
|||> - If the browser is not performing consent freshness on a candidate pa=
ir
|||>   for the above reason, it performs ICE keepalives (or RTP keepalives)=
 to
|||>   refresh NAT bindings.
|||>
|||> Of course, the browser could continue performing consent freshness eve=
n when it is not sending any
|||other traffic on that candidate pair and skip ICE keepalives.
|||
|||Ah, ok I understand with your explanation. It makes sense. There should
|||be a way to reformulate the text to make it clearer.
|||
|||Thanks,
|||Simon
|||_______________________________________________
|||rtcweb mailing list
|||rtcweb@ietf.org
|||https://www.ietf.org/mailman/listinfo/rtcweb

From rajmohanbanavi@gmail.com  Wed Jul 17 11:56:24 2013
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Hi Justin,

I reviewed draft-uberti-behave-turn-rest-00 and have the following comments.

   1. IMHO, calling the API as REST does not seem to be appropriate since
   the API does not follow the REST principles of addressing/operating on a
   resource.
   2. Sec 4.3 last paragraph - "Because the password is derived from the
   USERNAME, successful verification of the MESSAGE-INTEGRITY ensures that the
   username is trustworthy". I believe this validation is taken care of in the
   TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TURN
   client generates a long term key which is computed as => key =
   MD5(username ":" realm ":" SASLprep(password)). This key is used to perform
   hmac on the message. The same is done on the TURN server end as well. If
   the MESSAGE-INTEGRITY is verified, then it implies that the username is
   trustworthy. So why not just generate a TURN password randomly? This
   would avoid the need to have a secret key between the TURN server and the
   webrtc app. Am I missing any scenario here where this extra security is
   required?
   3. sec 4.1 Client - Does it make it more clear if we reword the last
   sentence as - and the "password" value as input to generate the HMAC digest
   key used while calculating MESSAGE-INTEGRITY hash.
   4. Sec 5.1 Revocation - Why is revoking of specific credentials not
   possible? Probably need more clarity on who would try to revoke the
   credentials and under what scenario?
   5. Sec 5.2 Key Rotation - I presume here that the shared secret
   mentioned is the one shared between the TURN server and the webrtc
   application. The TURN password once generated using say secretkey1 and
   user1 will be valid on the TURN server till the expiry timeout (as per
   suggested value of 1 day) value. Why do we then need the TURN server to
   validate the message integrity against 2 secret keys? Wouldn't this
   behavior not contradict the one stated in TURN RFC?

Thanks,
Rajmohan
MindBricks


On Tue, Jul 16, 2013 at 3:22 AM, Justin Uberti <juberti@google.com> wrote:

> I have changed the WG for this draft from RTCWEB to BEHAVE. Many, but not
> all of the comments I received on the RTCWEB mailing list have been
> addressed.
>
> BEHAVE chairs, I would like 10 minutes of agenda time to discuss this
> draft.
>
> ---------- Forwarded message ----------
> From: <internet-drafts@ietf.org>
> Date: Mon, Jul 15, 2013 at 5:49 PM
> Subject: New Version Notification for draft-uberti-behave-turn-rest-00.txt
> To: Justin Uberti <justin@uberti.name>
>
>
>
> A new version of I-D, draft-uberti-behave-turn-rest-00.txt
> has been successfully submitted by Justin Uberti and posted to the
> IETF repository.
>
> Filename:        draft-uberti-behave-turn-rest
> Revision:        00
> Title:           A REST API For Access To TURN Services
> Creation date:   2013-07-15
> Group:           Individual Submission
> Number of pages: 8
> URL:
> http://www.ietf.org/internet-drafts/draft-uberti-behave-turn-rest-00.txt
> Status:
> http://datatracker.ietf.org/doc/draft-uberti-behave-turn-rest
> Htmlized:
> http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
>
>
> Abstract:
>    This document describes a proposed standard REST API for obtaining
>    access to TURN services via ephemeral (i.e. time-limited)
>    credentials.  These credentials are vended by a web service over
>    HTTP, and then supplied to and checked by a TURN server using the
>    standard TURN protocol.  The usage of ephemeral credentials ensures
>    that access to the TURN server can be controlled even if the
>    credentials can be discovered by the user, as is the case in WebRTC
>    where TURN credentials must be specified in Javascript.
>
>
>
>
> The IETF Secretariat
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 

Life is here and now, not yesterday, not tomorrow...!

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<div dir=3D"ltr">Hi Justin,<div><br></div><div><span style=3D"font-family:a=
rial,sans-serif;font-size:13px">I reviewed=A0</span><span style=3D"font-fam=
ily:arial,sans-serif;font-size:13px">draft-uberti-behave-turn-rest-</span><=
span style=3D"font-family:arial,sans-serif;font-size:13px">00</span><font f=
ace=3D"arial, sans-serif">=A0and have the following comments.</font><br>
<ol><li>IMHO, calling the API as REST does not seem to be appropriate since=
 the API does not follow the REST principles of addressing/operating on a r=
esource.</li><li>Sec 4.3 last paragraph - &quot;Because the password is der=
ived from the USERNAME, successful verification of the MESSAGE-INTEGRITY en=
sures that the username is trustworthy&quot;. I believe this validation is =
taken care of in the TURN RFC itself. In order to generate the MESSAGE-INTE=
GRITY, the TURN client generates a long term key which is computed as =3D&g=
t;=A0<span style=3D"color:rgb(0,0,0);font-size:1em">key =3D MD5(username &q=
uot;:&quot; realm &quot;:&quot; SASLprep(password)). This key is used to pe=
rform hmac on the message. The same is done on the TURN server end as well.=
 If the MESSAGE-INTEGRITY is verified, then it implies that the username is=
 trustworthy. So=A0</span>why not just generate a TURN password randomly? T=
his would avoid the need to have a secret key between the TURN server and t=
he webrtc app. Am I missing any scenario here where this extra security is =
required?</li>
<li>sec 4.1 Client - Does it make it more clear if we reword the last sente=
nce as - and the &quot;password&quot; value as input to generate the HMAC d=
igest key used while calculating MESSAGE-INTEGRITY hash.</li><li>Sec 5.1 Re=
vocation - Why is revoking of specific credentials not possible? Probably n=
eed more clarity on who would try to revoke the credentials and under what =
scenario?</li>
<li>Sec 5.2 Key Rotation - I presume here that the shared secret mentioned =
is the one shared between the TURN server and the webrtc application. The T=
URN password once generated using say secretkey1 and user1 will be valid on=
 the TURN server till the expiry timeout (as per suggested value of 1 day) =
value. Why do we then need the TURN server to validate the message integrit=
y against 2 secret keys? Wouldn&#39;t this behavior not contradict the one =
stated in TURN RFC?</li>
</ol>Thanks,<div>Rajmohan</div><div>MindBricks<br></div></div></div><div cl=
ass=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Jul 16, 2013=
 at 3:22 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@=
google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">I have changed the WG for t=
his draft from RTCWEB to BEHAVE. Many, but not all of the comments I receiv=
ed on the RTCWEB mailing list have been addressed.<br>
<div class=3D"gmail_quote"><br><div dir=3D"ltr">BEHAVE chairs, I would like=
 10 minutes of agenda time to discuss this draft.<br>

<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</=
a>&gt;</span><br>




Date: Mon, Jul 15, 2013 at 5:49 PM<br>Subject: New Version Notification for=
 draft-uberti-behave-turn-rest-00.txt<br>To: Justin Uberti &lt;<a href=3D"m=
ailto:justin@uberti.name" target=3D"_blank">justin@uberti.name</a>&gt;<br>


<br><br><br>
A new version of I-D, draft-uberti-behave-turn-rest-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-uberti-behave-turn-rest<br>
Revision: =A0 =A0 =A0 =A000<br>
Title: =A0 =A0 =A0 =A0 =A0 A REST API For Access To TURN Services<br>
Creation date: =A0 2013-07-15<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 8<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-uberti-behave-turn-rest-00.txt" target=3D"_blank">http://www.ietf.or=
g/internet-drafts/draft-uberti-behave-turn-rest-00.txt</a><br>
Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-uberti-behave-turn-rest" target=3D"_blank">http://datatracker.ietf.org/doc=
/draft-uberti-behave-turn-rest</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-uberti=
-behave-turn-rest-00" target=3D"_blank">http://tools.ietf.org/html/draft-ub=
erti-behave-turn-rest-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0This document describes a proposed standard REST API for obtaining<b=
r>
=A0 =A0access to TURN services via ephemeral (i.e. time-limited)<br>
=A0 =A0credentials. =A0These credentials are vended by a web service over<b=
r>
=A0 =A0HTTP, and then supplied to and checked by a TURN server using the<br=
>
=A0 =A0standard TURN protocol. =A0The usage of ephemeral credentials ensure=
s<br>
=A0 =A0that access to the TURN server can be controlled even if the<br>
=A0 =A0credentials can be discovered by the user, as is the case in WebRTC<=
br>
=A0 =A0where TURN credentials must be specified in Javascript.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>
</div><br></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><br>Life=
 is here and now, not yesterday, not tomorrow...!
</div>

--047d7bdca46855c97504e1b9a6dd--

From fippo@goodadvice.pages.de  Wed Jul 17 13:40:09 2013
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Date: Wed, 17 Jul 2013 22:39:34 +0200
From: Philipp Hancke <fippo@goodadvice.pages.de>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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Am 17.07.2013 20:56, schrieb Rajmohan Banavi:
> Hi Justin,
>
> I reviewed draft-uberti-behave-turn-rest-00 and have the following comments.
[...]
>     2. Sec 4.3 last paragraph - "Because the password is derived from the
>     USERNAME, successful verification of the MESSAGE-INTEGRITY ensures that the
>     username is trustworthy". I believe this validation is taken care of in the
>     TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TURN
>     client generates a long term key which is computed as => key =
>     MD5(username ":" realm ":" SASLprep(password)). This key is used to perform
>     hmac on the message. The same is done on the TURN server end as well. If
>     the MESSAGE-INTEGRITY is verified, then it implies that the username is
>     trustworthy.

Let me try to address that with a lenghty example:
The web server returns
password = BASE64(HMAC-SHA1(username, sharedsecret))
to the browser.

For example, with username = "12334939:mbzrxpgjys" and sharedsecret 
"secret", the password given to the browser would be
"+4pCXR06gx/cqAikLYnBajtZW6E=" (hopefully)




The browser passes the username, uri and password to it's webrtc stack.
The password is exposed to the user of the browser (which might be 
anyone surfing your website) during that process.

That stack does long term authentication and uses this password string 
as input for calculating MESSAGE-INTEGRITY with
key = MD5(username ":" realm ":" SASLprep(password))

For the example this would be
key =MD5("12334939:mbzrxpgjys" ":" realm ":" 
SASLprep("+4pCXR06gx/cqAikLYnBajtZW6E=")




On the TURN server, the key for MESSAGE-INTEGRITY is calculated using
key = MD5(username ":" realm ":" SASLprep(BASE64(HMAC-SHA1(USERNAME, 
sharedsecret))

So for the example this would be
  = MD5("12334939:mbzrxpgjys" ":" realm ":" 
SASLprep(BASE64(HMAC-SHA1("12334939:mbzrxpgjys", "secret"))
  = MD5("12334939:mbzrxpgjys" ":" realm ":" 
SASLprep("+4pCXR06gx/cqAikLYnBajtZW6E=")

So the browser and turn server use the same input to the M-I. This can 
be done by looking at the request alone given the shared secret.


Does that make it clearer?

philipp,
who only understood it after implementing it

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Hadriel Kaplan:
>
> Howdy,
> Someone's asked me to explain the compromise I plan on presenting in
> Berlin now, so we have more time to argue over it or chew on it.  The rea=
son
> I was going to wait is because I wanted to explain how I got to the point=
 of
> thinking such a compromise would make sense.  So I'll try to do that in t=
his
> email, which will make this email long.  Nothin' much I can do about that=
.
>=20
> Obviously this is all my personal opinion, not that of my employer, and n=
ot a
> statement claiming to be objective fact, etc.

Likewise, in this reply where, though I like you, I don't like your proposa=
l.

>=20
> Background:
> I think many people would agree that SDES has some benefits compared to
> DTLS for those of us wishing to interface to the SIP world.  Some of thos=
e
> benefits are commercial/cost factors, some are more tangible like reducin=
g
> early media clipping.  Whether you agree with such benefits or not, so lo=
ng as
> WebRTC could support SDES in a sufficiently-secure manner, there'd be les=
s
> concern over having it.  Most of the arguments against SDES have been
> directed at whether it could in fact be "sufficiently secure".  Personall=
y I find
> most of the arguments against it being secure enough to be essentially FU=
D,
> and/or also apply to DTLS-SRTP.

I believe that once the entire system is considered, any system that allows=
 EKT shares the same security properties as one that allows SDES, and one t=
hat does not allow EKT requires a decrypt/re-encrypt phase at the gateway t=
hat is costly.

>=20
> On the flip-side, I think the commercial/cost factor for SDES is not a st=
rong
> cause, because I believe the market will bear whatever the extra cost may
> be. (I wasn't sure of this 2 years ago when this all started, but I'm fai=
rly
> confident of it now)  And I think having only DTLS-SRTP as MTI would have=
 a
> marketing benefit for WebRTC as a technology, which I value highly.

The market might bear the extra cost, but it doesn't mean that it isn't ext=
ra cost (and extra environmental damage through the extra energy consumed).

>=20
> Given those personal feelings, I didn't much care either way, so long as =
a
> decision was made - although I still preferred having SDES to avoid the e=
arly
> media clipping problem.  So when I was going to present on this topic bac=
k in
> the Vancouver meeting, my last slide basically said: "If we support SDES =
and it
> turns out later we were wrong, we can always rescind it".  This was based=
 on
> the notion that browsers get updated all the time, especially for securit=
y
> issues.
>=20
> But later it occurred to me that's actually the Achilles' heel of support=
ing SDES
> in WebRTC, for those of us wanting to gateway this stuff to SIP.  Imagine=
 if it
> turns out we were wrong, and some expected-or-unexpected vulnerability is
> exploited against some large WebRTC service provider, and makes the
> news/slashdot.  Browser vendors would instantly have new code removing
> SDES support, and browsers in devices would be updated virtually overnigh=
t -
> some users may take a long time to upgrade their specific browsers on the=
ir
> specific devices - but many of them would do so within days.  That would =
be
> untenable for a WebRTC service provider.=20

I can't imagine such a scenario that couldn't also happen with DTLS-SRTP, e=
specially DTLS-SRTP + EKT.

Plus there's lots of counterexamples where major sites have been compromise=
d in ways that could have been fixed browser-side, and yet browsers didn't =
bother to update.

And finally, what makes you think that the browser vendors would do somethi=
ng like that if the service really was popular? I guess if Google and Mozil=
la wanted everyone using Skype to switch to Internet Explorer they could pu=
ll such a stunt, but why? (In a hypothetical future where IE has webrtc sup=
port and Skype supports webrtc-equipped browsers as a platform for calling,=
 of course)

> Even the smaller Enterprises take
> a long time to upgrade code on their servers, so even for them changing t=
heir
> gateways to suddenly support DTLS-SRTP would be unrealistic.  Imagine wha=
t
> it would be for larger providers, who might even have to deploy more
> servers to handle the sudden additional overhead of DTLS-SRTP.  Meanwhile
> a growing perc  entage of their users can no longer use the service.  Tha=
t's
> bad mojo.

I argue that it'd never happen.

>=20
> The things that a WebRTC-to-SIP service provider can control, and arguabl=
y
> the much larger use-case for this stuff to begin with, are not really tru=
e
> "browsers" anyway - they're web-based-framework device Apps, provided
> by the service provider to begin with.  I.e., the apps they build using w=
eb
> application framework things like PhoneGap/Cordova, Appcelerator
> Titanium, etc.  I mean using a real Browser is nice for ad-hoc stuff, lik=
e being
> able to click on a "talk to a rep" button on a website, or a webex/meetec=
ho
> type thing; but for real communication "service", whether it be as a
> subscriber of a traditional carrier, or Skype, or an employee of an Enter=
prise,
> or a call-center attendant, or whatever - for those most people would nev=
er
> want to have to open a browser just to receive/make calls.  They'd give y=
ou
> an app to use instead.
>=20
> So it's the app use-case that would benefit the most from SDES, especiall=
y for
> issues like media clipping.  And the app use-case doesn't have the securi=
ty
> concerns nor concerns for unforeseen overnight updates.


And will probably support SDES.

>=20
> The Compromise:
> So given that background, I was planning to propose that the security doc
> keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
> statement that web-based application frameworks SHOULD also support
> SDES. (with text about why and how, etc.)

Disagree. That leads immediately to half (approximately) of the browser eco=
system not supporting SDES. That increases my cost for no true security ben=
efit to anyone.

>=20
> I don't think this is too controversial, because web-based frameworks are
> never beholden to following browser behavior anyway - they're used to
> build a native application, and native applications have a very different
> security/threat model in practice.  They're written for a specific purpos=
e, and
> installed by users from known sources for that known purpose.  Technicall=
y,
> afaik, nothing we do in RTCWEB WG or W3C's WEBRTC group have any
> requirements/mandates for native applications anyway - an app maker
> would just ignore something they don't think applies to them - but I thin=
k
> web-based frameworks do generally try to follow W3C models for Javascript
> APIs, and will likely read the IETF RFCs for the media-layer stuff too.  =
So I
> think having this SHOULD statement would be beneficial.
>=20
> Or if the WG prefers, we could even write a separate doc on what a web-
> based framework maker should consider supporting/not-supporting. (I can
> imagine a few other things they might want to offer that a browser can't)


They might be able to turn ICE off, even!

Matthew Kaufman



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References: <20130715214906.5314.83583.idtracker@ietfa.amsl.com> <CALe60zBA_unaQekMkKwKwKNRPbJjECAtJ9bAV=fv6V6Mdfon6Q@mail.gmail.com> <CAOJ7v-2WGi_fD9mVx+dtZBo+X4-sXxXZFek9mt2cAmrqFCyYMg@mail.gmail.com> <CAJWm+fGBDec_66WMBVhsv5TD8hVzDoOtd5CGs7xAHZqkYtDGBg@mail.gmail.com>
From: Justin Uberti <juberti@google.com>
Date: Wed, 17 Jul 2013 16:51:45 -0400
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To: Rajmohan Banavi <rajmohanbanavi@gmail.com>
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Thanks for your comments. Responses inline.

On Wed, Jul 17, 2013 at 2:56 PM, Rajmohan Banavi
<rajmohanbanavi@gmail.com>wrote:

> Hi Justin,
>
> I reviewed draft-uberti-behave-turn-rest-00 and have the following
> comments.
>
>    1. IMHO, calling the API as REST does not seem to be appropriate since
>    the API does not follow the REST principles of addressing/operating on a
>    resource.
>
> This came up in some earlier rtcweb feedback, but I think it falls within
the definition of REST, as it meets the REST design constraints:
http://en.wikipedia.org/wiki/**Representational_state_**transfer#Constraints<http://en.wikipedia.org/wiki/Representational_state_transfer#Constraints>


>
>    1. Sec 4.3 last paragraph - "Because the password is derived from the
>    USERNAME, successful verification of the MESSAGE-INTEGRITY ensures that the
>    username is trustworthy". I believe this validation is taken care of in the
>    TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TURN
>    client generates a long term key which is computed as => key =
>    MD5(username ":" realm ":" SASLprep(password)). This key is used to perform
>    hmac on the message. The same is done on the TURN server end as well. If
>    the MESSAGE-INTEGRITY is verified, then it implies that the username is
>    trustworthy. So why not just generate a TURN password randomly? This
>    would avoid the need to have a secret key between the TURN server and the
>    webrtc app. Am I missing any scenario here where this extra security is
>    required?
>
> RFC 5766 doesn't make any guarantees regarding the USERNAME attribute. You
know it hasn't been tampered with on the wire, via MESSAGE-INTEGRITY, but
you don't know that the USERNAME value is in fact something that the *web
server* previously gave out.

The construct defines in this draft ties the password to the USERNAME, so
that the client cannot change the USERNAME value and still have
MESSAGE-INTEGRITY validate properly.

This is also why a TURN password can't be generated randomly - the TURN
server wouldn't know what random value to use when computing its own
MESSAGE-INTEGRITY.

>
>    1. sec 4.1 Client - Does it make it more clear if we reword the last
>    sentence as - and the "password" value as input to generate the HMAC digest
>    key used while calculating MESSAGE-INTEGRITY hash.
>
> Yes, thanks.

>
>    1. Sec 5.1 Revocation - Why is revoking of specific credentials not
>    possible? Probably need more clarity on who would try to revoke the
>    credentials and under what scenario?
>
> There is no way to revoke credentials through the REST interface, since
there is no communication path between the web server and the TURN server.

>
>    1. Sec 5.2 Key Rotation - I presume here that the shared secret
>    mentioned is the one shared between the TURN server and the webrtc
>    application. The TURN password once generated using say secretkey1 and
>    user1 will be valid on the TURN server till the expiry timeout (as per
>    suggested value of 1 day) value. Why do we then need the TURN server to
>    validate the message integrity against 2 secret keys? Wouldn't this
>    behavior not contradict the one stated in TURN RFC?
>
> The lifetime of vended credentials may be 1 day, but if key rotation is
performed, sometimes a key rotation will occur during the lifetime of a
credential set. As such, the credentials will always need to be validated
against secretkey1 (old) and secretkey2(current) until 1 day after the key
rotation.

>
>
> Thanks,
> Rajmohan
> MindBricks
>
>
> On Tue, Jul 16, 2013 at 3:22 AM, Justin Uberti <juberti@google.com> wrote:
>
>> I have changed the WG for this draft from RTCWEB to BEHAVE. Many, but not
>> all of the comments I received on the RTCWEB mailing list have been
>> addressed.
>>
>> BEHAVE chairs, I would like 10 minutes of agenda time to discuss this
>> draft.
>>
>> ---------- Forwarded message ----------
>> From: <internet-drafts@ietf.org>
>> Date: Mon, Jul 15, 2013 at 5:49 PM
>> Subject: New Version Notification for draft-uberti-behave-turn-rest-00.txt
>> To: Justin Uberti <justin@uberti.name>
>>
>>
>>
>> A new version of I-D, draft-uberti-behave-turn-rest-00.txt
>> has been successfully submitted by Justin Uberti and posted to the
>> IETF repository.
>>
>> Filename:        draft-uberti-behave-turn-rest
>> Revision:        00
>> Title:           A REST API For Access To TURN Services
>> Creation date:   2013-07-15
>> Group:           Individual Submission
>> Number of pages: 8
>> URL:
>> http://www.ietf.org/internet-drafts/draft-uberti-behave-turn-rest-00.txt
>> Status:
>> http://datatracker.ietf.org/doc/draft-uberti-behave-turn-rest
>> Htmlized:
>> http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
>>
>>
>> Abstract:
>>    This document describes a proposed standard REST API for obtaining
>>    access to TURN services via ephemeral (i.e. time-limited)
>>    credentials.  These credentials are vended by a web service over
>>    HTTP, and then supplied to and checked by a TURN server using the
>>    standard TURN protocol.  The usage of ephemeral credentials ensures
>>    that access to the TURN server can be controlled even if the
>>    credentials can be discovered by the user, as is the case in WebRTC
>>    where TURN credentials must be specified in Javascript.
>>
>>
>>
>>
>> The IETF Secretariat
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> --
>
> Life is here and now, not yesterday, not tomorrow...!
>

--047d7ba97b942a598a04e1bb44fa
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for your comments. Responses inline.<div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Wed, Jul 17, 2013 at 2:56 PM,=
 Rajmohan Banavi <span dir=3D"ltr">&lt;<a href=3D"mailto:rajmohanbanavi@gma=
il.com" target=3D"_blank">rajmohanbanavi@gmail.com</a>&gt;</span> wrote:<br=
>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Hi Justin,<div><br></div><div><span style=
=3D"font-family:arial,sans-serif;font-size:13px">I reviewed=C2=A0</span><sp=
an style=3D"font-family:arial,sans-serif;font-size:13px">draft-uberti-behav=
e-turn-rest-</span><span style=3D"font-family:arial,sans-serif;font-size:13=
px">00</span><font face=3D"arial, sans-serif">=C2=A0and have the following =
comments.</font><br>



<ol><li>IMHO, calling the API as REST does not seem to be appropriate since=
 the API does not follow the REST principles of addressing/operating on a r=
esource.</li></ol></div></div></blockquote><div>This came up in some earlie=
r rtcweb feedback, but I think it falls within the definition of REST<font =
face=3D"arial, sans-serif">, as it meets the REST design constraints:</font=
></div>


<a href=3D"http://en.wikipedia.org/wiki/Representational_state_transfer#Con=
straints" style=3D"font-family:arial,sans-serif;font-size:13px" target=3D"_=
blank">http://en.wikipedia.org/wiki/<u></u>Representational_state_<u></u>tr=
ansfer#Constraints</a><div>


=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex"><div dir=3D"ltr"><div><ol><li>Sec 4.3 last par=
agraph - &quot;Because the password is derived from the USERNAME, successfu=
l verification of the MESSAGE-INTEGRITY ensures that the username is trustw=
orthy&quot;. I believe this validation is taken care of in the TURN RFC its=
elf. In order to generate the MESSAGE-INTEGRITY, the TURN client generates =
a long term key which is computed as =3D&gt;=C2=A0<span style=3D"font-size:=
1em">key =3D MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(passwo=
rd)). This key is used to perform hmac on the message. The same is done on =
the TURN server end as well. If the MESSAGE-INTEGRITY is verified, then it =
implies that the username is trustworthy. So=C2=A0</span>why not just gener=
ate a TURN password randomly? This would avoid the need to have a secret ke=
y between the TURN server and the webrtc app. Am I missing any scenario her=
e where this extra security is required?</li>


</ol></div></div></blockquote><div>RFC 5766 doesn&#39;t make any guarantees=
 regarding the USERNAME attribute. You know it hasn&#39;t been tampered wit=
h on the wire, via MESSAGE-INTEGRITY, but you don&#39;t know that the USERN=
AME value is in fact something that the *web server* previously gave out.=
=C2=A0<br>


</div><div><br></div><div>The construct defines in this draft ties the pass=
word to the USERNAME, so that the client cannot change the USERNAME value a=
nd still have MESSAGE-INTEGRITY validate properly.</div><div><br></div>


<div>This is also why a TURN password can&#39;t be generated randomly - the=
 TURN server wouldn&#39;t know what random value to use when computing its =
own MESSAGE-INTEGRITY.</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,20=
4);border-left-style:solid;padding-left:1ex">


<div dir=3D"ltr"><div><ol>
<li>sec 4.1 Client - Does it make it more clear if we reword the last sente=
nce as - and the &quot;password&quot; value as input to generate the HMAC d=
igest key used while calculating MESSAGE-INTEGRITY hash.</li></ol></div>


</div></blockquote><div>Yes, thanks.=C2=A0</div><blockquote class=3D"gmail_=
quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-=
color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir=
=3D"ltr">


<div><ol><li>Sec 5.1 Revocation - Why is revoking of specific credentials n=
ot possible? Probably need more clarity on who would try to revoke the cred=
entials and under what scenario?</li></ol></div></div></blockquote><div>


There is no way to revoke credentials through the REST interface, since the=
re is no communication path between the web server and the TURN server.=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex">


<div dir=3D"ltr"><div><ol>
<li>Sec 5.2 Key Rotation - I presume here that the shared secret mentioned =
is the one shared between the TURN server and the webrtc application. The T=
URN password once generated using say secretkey1 and user1 will be valid on=
 the TURN server till the expiry timeout (as per suggested value of 1 day) =
value. Why do we then need the TURN server to validate the message integrit=
y against 2 secret keys? Wouldn&#39;t this behavior not contradict the one =
stated in TURN RFC?</li>


</ol></div></div></blockquote><div>The lifetime of vended credentials may b=
e 1 day, but if key rotation is performed, sometimes a key rotation will oc=
cur during the lifetime of a credential set. As such, the credentials will =
always need to be validated against secretkey1 (old) and secretkey2(current=
) until 1 day after the key rotation.</div>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div><ol>
</ol>Thanks,<div>Rajmohan</div><div>MindBricks<br></div></div></div><div cl=
ass=3D"gmail_extra"><br><br><div class=3D"gmail_quote"><div><div>On Tue, Ju=
l 16, 2013 at 3:22 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> w=
rote:<br>



</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex"><div><div><div dir=3D"ltr">I have changed the =
WG for this draft from RTCWEB to BEHAVE. Many, but not all of the comments =
I received on the RTCWEB mailing list have been addressed.<br>



<div class=3D"gmail_quote"><br><div dir=3D"ltr">BEHAVE chairs, I would like=
 10 minutes of agenda time to discuss this draft.<br>

<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</=
a>&gt;</span><br>







Date: Mon, Jul 15, 2013 at 5:49 PM<br>Subject: New Version Notification for=
 draft-uberti-behave-turn-rest-00.txt<br>To: Justin Uberti &lt;<a href=3D"m=
ailto:justin@uberti.name" target=3D"_blank">justin@uberti.name</a>&gt;<br>





<br><br><br>
A new version of I-D, draft-uberti-behave-turn-rest-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Filename: =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-uberti-behave-turn-rest<br>
Revision: =C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 A REST API For Access To TURN Ser=
vices<br>
Creation date: =C2=A0 2013-07-15<br>
Group: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Number of pages: 8<br>
URL: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.o=
rg/internet-drafts/draft-uberti-behave-turn-rest-00.txt" target=3D"_blank">=
http://www.ietf.org/internet-drafts/draft-uberti-behave-turn-rest-00.txt</a=
><br>
Status: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://datatracker.iet=
f.org/doc/draft-uberti-behave-turn-rest" target=3D"_blank">http://datatrack=
er.ietf.org/doc/draft-uberti-behave-turn-rest</a><br>
Htmlized: =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/=
draft-uberti-behave-turn-rest-00" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-uberti-behave-turn-rest-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes a proposed standard REST API for obtai=
ning<br>
=C2=A0 =C2=A0access to TURN services via ephemeral (i.e. time-limited)<br>
=C2=A0 =C2=A0credentials. =C2=A0These credentials are vended by a web servi=
ce over<br>
=C2=A0 =C2=A0HTTP, and then supplied to and checked by a TURN server using =
the<br>
=C2=A0 =C2=A0standard TURN protocol. =C2=A0The usage of ephemeral credentia=
ls ensures<br>
=C2=A0 =C2=A0that access to the TURN server can be controlled even if the<b=
r>
=C2=A0 =C2=A0credentials can be discovered by the user, as is the case in W=
ebRTC<br>
=C2=A0 =C2=A0where TURN credentials must be specified in Javascript.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>
</div><br></div>
<br></div></div><div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></div></blockquote></div><span><font color=3D"#888888"><br><br clear=3D=
"all"><div><br></div>-- <br><br>Life is here and now, not yesterday, not to=
morrow...!
</font></span></div>
</blockquote></div><br></div></div>

--047d7ba97b942a598a04e1bb44fa--

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Thanks for the detailed example. I agree with the entire flow you have
given. My only concern is - why should the ephemeral password be generated
by TURN server using HMAC SHA on then username string? Why can't this be
some random unique string?

Taking your example -

Suppose TURN server generates credentials, username = "12334939:mbzrxpgjys"
and password ="+somerandomstring+". This is a randomly generated string and
passed onto webrtc app. The TURN server also stores these credentials for
later validation.
var iceServer = {
     "username": 12334939:mbzrxpgjys,
     "credential": +somerandomstring+,
   };

These credentials are passed from JS to the webrtc client stack in the
browser.

On Browser WebTRC Client.
key = MD5(12334939:mbzrxpgjys ":" realm ":" +somerandomstring+)
Similarly on the TURN server, the key for MESSAGE-INTEGRITY is calculated
using
key = MD5(12334939:mbzrxpgjys ":" realm ":" +somerandomstring+)

So the browser and turn server use the same input to the M-I. The one
implementation related advantage I see with this way of generation of TURN
password using HMAC is that the TURN server need not really store the HMAC
generated password, but can calculate the same using the USERNAME attribute
in the received TURN request.

In fact, I do not see why the sharedsecret needs to be shared between the
TURN server and the WebRTC application. The sharedsecret key "secret" is
only required by the TURN server for validating the received TURN ALLOCATE
requests once it has generated and sent out the ephemeral credentials. What
does the WebRTC application do with the shared secret key?

Thanks,
Rajmohan


On Thu, Jul 18, 2013 at 2:09 AM, Philipp Hancke
<fippo@goodadvice.pages.de>wrote:

> Am 17.07.2013 20:56, schrieb Rajmohan Banavi:
>
>  Hi Justin,
>>
>> I reviewed draft-uberti-behave-turn-rest-**00 and have the following
>> comments.
>>
> [...]
>
>>     2. Sec 4.3 last paragraph - "Because the password is derived from the
>>
>>     USERNAME, successful verification of the MESSAGE-INTEGRITY ensures
>> that the
>>     username is trustworthy". I believe this validation is taken care of
>> in the
>>     TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TURN
>>     client generates a long term key which is computed as => key =
>>     MD5(username ":" realm ":" SASLprep(password)). This key is used to
>> perform
>>     hmac on the message. The same is done on the TURN server end as well.
>> If
>>     the MESSAGE-INTEGRITY is verified, then it implies that the username
>> is
>>     trustworthy.
>>
>
> Let me try to address that with a lenghty example:
> The web server returns
> password = BASE64(HMAC-SHA1(username, sharedsecret))
> to the browser.
>
> For example, with username = "12334939:mbzrxpgjys" and sharedsecret
> "secret", the password given to the browser would be
> "+4pCXR06gx/cqAikLYnBajtZW6E=" (hopefully)
>
>
>
>
> The browser passes the username, uri and password to it's webrtc stack.
> The password is exposed to the user of the browser (which might be anyone
> surfing your website) during that process.
>
> That stack does long term authentication and uses this password string as
> input for calculating MESSAGE-INTEGRITY with
> key = MD5(username ":" realm ":" SASLprep(password))
>
> For the example this would be
> key =MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep("+4pCXR06gx/**
> cqAikLYnBajtZW6E=")
>
>
>
>
> On the TURN server, the key for MESSAGE-INTEGRITY is calculated using
> key = MD5(username ":" realm ":" SASLprep(BASE64(HMAC-SHA1(**USERNAME,
> sharedsecret))
>
> So for the example this would be
>  = MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep(BASE64(HMAC-SHA1("**12334939:mbzrxpgjys",
> "secret"))
>  = MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep("+4pCXR06gx/**
> cqAikLYnBajtZW6E=")
>
> So the browser and turn server use the same input to the M-I. This can be
> done by looking at the request alone given the shared secret.
>
>
> Does that make it clearer?
>
> philipp,
> who only understood it after implementing it
>



-- 

Life is here and now, not yesterday, not tomorrow...!

--047d7bdc10e47923a204e1bb811b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for the detailed example. I agree with the entire f=
low you have given. My only concern is - why should the ephemeral password =
be generated by TURN server using HMAC SHA on then username string? Why can=
&#39;t this be some random unique string?<div>
<br></div><div style>Taking your example -=A0</div><div style><br></div><di=
v style><span style=3D"font-family:arial,sans-serif;font-size:13px">Suppose=
 TURN server generates credentials, username =3D &quot;12334939:mbzrxpgjys&=
quot; and password =3D</span><span style=3D"font-family:arial,sans-serif;fo=
nt-size:13px">&quot;+somerandomstring+&quot;. This is a randomly generated =
string and passed onto webrtc app. The TURN server also stores these creden=
tials for later validation.</span><br style=3D"font-family:arial,sans-serif=
;font-size:13px">
<div><font face=3D"arial, sans-serif">var iceServer =3D {</font></div><div>=
<font face=3D"arial, sans-serif">=A0 =A0 =A0&quot;username&quot;:=A0</font>=
<span style=3D"font-family:arial,sans-serif;font-size:13px">12334939:mbzrxp=
gjys</span><font face=3D"arial, sans-serif">,</font></div>
<div><font face=3D"arial, sans-serif">=A0 =A0 =A0&quot;credential&quot;:=A0=
</font><span style=3D"font-family:arial,sans-serif;font-size:13px">+someran=
domstring+</span><font face=3D"arial, sans-serif">,</font></div><div><span =
style=3D"font-family:arial,sans-serif">=A0 =A0};</span><br>
</div><div><font face=3D"arial, sans-serif"><br></font></div><font face=3D"=
arial, sans-serif">These credentials are passed from JS to the webrtc clien=
t stack in the browser.</font><br style=3D"font-family:arial,sans-serif;fon=
t-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span></di=
v><div style><span style=3D"font-family:arial,sans-serif;font-size:13px">On=
 Browser WebTRC Client.=A0</span></div><div style><span style=3D"font-size:=
13px;font-family:arial,sans-serif">key =3D MD5(</span><span style=3D"font-f=
amily:arial,sans-serif;font-size:13px">12334939:mbzrxpgjys</span><span styl=
e=3D"font-size:13px;font-family:arial,sans-serif">=A0&quot;:&quot; realm &q=
uot;:&quot;=A0</span><span style=3D"font-family:arial,sans-serif;font-size:=
13px">+somerandomstring+</span><span style=3D"font-family:arial,sans-serif;=
font-size:13px">)</span></div>
<div style><span style=3D"font-family:arial,sans-serif;font-size:13px">Simi=
larly on the TURN server, the key for MESSAGE-INTEGRITY is calculated using=
</span><br style=3D"font-family:arial,sans-serif;font-size:13px"><span styl=
e=3D"font-family:arial,sans-serif;font-size:13px">key =3D MD5(</span><span =
style=3D"font-family:arial,sans-serif;font-size:13px">12334939:mbzrxpgjys</=
span><span style=3D"font-family:arial,sans-serif;font-size:13px">=A0&quot;:=
&quot; realm &quot;:&quot;=A0</span><span style=3D"font-family:arial,sans-s=
erif;font-size:13px">+somerandomstring+</span><span style=3D"font-family:ar=
ial,sans-serif;font-size:13px">)</span><br style=3D"font-family:arial,sans-=
serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">So the browser and turn server u=
se the same input to the M-I. The one implementation related advantage I se=
e with this way of generation of TURN password using HMAC is that the TURN =
server need not really store the HMAC generated password, but can calculate=
 the same using the USERNAME attribute in the received TURN request.</span>=
</div>
<div style><span style=3D"font-family:arial,sans-serif;font-size:13px"><br>=
</span></div><div style><span style=3D"font-family:arial,sans-serif;font-si=
ze:13px">In fact, I do not see why the sharedsecret needs to be shared betw=
een the TURN server and the WebRTC application. The sharedsecret key &quot;=
secret&quot; is only required by the TURN server for validating the receive=
d TURN ALLOCATE requests once it has generated and sent out the ephemeral c=
redentials. What does the WebRTC application do with the shared secret key?=
</span><br>
</div><div style><span style=3D"font-family:arial,sans-serif;font-size:13px=
"><br></span></div><div style><span style=3D"font-family:arial,sans-serif;f=
ont-size:13px">Thanks,</span></div><div style><span style=3D"font-family:ar=
ial,sans-serif;font-size:13px">Rajmohan</span></div>
</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Jul 18, 2013 at 2:09 AM, Philipp Hancke <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:fippo@goodadvice.pages.de" target=3D"_blank">fippo@goodadvice.pages.d=
e</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Am 17.07.2013 20:56, schrieb Rajmohan Banavi=
:<div class=3D"im"><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Justin,<br>
<br>
I reviewed draft-uberti-behave-turn-rest-<u></u>00 and have the following c=
omments.<br>
</blockquote></div>
[...]<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=A0 =A0 2. Sec 4.3 last paragraph - &quot;Because the password is derived f=
rom the<div class=3D"im"><br>
=A0 =A0 USERNAME, successful verification of the MESSAGE-INTEGRITY ensures =
that the<br>
=A0 =A0 username is trustworthy&quot;. I believe this validation is taken c=
are of in the<br>
=A0 =A0 TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TU=
RN<br>
=A0 =A0 client generates a long term key which is computed as =3D&gt; key =
=3D<br>
=A0 =A0 MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(password)).=
 This key is used to perform<br>
=A0 =A0 hmac on the message. The same is done on the TURN server end as wel=
l. If<br>
=A0 =A0 the MESSAGE-INTEGRITY is verified, then it implies that the usernam=
e is<br>
=A0 =A0 trustworthy.<br>
</div></blockquote>
<br>
Let me try to address that with a lenghty example:<br>
The web server returns<br>
password =3D BASE64(HMAC-SHA1(username, sharedsecret))<br>
to the browser.<br>
<br>
For example, with username =3D &quot;12334939:mbzrxpgjys&quot; and sharedse=
cret &quot;secret&quot;, the password given to the browser would be<br>
&quot;+4pCXR06gx/cqAikLYnBajtZW6E=3D&quot; (hopefully)<br>
<br>
<br>
<br>
<br>
The browser passes the username, uri and password to it&#39;s webrtc stack.=
<br>
The password is exposed to the user of the browser (which might be anyone s=
urfing your website) during that process.<br>
<br>
That stack does long term authentication and uses this password string as i=
nput for calculating MESSAGE-INTEGRITY with<br>
key =3D MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(password))<=
br>
<br>
For the example this would be<br>
key =3DMD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&quot=
; SASLprep(&quot;+4pCXR06gx/<u></u>cqAikLYnBajtZW6E=3D&quot;)<br>
<br>
<br>
<br>
<br>
On the TURN server, the key for MESSAGE-INTEGRITY is calculated using<br>
key =3D MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(BASE64(HMAC=
-SHA1(<u></u>USERNAME, sharedsecret))<br>
<br>
So for the example this would be<br>
=A0=3D MD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&quot=
; SASLprep(BASE64(HMAC-SHA1(&quot;<u></u>12334939:mbzrxpgjys&quot;, &quot;s=
ecret&quot;))<br>
=A0=3D MD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&quot=
; SASLprep(&quot;+4pCXR06gx/<u></u>cqAikLYnBajtZW6E=3D&quot;)<br>
<br>
So the browser and turn server use the same input to the M-I. This can be d=
one by looking at the request alone given the shared secret.<br>
<br>
<br>
Does that make it clearer?<br>
<br>
philipp,<br>
who only understood it after implementing it<br>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><br>Life is =
here and now, not yesterday, not tomorrow...!
</div>

--047d7bdc10e47923a204e1bb811b--

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To: Ted Hardie <ted.ietf@gmail.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Thread-Topic: [rtcweb] Locus of API discussion
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Ted Hardie:
	Howdy,

	The recent set of API discussions has been spread across both the rtcweb a=
nd public-webrtc mailing lists.=A0 That's making it both harder to follow a=
nd harder for folks to work
	out who is saying what under which rules.=A0 The chairs of both groups bel=
ieve that the right place for the discussion to continue should be public-w=
ebrtc.=A0 Please direct follow-
	ups on this topic to that list.


Does that include moving the SDP "API surface" discussions and decisions to=
 the public-webrtc mailing lists?=20

I am frankly quite concerned given earlier comments that yes, the W3C had g=
iven responsibility for "JSEP" and the resulting SDP work to IETF, and so n=
ow we have the IETF WG making decisions but the discussion happening on the=
 W3C list -- which means there's no "paper trail" prior to the IETF meeting=
 on an IETF list about how the participants have weighed in -- and then con=
versely when the W3C spec gets to last call there will be no "paper trail" =
at the W3C showing how decisions had been reached about the SDP "API", as t=
hat happened over in the MMUSIC WG, which list was not even copied on this =
message.

Matthew Kaufman


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On Wed, Jul 17, 2013 at 3:48 PM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

> Hadriel Kaplan:
> >
> > Howdy,
> > Someone's asked me to explain the compromise I plan on presenting in
> > Berlin now, so we have more time to argue over it or chew on it.  The
> reason
> > I was going to wait is because I wanted to explain how I got to the
> point of
> > thinking such a compromise would make sense.  So I'll try to do that in
> this
> > email, which will make this email long.  Nothin' much I can do about
> that.
> >
> > Obviously this is all my personal opinion, not that of my employer, and
> not a
> > statement claiming to be objective fact, etc.
>
> Likewise, in this reply where, though I like you, I don't like your
> proposal.
>
> >
> > Background:
> > I think many people would agree that SDES has some benefits compared to
> > DTLS for those of us wishing to interface to the SIP world.  Some of
> those
> > benefits are commercial/cost factors, some are more tangible like
> reducing
> > early media clipping.  Whether you agree with such benefits or not, so
> long as
> > WebRTC could support SDES in a sufficiently-secure manner, there'd be
> less
> > concern over having it.  Most of the arguments against SDES have been
> > directed at whether it could in fact be "sufficiently secure".
>  Personally I find
> > most of the arguments against it being secure enough to be essentially
> FUD,
> > and/or also apply to DTLS-SRTP.
>
> I believe that once the entire system is considered, any system that
> allows EKT shares the same security properties as one that allows SDES, and
> one that does not allow EKT requires a decrypt/re-encrypt phase at the
> gateway that is costly.
>
> >
> > On the flip-side, I think the commercial/cost factor for SDES is not a
> strong
> > cause, because I believe the market will bear whatever the extra cost may
> > be. (I wasn't sure of this 2 years ago when this all started, but I'm
> fairly
> > confident of it now)  And I think having only DTLS-SRTP as MTI would
> have a
> > marketing benefit for WebRTC as a technology, which I value highly.
>
> The market might bear the extra cost, but it doesn't mean that it isn't
> extra cost (and extra environmental damage through the extra energy
> consumed).
>
> >
> > Given those personal feelings, I didn't much care either way, so long as
> a
> > decision was made - although I still preferred having SDES to avoid the
> early
> > media clipping problem.  So when I was going to present on this topic
> back in
> > the Vancouver meeting, my last slide basically said: "If we support SDES
> and it
> > turns out later we were wrong, we can always rescind it".  This was
> based on
> > the notion that browsers get updated all the time, especially for
> security
> > issues.
> >
> > But later it occurred to me that's actually the Achilles' heel of
> supporting SDES
> > in WebRTC, for those of us wanting to gateway this stuff to SIP.
>  Imagine if it
> > turns out we were wrong, and some expected-or-unexpected vulnerability is
> > exploited against some large WebRTC service provider, and makes the
> > news/slashdot.  Browser vendors would instantly have new code removing
> > SDES support, and browsers in devices would be updated virtually
> overnight -
> > some users may take a long time to upgrade their specific browsers on
> their
> > specific devices - but many of them would do so within days.  That would
> be
> > untenable for a WebRTC service provider.
>
> I can't imagine such a scenario that couldn't also happen with DTLS-SRTP,
> especially DTLS-SRTP + EKT.
>
> Plus there's lots of counterexamples where major sites have been
> compromised in ways that could have been fixed browser-side, and yet
> browsers didn't bother to update.
>
> And finally, what makes you think that the browser vendors would do
> something like that if the service really was popular? I guess if Google
> and Mozilla wanted everyone using Skype to switch to Internet Explorer they
> could pull such a stunt, but why? (In a hypothetical future where IE has
> webrtc support and Skype supports webrtc-equipped browsers as a platform
> for calling, of course)
>
> > Even the smaller Enterprises take
> > a long time to upgrade code on their servers, so even for them changing
> their
> > gateways to suddenly support DTLS-SRTP would be unrealistic.  Imagine
> what
> > it would be for larger providers, who might even have to deploy more
> > servers to handle the sudden additional overhead of DTLS-SRTP.  Meanwhile
> > a growing perc  entage of their users can no longer use the service.
>  That's
> > bad mojo.
>
> I argue that it'd never happen.
>
> >
> > The things that a WebRTC-to-SIP service provider can control, and
> arguably
> > the much larger use-case for this stuff to begin with, are not really
> true
> > "browsers" anyway - they're web-based-framework device Apps, provided
> > by the service provider to begin with.  I.e., the apps they build using
> web
> > application framework things like PhoneGap/Cordova, Appcelerator
> > Titanium, etc.  I mean using a real Browser is nice for ad-hoc stuff,
> like being
> > able to click on a "talk to a rep" button on a website, or a
> webex/meetecho
> > type thing; but for real communication "service", whether it be as a
> > subscriber of a traditional carrier, or Skype, or an employee of an
> Enterprise,
> > or a call-center attendant, or whatever - for those most people would
> never
> > want to have to open a browser just to receive/make calls.  They'd give
> you
> > an app to use instead.
> >
> > So it's the app use-case that would benefit the most from SDES,
> especially for
> > issues like media clipping.  And the app use-case doesn't have the
> security
> > concerns nor concerns for unforeseen overnight updates.
>
>
> And will probably support SDES.
>
> >
> > The Compromise:
> > So given that background, I was planning to propose that the security doc
> > keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
> > statement that web-based application frameworks SHOULD also support
> > SDES. (with text about why and how, etc.)
>
> Disagree. That leads immediately to half (approximately) of the browser
> ecosystem not supporting SDES. That increases my cost for no true security
> benefit to anyone.
>
> >
> > I don't think this is too controversial, because web-based frameworks are
> > never beholden to following browser behavior anyway - they're used to
> > build a native application, and native applications have a very different
> > security/threat model in practice.  They're written for a specific
> purpose, and
> > installed by users from known sources for that known purpose.
>  Technically,
> > afaik, nothing we do in RTCWEB WG or W3C's WEBRTC group have any
> > requirements/mandates for native applications anyway - an app maker
> > would just ignore something they don't think applies to them - but I
> think
> > web-based frameworks do generally try to follow W3C models for Javascript
> > APIs, and will likely read the IETF RFCs for the media-layer stuff too.
>  So I
> > think having this SHOULD statement would be beneficial.
> >
> > Or if the WG prefers, we could even write a separate doc on what a web-
> > based framework maker should consider supporting/not-supporting. (I can
> > imagine a few other things they might want to offer that a browser can't)
>
>
> They might be able to turn ICE off, even!
>

Amen.  I think ICE will be the first barrier to initiating communications
with legacy equipment anyways... Anybody that can get past the ICE step
will then be allowed to deal with the problems of the mandatorily included
encryption mechanisms as a next step.

I don't know if there is very much "legacy" equipment that will be able to
interop with WebRTC media sessions without some sort of translation.


Matthew Fredrickson

--001a11c27cacf1530004e1bbfb05
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">On Wed, Jul 17, 2013 at 3:48 PM, Matthew Kaufman (SKYPE) <=
span dir=3D"ltr">&lt;<a href=3D"mailto:matthew.kaufman@skype.net" target=3D=
"_blank">matthew.kaufman@skype.net</a>&gt;</span> wrote:<br><div class=3D"g=
mail_extra">
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hadriel Kaplan:<b=
r>
<div class=3D"im">&gt;<br>
&gt; Howdy,<br>
&gt; Someone&#39;s asked me to explain the compromise I plan on presenting =
in<br>
&gt; Berlin now, so we have more time to argue over it or chew on it. =A0Th=
e reason<br>
&gt; I was going to wait is because I wanted to explain how I got to the po=
int of<br>
&gt; thinking such a compromise would make sense. =A0So I&#39;ll try to do =
that in this<br>
&gt; email, which will make this email long. =A0Nothin&#39; much I can do a=
bout that.<br>
&gt;<br>
&gt; Obviously this is all my personal opinion, not that of my employer, an=
d not a<br>
&gt; statement claiming to be objective fact, etc.<br>
<br>
</div>Likewise, in this reply where, though I like you, I don&#39;t like yo=
ur proposal.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; Background:<br>
&gt; I think many people would agree that SDES has some benefits compared t=
o<br>
&gt; DTLS for those of us wishing to interface to the SIP world. =A0Some of=
 those<br>
&gt; benefits are commercial/cost factors, some are more tangible like redu=
cing<br>
&gt; early media clipping. =A0Whether you agree with such benefits or not, =
so long as<br>
&gt; WebRTC could support SDES in a sufficiently-secure manner, there&#39;d=
 be less<br>
&gt; concern over having it. =A0Most of the arguments against SDES have bee=
n<br>
&gt; directed at whether it could in fact be &quot;sufficiently secure&quot=
;. =A0Personally I find<br>
&gt; most of the arguments against it being secure enough to be essentially=
 FUD,<br>
&gt; and/or also apply to DTLS-SRTP.<br>
<br>
</div>I believe that once the entire system is considered, any system that =
allows EKT shares the same security properties as one that allows SDES, and=
 one that does not allow EKT requires a decrypt/re-encrypt phase at the gat=
eway that is costly.<br>

<div class=3D"im"><br>
&gt;<br>
&gt; On the flip-side, I think the commercial/cost factor for SDES is not a=
 strong<br>
&gt; cause, because I believe the market will bear whatever the extra cost =
may<br>
&gt; be. (I wasn&#39;t sure of this 2 years ago when this all started, but =
I&#39;m fairly<br>
&gt; confident of it now) =A0And I think having only DTLS-SRTP as MTI would=
 have a<br>
&gt; marketing benefit for WebRTC as a technology, which I value highly.<br=
>
<br>
</div>The market might bear the extra cost, but it doesn&#39;t mean that it=
 isn&#39;t extra cost (and extra environmental damage through the extra ene=
rgy consumed).<br>
<div class=3D"im"><br>
&gt;<br>
&gt; Given those personal feelings, I didn&#39;t much care either way, so l=
ong as a<br>
&gt; decision was made - although I still preferred having SDES to avoid th=
e early<br>
&gt; media clipping problem. =A0So when I was going to present on this topi=
c back in<br>
&gt; the Vancouver meeting, my last slide basically said: &quot;If we suppo=
rt SDES and it<br>
&gt; turns out later we were wrong, we can always rescind it&quot;. =A0This=
 was based on<br>
&gt; the notion that browsers get updated all the time, especially for secu=
rity<br>
&gt; issues.<br>
&gt;<br>
&gt; But later it occurred to me that&#39;s actually the Achilles&#39; heel=
 of supporting SDES<br>
&gt; in WebRTC, for those of us wanting to gateway this stuff to SIP. =A0Im=
agine if it<br>
&gt; turns out we were wrong, and some expected-or-unexpected vulnerability=
 is<br>
&gt; exploited against some large WebRTC service provider, and makes the<br=
>
&gt; news/slashdot. =A0Browser vendors would instantly have new code removi=
ng<br>
&gt; SDES support, and browsers in devices would be updated virtually overn=
ight -<br>
&gt; some users may take a long time to upgrade their specific browsers on =
their<br>
&gt; specific devices - but many of them would do so within days. =A0That w=
ould be<br>
&gt; untenable for a WebRTC service provider.<br>
<br>
</div>I can&#39;t imagine such a scenario that couldn&#39;t also happen wit=
h DTLS-SRTP, especially DTLS-SRTP + EKT.<br>
<br>
Plus there&#39;s lots of counterexamples where major sites have been compro=
mised in ways that could have been fixed browser-side, and yet browsers did=
n&#39;t bother to update.<br>
<br>
And finally, what makes you think that the browser vendors would do somethi=
ng like that if the service really was popular? I guess if Google and Mozil=
la wanted everyone using Skype to switch to Internet Explorer they could pu=
ll such a stunt, but why? (In a hypothetical future where IE has webrtc sup=
port and Skype supports webrtc-equipped browsers as a platform for calling,=
 of course)<br>

<div class=3D"im"><br>
&gt; Even the smaller Enterprises take<br>
&gt; a long time to upgrade code on their servers, so even for them changin=
g their<br>
&gt; gateways to suddenly support DTLS-SRTP would be unrealistic. =A0Imagin=
e what<br>
&gt; it would be for larger providers, who might even have to deploy more<b=
r>
&gt; servers to handle the sudden additional overhead of DTLS-SRTP. =A0Mean=
while<br>
&gt; a growing perc =A0entage of their users can no longer use the service.=
 =A0That&#39;s<br>
&gt; bad mojo.<br>
<br>
</div>I argue that it&#39;d never happen.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; The things that a WebRTC-to-SIP service provider can control, and argu=
ably<br>
&gt; the much larger use-case for this stuff to begin with, are not really =
true<br>
&gt; &quot;browsers&quot; anyway - they&#39;re web-based-framework device A=
pps, provided<br>
&gt; by the service provider to begin with. =A0I.e., the apps they build us=
ing web<br>
&gt; application framework things like PhoneGap/Cordova, Appcelerator<br>
&gt; Titanium, etc. =A0I mean using a real Browser is nice for ad-hoc stuff=
, like being<br>
&gt; able to click on a &quot;talk to a rep&quot; button on a website, or a=
 webex/meetecho<br>
&gt; type thing; but for real communication &quot;service&quot;, whether it=
 be as a<br>
&gt; subscriber of a traditional carrier, or Skype, or an employee of an En=
terprise,<br>
&gt; or a call-center attendant, or whatever - for those most people would =
never<br>
&gt; want to have to open a browser just to receive/make calls. =A0They&#39=
;d give you<br>
&gt; an app to use instead.<br>
&gt;<br>
&gt; So it&#39;s the app use-case that would benefit the most from SDES, es=
pecially for<br>
&gt; issues like media clipping. =A0And the app use-case doesn&#39;t have t=
he security<br>
&gt; concerns nor concerns for unforeseen overnight updates.<br>
<br>
<br>
</div>And will probably support SDES.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; The Compromise:<br>
&gt; So given that background, I was planning to propose that the security =
doc<br>
&gt; keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a<br=
>
&gt; statement that web-based application frameworks SHOULD also support<br=
>
&gt; SDES. (with text about why and how, etc.)<br>
<br>
</div>Disagree. That leads immediately to half (approximately) of the brows=
er ecosystem not supporting SDES. That increases my cost for no true securi=
ty benefit to anyone.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; I don&#39;t think this is too controversial, because web-based framewo=
rks are<br>
&gt; never beholden to following browser behavior anyway - they&#39;re used=
 to<br>
&gt; build a native application, and native applications have a very differ=
ent<br>
&gt; security/threat model in practice. =A0They&#39;re written for a specif=
ic purpose, and<br>
&gt; installed by users from known sources for that known purpose. =A0Techn=
ically,<br>
&gt; afaik, nothing we do in RTCWEB WG or W3C&#39;s WEBRTC group have any<b=
r>
&gt; requirements/mandates for native applications anyway - an app maker<br=
>
&gt; would just ignore something they don&#39;t think applies to them - but=
 I think<br>
&gt; web-based frameworks do generally try to follow W3C models for Javascr=
ipt<br>
&gt; APIs, and will likely read the IETF RFCs for the media-layer stuff too=
. =A0So I<br>
&gt; think having this SHOULD statement would be beneficial.<br>
&gt;<br>
&gt; Or if the WG prefers, we could even write a separate doc on what a web=
-<br>
&gt; based framework maker should consider supporting/not-supporting. (I ca=
n<br>
&gt; imagine a few other things they might want to offer that a browser can=
&#39;t)<br>
<br>
<br>
</div>They might be able to turn ICE off, even!<br></blockquote><div><br></=
div><div>Amen. =A0I think ICE will be the first barrier to initiating commu=
nications with legacy equipment anyways... Anybody that can get past the IC=
E step will then be allowed to deal with the problems of the mandatorily in=
cluded encryption mechanisms as a next step.</div>
<div><br></div><div>I don&#39;t know if there is very much &quot;legacy&quo=
t; equipment that will be able to interop with WebRTC media sessions withou=
t some sort of translation.</div><div><br></div><div><br></div><div>Matthew=
 Fredrickson</div>
</div></div></div>

--001a11c27cacf1530004e1bbfb05--

From matthew.kaufman@skype.net  Wed Jul 17 14:59:53 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Bernard Aboba <bernard_aboba@hotmail.com>, =?utf-8?B?U3RlZmFuIEjDpWthbnNzb24gTEs=?= <stefan.lk.hakansson@ericsson.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Date: Wed, 17 Jul 2013 21:58:37 +0000
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the	current WebRTC API and SDP as a control surface
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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 18 Jul 2013 05:59:40 +0800
Message-ID: <CABcZeBN4i=Nbiybksg8QrOeH+GYwNKeqUqQuxM-O40dtPkqx4A@mail.gmail.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Cc: "behave@ietf.org" <behave@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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--047d7bb03e6a41619d04e1bc38e6
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On Wed, Jul 17, 2013 at 8:47 PM, Muthu Arul Mozhi Perumal (mperumal) <
mperumal@cisco.com> wrote:

>  |Uh, then how would they be sent to the other side?****
>
> ** **
>
> Ah, the browser would have to modify the SDP when it is sent on the wire.=
.
>

I don't really see how this helps since the server is going to see it and
call tell
the JS (remember, that's where the JS came from).

-Ekr


>
> |As far as I can tell, with any plausible API SDES requires exposing the
> keys to JS.****
>
> ** **
>
> That helps..thanks.****
>
> ** **
>
> Muthu****
>
> ** **
>
> *From:* Eric Rescorla [mailto:ekr@rtfm.com]
> *Sent:* Wednesday, July 17, 2013 2:32 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* Simon Perreault; behave@ietf.org; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] [BEHAVE] FW: I-D Action:
> draft-muthu-behave-consent-freshness-04.txt****
>
> ** **
>
> ** **
>
> ** **
>
> On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Perumal (mperumal) <
> mperumal@cisco.com> wrote:****
>
> [Added rtcweb since I am not sure if everyone involved there are followin=
g
> this discussion in behave]
>
> Thanks for the review. See inline..
>
> |-----Original Message-----
> |From: behave-bounces@ietf.org [mailto:behave-bounces@ietf.org] On Behalf
> Of Simon Perreault
> |Sent: Tuesday, July 16, 2013 3:05 PM
> |To: behave@ietf.org
> |Subject: Re: [BEHAVE] FW: I-D Action:
> draft-muthu-behave-consent-freshness-04.txt
> |
> |Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :****
>
> |> The text and the algorithm in the draft are significantly simplified i=
n
> this updated version.
> |>
> |> Comments welcome..
> |****
>
> |MUCH better introduction. Now I feel like I understand the need exactly.
> |
> |The "Design Considerations" section is still very confusing to me.
> |
> |>    Though ICE specifies STUN Binding indications to be used for
> |>    keepalives, it requires that an agent be prepared to receive
> |>    connectivity check as well.  If a connectivity check is received, a
> |>    response is generated, but there is no impact on ICE processing, as
> |>    described in section 10 of [RFC5245].
> |
> |...so? Why is "an impact on ICE processing" necessary?
>
> Meant to stress these Binding request/response doesn't trigger an ICE
> restart..
>
> |
> |>    While a WebRTC browser could verify whether the peer continues to
> |>    send SRTCP reports before sending traffic to the peer, the usage of
> |>    SRTCP together with Security Descriptions [RFC4568] requires exposi=
ng
> |>    the media keys to the JavaScript and renders SRTCP unsuitable for
> |>    consent freshness.
> |
> |Why does it "require exposing the media keys to the JavaScript"? Is this
> |because of a law of nature, or is it because of the way the JavaScript
> |API is being designed? Could the JS API be changed to accommodate
> |SRTCP+SDES?
>
> That's how the API construct looks like today -- the JavaScript would get
> an SDP blob from the browser containing the crypto keys used for SDES-SRT=
P.
> Of course, the browser could hide those keys by putting a "****" in SDP -=
:).
> ****
>
> ** **
>
> ** **
>
> Uh, then how would they be sent to the other side?****
>
> ** **
>
> As far as I can tell, with any plausible API SDES requires exposing the
> keys to JS.****
>
> ** **
>
> -Ekr****
>
> ** **
>

--047d7bb03e6a41619d04e1bc38e6
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 17, 2013 at 8:47 PM, Muthu Arul Mozhi Perumal (mperumal=
) <span dir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_bl=
ank">mperumal@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div><div class=3D"im">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|Uh, then how would they be sent to the other side?<u></u>=
<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
</div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&q=
uot;Courier New&quot;">Ah, the browser would have to modify the SDP when it=
 is sent on the wire..</span></p></div></div></blockquote><div><br></div><d=
iv>

I don&#39;t really see how this helps since the server is going to see it a=
nd call tell</div><div>the JS (remember, that&#39;s where the JS came from)=
.</div><div><br></div><div>-Ekr</div><div><br></div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-family:&#39;Courier New&#39;;font-size:10pt;color:r=
gb(80,0,80)">=A0</span></p><div class=3D"im">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|As far as I can tell, with any plausible API SDES require=
s exposing the keys to JS.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
</div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&q=
uot;Courier New&quot;">That helps..thanks.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Eric Res=
corla [mailto:<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.co=
m</a>]
<br>
<b>Sent:</b> Wednesday, July 17, 2013 2:32 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> Simon Perreault; <a href=3D"mailto:behave@ietf.org" target=3D"_b=
lank">behave@ietf.org</a>; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_bl=
ank">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-co=
nsent-freshness-04.txt<u></u><u></u></span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 16, 2013 at 6:49 PM, Muthu Arul Mozhi Pe=
rumal (mperumal) &lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_blank=
">mperumal@cisco.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">[Added rtcweb since I am not sure if everyone involv=
ed there are following this discussion in behave]<br>
<br>
Thanks for the review. See inline..<br>
<br>
|-----Original Message-----<br>
|From: <a href=3D"mailto:behave-bounces@ietf.org" target=3D"_blank">behave-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:behave-bounces@ietf.org" tar=
get=3D"_blank">behave-bounces@ietf.org</a>] On Behalf Of Simon Perreault<br=
>
|Sent: Tuesday, July 16, 2013 3:05 PM<br>
|To: <a href=3D"mailto:behave@ietf.org" target=3D"_blank">behave@ietf.org</=
a><br>
|Subject: Re: [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness=
-04.txt<br>
|<br>
|Le 2013-07-15 20:42, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :<u></u=
><u></u></p>
<div>
<p class=3D"MsoNormal">|&gt; The text and the algorithm in the draft are si=
gnificantly simplified in this updated version.<br>
|&gt;<br>
|&gt; Comments welcome..<br>
|<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">|MUCH better introduction. Now I feel like I underst=
and the need exactly.<br>
|<br>
|The &quot;Design Considerations&quot; section is still very confusing to m=
e.<br>
|<br>
|&gt; =A0 =A0Though ICE specifies STUN Binding indications to be used for<b=
r>
|&gt; =A0 =A0keepalives, it requires that an agent be prepared to receive<b=
r>
|&gt; =A0 =A0connectivity check as well. =A0If a connectivity check is rece=
ived, a<br>
|&gt; =A0 =A0response is generated, but there is no impact on ICE processin=
g, as<br>
|&gt; =A0 =A0described in section 10 of [RFC5245].<br>
|<br>
|...so? Why is &quot;an impact on ICE processing&quot; necessary?<br>
<br>
Meant to stress these Binding request/response doesn&#39;t trigger an ICE r=
estart..<br>
<br>
|<br>
|&gt; =A0 =A0While a WebRTC browser could verify whether the peer continues=
 to<br>
|&gt; =A0 =A0send SRTCP reports before sending traffic to the peer, the usa=
ge of<br>
|&gt; =A0 =A0SRTCP together with Security Descriptions [RFC4568] requires e=
xposing<br>
|&gt; =A0 =A0the media keys to the JavaScript and renders SRTCP unsuitable =
for<br>
|&gt; =A0 =A0consent freshness.<br>
|<br>
|Why does it &quot;require exposing the media keys to the JavaScript&quot;?=
 Is this<br>
|because of a law of nature, or is it because of the way the JavaScript<br>
|API is being designed? Could the JS API be changed to accommodate<br>
|SRTCP+SDES?<br>
<br>
That&#39;s how the API construct looks like today -- the JavaScript would g=
et an SDP blob from the browser containing the crypto keys used for SDES-SR=
TP. Of course, the browser could hide those keys by putting a &quot;****&qu=
ot; in SDP -:).<u></u><u></u></p>


<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Uh, then how would they be sent to the other side?<u=
></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As far as I can tell, with any plausible API SDES re=
quires exposing the keys to JS.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">-Ekr<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
</div>
</div>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br></div></div>

--047d7bb03e6a41619d04e1bc38e6--

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From: Justin Uberti <juberti@google.com>
Date: Wed, 17 Jul 2013 18:11:42 -0400
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--14dae9cc955c1083aa04e1bc627c
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On Wed, Jul 17, 2013 at 5:09 PM, Rajmohan Banavi
<rajmohanbanavi@gmail.com>wrote:

> Thanks for the detailed example. I agree with the entire flow you have
> given. My only concern is - why should the ephemeral password be generated
> by TURN server using HMAC SHA on then username string? Why can't this be
> some random unique string?
>
> Taking your example -
>
> Suppose TURN server generates credentials, username =
> "12334939:mbzrxpgjys" and password ="+somerandomstring+". This is a
> randomly generated string and passed onto webrtc app. The TURN server also
> stores these credentials for later validation.
> var iceServer = {
>      "username": 12334939:mbzrxpgjys,
>      "credential": +somerandomstring+,
>    };
>
> These credentials are passed from JS to the webrtc client stack in the
> browser.
>
> On Browser WebTRC Client.
> key = MD5(12334939:mbzrxpgjys ":" realm ":" +somerandomstring+)
> Similarly on the TURN server, the key for MESSAGE-INTEGRITY is calculated
> using
> key = MD5(12334939:mbzrxpgjys ":" realm ":" +somerandomstring+)
>
> So the browser and turn server use the same input to the M-I. The one
> implementation related advantage I see with this way of generation of TURN
> password using HMAC is that the TURN server need not really store the HMAC
> generated password, but can calculate the same using the USERNAME attribute
> in the received TURN request.
>

Correct, but that is a significant advantage, especially when the web and
TURN servers are separate entities.

>
> In fact, I do not see why the sharedsecret needs to be shared between the
> TURN server and the WebRTC application. The sharedsecret key "secret" is
> only required by the TURN server for validating the received TURN ALLOCATE
> requests once it has generated and sent out the ephemeral credentials. What
> does the WebRTC application do with the shared secret key?
>

The WebRTC client application doesn't see the secret key, this is only
shared between the web and TURN servers.


>
> On Thu, Jul 18, 2013 at 2:09 AM, Philipp Hancke <fippo@goodadvice.pages.de
> > wrote:
>
>> Am 17.07.2013 20:56, schrieb Rajmohan Banavi:
>>
>>  Hi Justin,
>>>
>>> I reviewed draft-uberti-behave-turn-rest-**00 and have the following
>>> comments.
>>>
>> [...]
>>
>>>     2. Sec 4.3 last paragraph - "Because the password is derived from the
>>>
>>>     USERNAME, successful verification of the MESSAGE-INTEGRITY ensures
>>> that the
>>>     username is trustworthy". I believe this validation is taken care of
>>> in the
>>>     TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, the TURN
>>>     client generates a long term key which is computed as => key =
>>>     MD5(username ":" realm ":" SASLprep(password)). This key is used to
>>> perform
>>>     hmac on the message. The same is done on the TURN server end as
>>> well. If
>>>     the MESSAGE-INTEGRITY is verified, then it implies that the username
>>> is
>>>     trustworthy.
>>>
>>
>> Let me try to address that with a lenghty example:
>> The web server returns
>> password = BASE64(HMAC-SHA1(username, sharedsecret))
>> to the browser.
>>
>> For example, with username = "12334939:mbzrxpgjys" and sharedsecret
>> "secret", the password given to the browser would be
>> "+4pCXR06gx/cqAikLYnBajtZW6E=" (hopefully)
>>
>>
>>
>>
>> The browser passes the username, uri and password to it's webrtc stack.
>> The password is exposed to the user of the browser (which might be anyone
>> surfing your website) during that process.
>>
>> That stack does long term authentication and uses this password string as
>> input for calculating MESSAGE-INTEGRITY with
>> key = MD5(username ":" realm ":" SASLprep(password))
>>
>> For the example this would be
>> key =MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep("+4pCXR06gx/**
>> cqAikLYnBajtZW6E=")
>>
>>
>>
>>
>> On the TURN server, the key for MESSAGE-INTEGRITY is calculated using
>> key = MD5(username ":" realm ":" SASLprep(BASE64(HMAC-SHA1(**USERNAME,
>> sharedsecret))
>>
>> So for the example this would be
>>  = MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep(BASE64(HMAC-SHA1("**12334939:mbzrxpgjys",
>> "secret"))
>>  = MD5("12334939:mbzrxpgjys" ":" realm ":" SASLprep("+4pCXR06gx/**
>> cqAikLYnBajtZW6E=")
>>
>> So the browser and turn server use the same input to the M-I. This can be
>> done by looking at the request alone given the shared secret.
>>
>>
>> Does that make it clearer?
>>
>> philipp,
>> who only understood it after implementing it
>>
>
>
>
> --
>
> Life is here and now, not yesterday, not tomorrow...!
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--14dae9cc955c1083aa04e1bc627c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 17, 2013 at 5:09 PM, Rajmohan Banavi <span dir=3D"ltr">=
&lt;<a href=3D"mailto:rajmohanbanavi@gmail.com" target=3D"_blank">rajmohanb=
anavi@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">Thanks for the detailed exa=
mple. I agree with the entire flow you have given. My only concern is - why=
 should the ephemeral password be generated by TURN server using HMAC SHA o=
n then username string? Why can&#39;t this be some random unique string?<di=
v>


<br></div><div>Taking your example -=C2=A0</div><div><br></div><div><span s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">Suppose TURN server ge=
nerates credentials, username =3D &quot;12334939:mbzrxpgjys&quot; and passw=
ord =3D</span><span style=3D"font-family:arial,sans-serif;font-size:13px">&=
quot;+somerandomstring+&quot;. This is a randomly generated string and pass=
ed onto webrtc app. The TURN server also stores these credentials for later=
 validation.</span><br style=3D"font-family:arial,sans-serif;font-size:13px=
">


<div><font face=3D"arial, sans-serif">var iceServer =3D {</font></div><div>=
<font face=3D"arial, sans-serif">=C2=A0 =C2=A0 =C2=A0&quot;username&quot;:=
=C2=A0</font><span style=3D"font-family:arial,sans-serif;font-size:13px">12=
334939:mbzrxpgjys</span><font face=3D"arial, sans-serif">,</font></div>


<div><font face=3D"arial, sans-serif">=C2=A0 =C2=A0 =C2=A0&quot;credential&=
quot;:=C2=A0</font><span style=3D"font-family:arial,sans-serif;font-size:13=
px">+somerandomstring+</span><font face=3D"arial, sans-serif">,</font></div=
><div><span style=3D"font-family:arial,sans-serif">=C2=A0 =C2=A0};</span><b=
r>


</div><div><font face=3D"arial, sans-serif"><br></font></div><font face=3D"=
arial, sans-serif">These credentials are passed from JS to the webrtc clien=
t stack in the browser.</font><br style=3D"font-family:arial,sans-serif;fon=
t-size:13px">


<span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span></di=
v><div><span style=3D"font-family:arial,sans-serif;font-size:13px">On Brows=
er WebTRC Client.=C2=A0</span></div><div><span style=3D"font-size:13px;font=
-family:arial,sans-serif">key =3D MD5(</span><span style=3D"font-family:ari=
al,sans-serif;font-size:13px">12334939:mbzrxpgjys</span><span style=3D"font=
-size:13px;font-family:arial,sans-serif">=C2=A0&quot;:&quot; realm &quot;:&=
quot;=C2=A0</span><span style=3D"font-family:arial,sans-serif;font-size:13p=
x">+somerandomstring+</span><span style=3D"font-family:arial,sans-serif;fon=
t-size:13px">)</span></div>


<div><span style=3D"font-family:arial,sans-serif;font-size:13px">Similarly =
on the TURN server, the key for MESSAGE-INTEGRITY is calculated using</span=
><br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"f=
ont-family:arial,sans-serif;font-size:13px">key =3D MD5(</span><span style=
=3D"font-family:arial,sans-serif;font-size:13px">12334939:mbzrxpgjys</span>=
<span style=3D"font-family:arial,sans-serif;font-size:13px">=C2=A0&quot;:&q=
uot; realm &quot;:&quot;=C2=A0</span><span style=3D"font-family:arial,sans-=
serif;font-size:13px">+somerandomstring+</span><span style=3D"font-family:a=
rial,sans-serif;font-size:13px">)</span><br style=3D"font-family:arial,sans=
-serif;font-size:13px">


<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">So the browser and turn server u=
se the same input to the M-I. The one implementation related advantage I se=
e with this way of generation of TURN password using HMAC is that the TURN =
server need not really store the HMAC generated password, but can calculate=
 the same using the USERNAME attribute in the received TURN request.</span>=
</div>

</div></blockquote><div><br></div><div>Correct, but that is a significant a=
dvantage, especially when the web and TURN servers are separate entities.=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr">
<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">In =
fact, I do not see why the sharedsecret needs to be shared between the TURN=
 server and the WebRTC application. The sharedsecret key &quot;secret&quot;=
 is only required by the TURN server for validating the received TURN ALLOC=
ATE requests once it has generated and sent out the ephemeral credentials. =
What does the WebRTC application do with the shared secret key?</span></div=
>

</div></blockquote><div><br></div><div>The WebRTC client application doesn&=
#39;t see the secret key, this is only shared between the web and TURN serv=
ers.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr"><div></div></div><div class=3D"gmail_extra"><div><div clas=
s=3D"h5"><br><div class=3D"gmail_quote">On Thu, Jul 18, 2013 at 2:09 AM, Ph=
ilipp Hancke <span dir=3D"ltr">&lt;<a href=3D"mailto:fippo@goodadvice.pages=
.de" target=3D"_blank">fippo@goodadvice.pages.de</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Am 17.07.2013 20:56, schrieb Rajmohan Banavi=
:<div><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Justin,<br>
<br>
I reviewed draft-uberti-behave-turn-rest-<u></u>00 and have the following c=
omments.<br>
</blockquote></div>
[...]<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 =C2=A0 2. Sec 4.3 last paragraph - &quot;Because the password is der=
ived from the<div><br>
=C2=A0 =C2=A0 USERNAME, successful verification of the MESSAGE-INTEGRITY en=
sures that the<br>
=C2=A0 =C2=A0 username is trustworthy&quot;. I believe this validation is t=
aken care of in the<br>
=C2=A0 =C2=A0 TURN RFC itself. In order to generate the MESSAGE-INTEGRITY, =
the TURN<br>
=C2=A0 =C2=A0 client generates a long term key which is computed as =3D&gt;=
 key =3D<br>
=C2=A0 =C2=A0 MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(passw=
ord)). This key is used to perform<br>
=C2=A0 =C2=A0 hmac on the message. The same is done on the TURN server end =
as well. If<br>
=C2=A0 =C2=A0 the MESSAGE-INTEGRITY is verified, then it implies that the u=
sername is<br>
=C2=A0 =C2=A0 trustworthy.<br>
</div></blockquote>
<br>
Let me try to address that with a lenghty example:<br>
The web server returns<br>
password =3D BASE64(HMAC-SHA1(username, sharedsecret))<br>
to the browser.<br>
<br>
For example, with username =3D &quot;12334939:mbzrxpgjys&quot; and sharedse=
cret &quot;secret&quot;, the password given to the browser would be<br>
&quot;+4pCXR06gx/cqAikLYnBajtZW6E=3D&quot; (hopefully)<br>
<br>
<br>
<br>
<br>
The browser passes the username, uri and password to it&#39;s webrtc stack.=
<br>
The password is exposed to the user of the browser (which might be anyone s=
urfing your website) during that process.<br>
<br>
That stack does long term authentication and uses this password string as i=
nput for calculating MESSAGE-INTEGRITY with<br>
key =3D MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(password))<=
br>
<br>
For the example this would be<br>
key =3DMD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&quot=
; SASLprep(&quot;+4pCXR06gx/<u></u>cqAikLYnBajtZW6E=3D&quot;)<br>
<br>
<br>
<br>
<br>
On the TURN server, the key for MESSAGE-INTEGRITY is calculated using<br>
key =3D MD5(username &quot;:&quot; realm &quot;:&quot; SASLprep(BASE64(HMAC=
-SHA1(<u></u>USERNAME, sharedsecret))<br>
<br>
So for the example this would be<br>
=C2=A0=3D MD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&q=
uot; SASLprep(BASE64(HMAC-SHA1(&quot;<u></u>12334939:mbzrxpgjys&quot;, &quo=
t;secret&quot;))<br>
=C2=A0=3D MD5(&quot;12334939:mbzrxpgjys&quot; &quot;:&quot; realm &quot;:&q=
uot; SASLprep(&quot;+4pCXR06gx/<u></u>cqAikLYnBajtZW6E=3D&quot;)<br>
<br>
So the browser and turn server use the same input to the M-I. This can be d=
one by looking at the request alone given the shared secret.<br>
<br>
<br>
Does that make it clearer?<br>
<br>
philipp,<br>
who only understood it after implementing it<br>
</blockquote></div><br><br clear=3D"all"><div><br></div></div></div><div cl=
ass=3D"im">-- <br><br>Life is here and now, not yesterday, not tomorrow...!
</div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--14dae9cc955c1083aa04e1bc627c--

From ted.ietf@gmail.com  Wed Jul 17 15:40:38 2013
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Date: Wed, 17 Jul 2013 15:40:36 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
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Subject: Re: [rtcweb] Locus of API discussion
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On Wed, Jul 17, 2013 at 2:29 PM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

>
> I am frankly quite concerned given earlier comments that yes, the W3C had
> given responsibility for "JSEP" and the resulting SDP work to IETF, and so
> now we have the IETF WG making decisions but the discussion happening on
> the W3C list -- which means there's no "paper trail" prior to the IETF
> meeting on an IETF list about how the participants have weighed in -- and
> then conversely when the W3C spec gets to last call there will be no "paper
> trail" at the W3C showing how decisions had been reached about the SDP
> "API", as that happened over in the MMUSIC WG, which list was not even
> copied on this message.
>
>
All three lists you cited have public archives, so it is clearly not the
case that there is no "paper trail".   But it does occasionally get
confusing, and the chairs made the "locus of API discussions" point in
order to take a particular set of discussions proposing a significant
change to the API and reduce the number of places someone who have to
follow to contribute to that discussion.

I grieve that this has disappointed you,

Ted

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On Wed, Jul 17, 2013 at 2:29 PM, Matthew Kaufman (SKYPE) <span dir=3D"ltr">=
&lt;<a href=3D"mailto:matthew.kaufman@skype.net" target=3D"_blank">matthew.=
kaufman@skype.net</a>&gt;</span> wrote:<br>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class=
=3D"h5">=A0<br></div></div>
I am frankly quite concerned given earlier comments that yes, the W3C had g=
iven responsibility for &quot;JSEP&quot; and the resulting SDP work to IETF=
, and so now we have the IETF WG making decisions but the discussion happen=
ing on the W3C list -- which means there&#39;s no &quot;paper trail&quot; p=
rior to the IETF meeting on an IETF list about how the participants have we=
ighed in -- and then conversely when the W3C spec gets to last call there w=
ill be no &quot;paper trail&quot; at the W3C showing how decisions had been=
 reached about the SDP &quot;API&quot;, as that happened over in the MMUSIC=
 WG, which list was not even copied on this message.<br>

<span class=3D"HOEnZb"><font color=3D"#888888"><br></font></span></blockquo=
te><div>=A0<br>All three lists you cited have public archives, so it is cle=
arly not the case that there is no &quot;paper trail&quot;.=A0=A0 But it do=
es occasionally get confusing, and the chairs made the &quot;locus of API d=
iscussions&quot; point in order to take a particular set of discussions pro=
posing a significant change to the API and reduce the number of places some=
one who have to follow to contribute to that discussion.=A0 <br>
<br>I grieve that this has disappointed you,<br><br>Ted<br></div></div>

--089e015370963e3d4a04e1bcc86a--

From ted.ietf@gmail.com  Wed Jul 17 16:05:14 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] (possibly very) draft agenda uploaded
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Today is the required date for uploading an agenda for the working group
and I have done so here:

http://www.ietf.org/proceedings/87/agenda/agenda-87-rtcweb

There is still ongoing discussion and further agenda bashing is possible,
so please continue to provide feedback.
We are confident that we will discuss SDES, the results of the MMUSIC plan
discussions from earlier
in the week, and the data channel documents.  Placement in the agenda and
time of those three may
still shift, though, so please plan on attending both days if there are
topics where you want to be in
the face to face discussion.

thanks,

Ted Hardie

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Today is the required date for uploading an agenda for the working group an=
d I have done so here:<br><br><a href=3D"http://www.ietf.org/proceedings/87=
/agenda/agenda-87-rtcweb">http://www.ietf.org/proceedings/87/agenda/agenda-=
87-rtcweb</a><br>
<br>There is still ongoing discussion and further agenda bashing is possibl=
e, so please continue to provide feedback.<br>We are confident that we will=
 discuss SDES, the results of the MMUSIC plan discussions from earlier<br>
in the week, and the data channel documents.=A0 Placement in the agenda and=
 time of those three may<br>still shift, though, so please plan on attendin=
g both days if there are topics where you want to be in<br>the face to face=
 discussion.<br>
<br>thanks,<br><br>Ted Hardie<br><br><br>

--047d7bdca46846cbe004e1bd206e--

From fineberg@vline.me  Wed Jul 17 17:09:54 2013
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Date: Wed, 17 Jul 2013 17:09:46 -0700
From: Adam Fineberg <fineberg@vline.me>
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Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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This is a multi-part message in MIME format.
--------------030403020500010002050000
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Hi,

I'm working on WebRTC services and have found that while developing 
services that forward VP8 video streams if we want to take advantage of 
the VP8 temporal scaling we must get the temporal layer information from 
the RTP header which requires us to decrypt the SRTP packets. This is 
undesirable both because the middle-box needs to have access to the keys 
as well as the because of the added overhead of the decrypt/encrypt 
cycle. This draft proposes an RTP header extension that will allow us to 
use the VP8 temporal layer information included in the header extension 
and therefore do forwarding without SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com

-------- Original Message --------
Subject: 	New Version Notification for 
draft-fineberg-avtext-temporal-layer-ext-00.txt
Date: 	Tue, 09 Jul 2013 10:02:05 -0700
From: 	internet-drafts at ietf.org
To: 	Adam Fineberg<fineberg at vline.com>



A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00


Abstract:
    This document defines a mechanism by which packets of Real-Time
    Tranport Protocol (RTP) video streams encoded with the VP8 codec can
    indicate, in an RTP header extension, the temporal layer information
    about the frame encoded in the RTP packet.  This information can be
    used in a middlebox performing bandwidth management of streams
    without requiring it to decrypt the streams.


--------------030403020500010002050000
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
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      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none;">Hi,</span><br
      style="color: rgb(0, 0, 0); font-family: Times; font-size: medium;
      font-style: normal; font-variant: normal; font-weight: normal;
      letter-spacing: normal; line-height: normal; orphans: auto;
      text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255);">
    <br style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
      auto; text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255);">
    <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
      auto; text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none;">I'm working on
      WebRTC services and have found that while developing services that
      forward VP8 video streams if we want to take advantage of the VP8
      temporal scaling we must get the temporal layer information from
      the RTP header which requires us to decrypt the SRTP packets. This
      is undesirable both because the middle-box needs to have access to
      the keys as well as the because of the added overhead of the
      decrypt/encrypt cycle. This draft proposes an RTP header extension
      that will allow us to use the VP8 temporal layer information
      included in the header extension and therefore do forwarding
      without SRTP decryption. Comments welcome.</span><br style="color:
      rgb(0, 0, 0); font-family: Times; font-size: medium; font-style:
      normal; font-variant: normal; font-weight: normal; letter-spacing:
      normal; line-height: normal; orphans: auto; text-align: start;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
      background-color: rgb(255, 255, 255);">
    <br style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
      auto; text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255);">
    <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
      auto; text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none;">Regards,</span><br
      style="color: rgb(0, 0, 0); font-family: Times; font-size: medium;
      font-style: normal; font-variant: normal; font-weight: normal;
      letter-spacing: normal; line-height: normal; orphans: auto;
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      255);">
    <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
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      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none;">Adam Fineberg</span><br
      style="color: rgb(0, 0, 0); font-family: Times; font-size: medium;
      font-style: normal; font-variant: normal; font-weight: normal;
      letter-spacing: normal; line-height: normal; orphans: auto;
      text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255);">
    <div class="moz-forward-container" style="color: rgb(0, 0, 0);
      font-family: Times; font-size: medium; font-style: normal;
      font-variant: normal; font-weight: normal; letter-spacing: normal;
      line-height: normal; orphans: auto; text-align: start;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
      background-color: rgb(255, 255, 255);"><a rel="nofollow"
        class="moz-txt-link-abbreviated"
        href="mailto:fineberg%20at%20vline.com">fineberg at vline.com</a><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
            <td>New Version Notification for
              draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
            <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
            <td><a rel="nofollow" class="moz-txt-link-abbreviated"
                href="mailto:internet-drafts%20at%20ietf.org">internet-drafts

                at ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
            <td>Adam Fineberg<span class="Apple-converted-space">&nbsp;</span><a
                rel="nofollow" class="moz-txt-link-rfc2396E"
                href="mailto:fineberg%20at%20vline.com">&lt;fineberg at
                vline.com&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre style="white-space: pre-wrap; word-wrap: break-word; width: 939.5999755859375px;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a rel="nofollow" class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a rel="nofollow" class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a rel="nofollow" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
    </div>
  </body>
</html>

--------------030403020500010002050000--

From bernard_aboba@hotmail.com  Wed Jul 17 20:27:05 2013
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--_4899aed9-d0eb-4689-81a9-9545eb48347c_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal=2C spatial and quality scalability)=2C why=20
 a VP8-specific RTP extension?=20
=20
Date: Wed=2C 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me
To: rtcweb@ietf.org
Subject: [rtcweb] Fwd: New Version Notification for	draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

=0A=
  =0A=
=0A=
    =0A=
  =0A=
  =0A=
    Hi=2C=0A=
    =0A=
    I'm working on=0A=
      WebRTC services and have found that while developing services that=0A=
      forward VP8 video streams if we want to take advantage of the VP8=0A=
      temporal scaling we must get the temporal layer information from=0A=
      the RTP header which requires us to decrypt the SRTP packets. This=0A=
      is undesirable both because the middle-box needs to have access to=0A=
      the keys as well as the because of the added overhead of the=0A=
      decrypt/encrypt cycle. This draft proposes an RTP header extension=0A=
      that will allow us to use the VP8 temporal layer information=0A=
      included in the header extension and therefore do forwarding=0A=
      without SRTP decryption. Comments welcome.=0A=
    =0A=
    Regards=2C=0A=
    Adam Fineberg=0A=
    fineberg at vline.com
=0A=
     =20
=0A=
      -------- Original Message --------=0A=
      =0A=
        =0A=
          =0A=
            Subject:=0A=
            New Version Notification for=0A=
              draft-fineberg-avtext-temporal-layer-ext-00.txt=0A=
          =0A=
          =0A=
            Date:=0A=
            Tue=2C 09 Jul 2013 10:02:05 -0700=0A=
          =0A=
          =0A=
            From:=0A=
            internet-drafts=0A=
=0A=
                at ietf.org=0A=
          =0A=
          =0A=
            To:=0A=
            Adam Fineberg <fineberg at=0A=
                vline.com>=0A=
          =0A=
        =0A=
      =0A=
     =20
=0A=
     =20
=0A=
      A new version of I-D=2C draft-fineberg-avtext-temporal-layer-ext-00.t=
xt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt=0A=
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext=0A=
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
=0A=
    =0A=
  =0A=
=0A=

_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb 		 	   		  =

--_4899aed9-d0eb-4689-81a9-9545eb48347c_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Since the need is not codec spec=
ific (e.g. it arises with any codec supporting temporal=2C spatial and qual=
ity scalability)=2C why <br>&nbsp=3Ba VP8-specific RTP extension? <BR>&nbsp=
=3B<BR><div><hr id=3D"stopSpelling">Date: Wed=2C 17 Jul 2013 17:09:46 -0700=
<br>From: fineberg@vline.me<br>To: rtcweb@ietf.org<br>Subject: [rtcweb] Fwd=
: New Version Notification for	draft-fineberg-avtext-temporal-layer-ext-00.=
txt<br><br>=0A=
  =0A=
=0A=
    =0A=
  =0A=
  =0A=
    <span style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-=
indent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B display: inli=
ne !important=3B white-space: normal=3B background-color: rgb(255=2C 255=2C=
 255)=3B">Hi=2C</span><br style=3D"color: rgb(0=2C 0=2C 0)=3B text-transfor=
m: none=3B text-indent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=
=3B white-space: normal=3B background-color: rgb(255=2C 255=2C 255)=3B">=0A=
    <br style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-in=
dent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B white-space: no=
rmal=3B background-color: rgb(255=2C 255=2C 255)=3B">=0A=
    <span style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-=
indent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B display: inli=
ne !important=3B white-space: normal=3B background-color: rgb(255=2C 255=2C=
 255)=3B">I'm working on=0A=
      WebRTC services and have found that while developing services that=0A=
      forward VP8 video streams if we want to take advantage of the VP8=0A=
      temporal scaling we must get the temporal layer information from=0A=
      the RTP header which requires us to decrypt the SRTP packets. This=0A=
      is undesirable both because the middle-box needs to have access to=0A=
      the keys as well as the because of the added overhead of the=0A=
      decrypt/encrypt cycle. This draft proposes an RTP header extension=0A=
      that will allow us to use the VP8 temporal layer information=0A=
      included in the header extension and therefore do forwarding=0A=
      without SRTP decryption. Comments welcome.</span><br style=3D"color: =
rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-indent: 0px=3B letter-spac=
ing: normal=3B word-spacing: 0px=3B white-space: normal=3B background-color=
: rgb(255=2C 255=2C 255)=3B">=0A=
    <br style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-in=
dent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B white-space: no=
rmal=3B background-color: rgb(255=2C 255=2C 255)=3B">=0A=
    <span style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-=
indent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B display: inli=
ne !important=3B white-space: normal=3B background-color: rgb(255=2C 255=2C=
 255)=3B">Regards=2C</span><br style=3D"color: rgb(0=2C 0=2C 0)=3B text-tra=
nsform: none=3B text-indent: 0px=3B letter-spacing: normal=3B word-spacing:=
 0px=3B white-space: normal=3B background-color: rgb(255=2C 255=2C 255)=3B"=
>=0A=
    <span style=3D"color: rgb(0=2C 0=2C 0)=3B text-transform: none=3B text-=
indent: 0px=3B letter-spacing: normal=3B word-spacing: 0px=3B display: inli=
ne !important=3B white-space: normal=3B background-color: rgb(255=2C 255=2C=
 255)=3B">Adam Fineberg</span><br style=3D"color: rgb(0=2C 0=2C 0)=3B text-=
transform: none=3B text-indent: 0px=3B letter-spacing: normal=3B word-spaci=
ng: 0px=3B white-space: normal=3B background-color: rgb(255=2C 255=2C 255)=
=3B">=0A=
    <div class=3D"ecxmoz-forward-container" style=3D"color: rgb(0=2C 0=2C 0=
)=3B text-transform: none=3B text-indent: 0px=3B letter-spacing: normal=3B =
word-spacing: 0px=3B white-space: normal=3B background-color: rgb(255=2C 25=
5=2C 255)=3B"><a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:fineb=
erg%20at%20vline.com" rel=3D"nofollow">fineberg at vline.com</a><br>=0A=
      <br>=0A=
      -------- Original Message --------=0A=
      <table class=3D"ecxmoz-email-headers-table" border=3D"0" cellspacing=
=3D"0" cellpadding=3D"0">=0A=
        <tbody>=0A=
          <tr>=0A=
            <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Subje=
ct:</th>=0A=
            <td>New Version Notification for=0A=
              draft-fineberg-avtext-temporal-layer-ext-00.txt</td>=0A=
          </tr>=0A=
          <tr>=0A=
            <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Date:=
</th>=0A=
            <td>Tue=2C 09 Jul 2013 10:02:05 -0700</td>=0A=
          </tr>=0A=
          <tr>=0A=
            <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">From:=
</th>=0A=
            <td><a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:int=
ernet-drafts%20at%20ietf.org" rel=3D"nofollow">internet-drafts=0A=
=0A=
                at ietf.org</a></td>=0A=
          </tr>=0A=
          <tr>=0A=
            <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">To:</=
th>=0A=
            <td>Adam Fineberg<span class=3D"ecxApple-converted-space">&nbsp=
=3B</span><a class=3D"ecxmoz-txt-link-rfc2396E" href=3D"mailto:fineberg%20a=
t%20vline.com" rel=3D"nofollow">&lt=3Bfineberg at=0A=
                vline.com&gt=3B</a></td>=0A=
          </tr>=0A=
        </tbody>=0A=
      </table>=0A=
      <br>=0A=
      <br>=0A=
      <pre style=3D"width: 939.59px=3B white-space: pre-wrap=3B -ms-word-wr=
ap: break-word=3B">A new version of I-D=2C draft-fineberg-avtext-temporal-l=
ayer-ext-00.txt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://www.i=
etf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" ta=
rget=3D"_blank" rel=3D"nofollow">http://www.ietf.org/internet-drafts/draft-=
fineberg-avtext-temporal-layer-ext-00.txt</a>=0A=
Status:          <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://datat=
racker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target=3D"_bl=
ank" rel=3D"nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext=
-temporal-layer-ext</a>=0A=
Htmlized:        <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://tools=
.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target=3D"_blan=
k" rel=3D"nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-tempor=
al-layer-ext-00</a>=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
</pre>=0A=
    </div>=0A=
  =0A=
=0A=
<br>_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
</html>=

--_4899aed9-d0eb-4689-81a9-9545eb48347c_--

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Subject: [rtcweb] e= lines (Re: Summary of Application Developers' opinions ..)
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Matthew, thanks for sending 12 messages in 3 hours to the mailing list.

Just one point on which I corrected you before, but since I see you're
repeating it:

On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:
> - What if someone adds an r= or p= or e=?

The code for handling an e= line has been in Chrome since a very early
release.
You claimed in Lyon that you'd tested it and failed to have it parse,
but I never saw the test case; you may have inserted it into the wrong
place, rendering the SDP invalid.

The SDP specification is quite strict about the order of these fields,
exactly to make it simpler to write parsers.


From harald@alvestrand.no  Thu Jul 18 01:17:06 2013
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Date: Thu, 18 Jul 2013 10:16:50 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: rtcweb@ietf.org, Adam Fineberg <fineberg@vline.me>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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On 07/18/2013 02:09 AM, Adam Fineberg wrote:
> Hi,
>
> I'm working on WebRTC services and have found that while developing 
> services that forward VP8 video streams if we want to take advantage 
> of the VP8 temporal scaling we must get the temporal layer information 
> from the RTP header which requires us to decrypt the SRTP packets. 
> This is undesirable both because the middle-box needs to have access 
> to the keys as well as the because of the added overhead of the 
> decrypt/encrypt cycle. This draft proposes an RTP header extension 
> that will allow us to use the VP8 temporal layer information included 
> in the header extension and therefore do forwarding without SRTP 
> decryption. Comments welcome.

Just checking - you named your draft with the -avtext- convention; this 
means that you want comments to go to avtext, yes?


>
> Regards,
> Adam Fineberg
> fineberg at vline.com
>
> -------- Original Message --------
> Subject: 	New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
> Date: 	Tue, 09 Jul 2013 10:02:05 -0700
> From: 	internet-drafts at ietf.org
> To: 	Adam Fineberg<fineberg at vline.com>
>
>
>
> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
> has been successfully submitted by Adam Fineberg and posted to the
> IETF repository.
>
> Filename:	 draft-fineberg-avtext-temporal-layer-ext
> Revision:	 00
> Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
> Creation date:	 2013-07-08
> Group:		 Individual Submission
> Number of pages: 6
> URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
> Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
> Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>
> Abstract:
>     This document defines a mechanism by which packets of Real-Time
>     Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>     indicate, in an RTP header extension, the temporal layer information
>     about the frame encoded in the RTP packet.  This information can be
>     used in a middlebox performing bandwidth management of streams
>     without requiring it to decrypt the streams.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
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      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 07/18/2013 02:09 AM, Adam Fineberg
      wrote:<br>
    </div>
    <blockquote cite="mid:51E7324A.3090405@vline.me" type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255); display: inline !important; float: none;">Hi,</span><br
        style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255);">
      <br style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255);">
      <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255); display: inline !important; float: none;">I'm working on
        WebRTC services and have found that while developing services
        that forward VP8 video streams if we want to take advantage of
        the VP8 temporal scaling we must get the temporal layer
        information from the RTP header which requires us to decrypt the
        SRTP packets. This is undesirable both because the middle-box
        needs to have access to the keys as well as the because of the
        added overhead of the decrypt/encrypt cycle. This draft proposes
        an RTP header extension that will allow us to use the VP8
        temporal layer information included in the header extension and
        therefore do forwarding without SRTP decryption. Comments
        welcome.</span><br style="color: rgb(0, 0, 0); font-family:
        Times; font-size: medium; font-style: normal; font-variant:
        normal; font-weight: normal; letter-spacing: normal;
        line-height: normal; orphans: auto; text-align: start;
        text-indent: 0px; text-transform: none; white-space: normal;
        widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
        background-color: rgb(255, 255, 255);">
    </blockquote>
    <br>
    Just checking - you named your draft with the -avtext- convention;
    this means that you want comments to go to avtext, yes?<br>
    <br>
    <br>
    <blockquote cite="mid:51E7324A.3090405@vline.me" type="cite"> <br
        style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255);">
      <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255); display: inline !important; float: none;">Regards,</span><br
        style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255);">
      <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255); display: inline !important; float: none;">Adam Fineberg</span><br
        style="color: rgb(0, 0, 0); font-family: Times; font-size:
        medium; font-style: normal; font-variant: normal; font-weight:
        normal; letter-spacing: normal; line-height: normal; orphans:
        auto; text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: auto; word-spacing: 0px;
        -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
        255);">
      <div class="moz-forward-container" style="color: rgb(0, 0, 0);
        font-family: Times; font-size: medium; font-style: normal;
        font-variant: normal; font-weight: normal; letter-spacing:
        normal; line-height: normal; orphans: auto; text-align: start;
        text-indent: 0px; text-transform: none; white-space: normal;
        widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
        background-color: rgb(255, 255, 255);"><a moz-do-not-send="true"
          rel="nofollow" class="moz-txt-link-abbreviated"
          href="mailto:fineberg%20at%20vline.com">fineberg at vline.com</a><br>
        <br>
        -------- Original Message --------
        <table class="moz-email-headers-table" border="0"
          cellpadding="0" cellspacing="0">
          <tbody>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
              <td>New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
              <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
              <td><a moz-do-not-send="true" rel="nofollow"
                  class="moz-txt-link-abbreviated"
                  href="mailto:internet-drafts%20at%20ietf.org">internet-drafts


                  at ietf.org</a></td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
              <td>Adam Fineberg<span class="Apple-converted-space">&nbsp;</span><a
                  moz-do-not-send="true" rel="nofollow"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:fineberg%20at%20vline.com">&lt;fineberg
                  at vline.com&gt;</a></td>
            </tr>
          </tbody>
        </table>
        <br>
        <br>
        <pre style="white-space: pre-wrap; word-wrap: break-word; width: 939.5999755859375px;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
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Subject: Re: [rtcweb] e= lines (Re: Summary of Application Developers' opinions ..)
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Since Matthew was taking about API, this belongs on public-webrtc, not 
rtcweb. Please ignore.

           Harald

On 07/18/2013 06:47 AM, Harald Alvestrand wrote:
> Matthew, thanks for sending 12 messages in 3 hours to the mailing list.
>
> Just one point on which I corrected you before, but since I see you're
> repeating it:
>
> On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:
>> - What if someone adds an r= or p= or e=?
> The code for handling an e= line has been in Chrome since a very early
> release.
> You claimed in Lyon that you'd tested it and failed to have it parse,
> but I never saw the test case; you may have inserted it into the wrong
> place, rendering the SDP invalid.
>
> The SDP specification is quite strict about the order of these fields,
> exactly to make it simpler to write parsers.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Michael.Tuexen@lurchi.franken.de  Thu Jul 18 02:03:59 2013
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-05.txt
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On Jul 16, 2013, at 2:43 AM, Peter Thatcher <pthatcher@google.com> =
wrote:

> Hey Randell, can you give a short summary of the changes, if any, that =
I would or other implementors would need to know about?
Hi Peter,

the changes don't require any code change. The changes are a consequence =
of
moving the RTCWeb specific things from
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-01
to
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05
There is no technical change involved.

Best regards
Michael
>=20
>=20
> On Mon, Jul 15, 2013 at 12:23 PM, <internet-drafts@ietf.org> wrote:
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>         Title           : RTCWeb Data Channels
>         Author(s)       : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-channel-05.txt
>         Pages           : 13
>         Date            : 2013-07-15
>=20
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged =
to
>    provide protocol support for direct interactive rich communication
>    using audio, video, and data between two peers' web-browsers.  This
>    document specifies the non-media data transport aspects of the =
WebRTC
>    framework.  It provides an architectural overview of how the Stream
>    Control Transmission Protocol (SCTP) is used in the WebRTC context =
as
>    a generic transport service allowing Web Browser to exchange =
generic
>    data from peer to peer.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-channel-05
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On Jul 16, 2013, at 2:43 AM, Peter Thatcher <pthatcher@google.com> =
wrote:

> Hey Randell, can you give a short summary of the changes, if any, that =
I would or other implementors would need to know about?
Hi Peter,

using
=
http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-jesup-=
rtcweb-data-protocol-04.txt&url2=3Dhttp://tools.ietf.org/id/draft-ietf-rtc=
web-data-protocol-00.txt
you see that in addition to clarifications and cleanups
* external negotiation of data channels is described
* the packet format of the DATA_CHANNEL_OPEN message as been changed
Both of these changes require code changes.

Best regards
Michael
>=20
>=20
> On Mon, Jul 15, 2013 at 1:31 PM, <internet-drafts@ietf.org> wrote:
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>         Title           : WebRTC Data Channel Protocol
>         Author(s)       : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-00.txt
>         Pages           : 11
>         Date            : 2013-07-15
>=20
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged =
to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies an actual (minor) protocol for =
how
>    the JS-layer DataChannel objects provide the data channels between
>    the peers.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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One more question - Would consent freshness be an additional feature as par=
t of ICE functionality?

Regards
Ranjit



-----Original Message-----
From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Wednesday, July 17, 2013 7:48 PM
To: Avasarala, Ranjit (NSN - IN/Bangalore)
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 2:36 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|My comments inline <Ranjit>
|
|Regards
|Ranjit
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 2:16 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
||If many public STUN servers and call servers that implement STUN or ICEli=
te
||functionalities, are not expected to support
|
|I should have said they have no role to play. It is similar to peer-to-pee=
r ICE connectivity checks --
|those checks don't go through public STUN or call servers at all.
|[Ranjit] But there can be scenarios where RTCWebBrowsers do talk to WebRTC=
 Gateways which could be co-
|located with call handling servers. Then it won't make sense to have this =
feature.

An RTCWeb gateway wouldn't have to do anything fancier than what it already=
 does for ICE connectivity checks.=20

|
|The difference between ICE connectivity checks and consent freshness check=
s is that the former is
|performed before session establishment and the later is performed after se=
ssion establishment.
|[Ranjit] Why do you need to check consent of the peer after session is est=
ablished? Ideally I should
|check consent before session establishment. If the peer does not want to r=
eceive traffic, then session
|itself should not be established.

See draft-ietf-rtcweb-security-arch:

<snip>
4.4.  Communications and Consent Freshness

   From a security perspective, everything from here on in is a little
   anticlimactic:  Alice and Bob exchange data protected by the keys
   negotiated by DTLS.  Because of the security guarantees discussed in
   the previous sections, they know that the communications are
   encrypted and authenticated.

   The one remaining security property we need to establish is "consent
   freshness", i.e., allowing Alice to verify that Bob is still prepared
   to receive her communications so that Alice does not continue to send
   large traffic volumes to entities which went abruptly offline.  ICE
   specifies periodic STUN keepalizes but only if media is not flowing.
   Because the consent issue is more difficult here, we require RTCWeb
   implementations to periodically send keepalives.  As described in
   Section 5.3, these keepalives MUST be based on the consent freshness
   mechanism specified in [I-D.muthu-behave-consent-freshness].  If a
   keepalive fails and no new ICE channels can be established, then the
   session is terminated.
</snip>

|
||then who would have interest to support this functionality?
|
|RTCWEB browsers and browser-like platforms.
|[Ranjit] browser may implement, but it would be very customized one. I do =
not think the intermediate
|entities would want to support.

You seem to be completely missing the very purpose of consent.

|
||Ideally getting consent for receiving traffic or call should be part of s=
ignaling.
||I do not think using STUN is appropriate.
|
|ICE already does that. You think it is inappropriate?
|[Ranjit] I am saying that using STUN for consent check is not appropriate =
as STUN has other dedicated
|functionality to do - like connectivity checks, keep alives, etc.=20

See draft-ietf-rtcweb-security and draft-ietf-rtcweb-security.

|Also in cases of Symmetric NATs where STUN fails, TURN is needed. Then the=
 proposed feature would anyway fail.

The solution is built on top of ICE, so should work across these.

Muthu

|
|Muthu
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 12:16 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Hi Muthu
||
||If many public STUN servers and call servers that implement STUN or ICEli=
te functionalities, are not
||expected to support, then who would have interest to support this functio=
nality? Ideally getting
||consent for receiving traffic or call should be part of signaling. I do n=
ot think using STUN is
||appropriate.
||
||Regards
||Ranjit
||
||
||
||
||-----Original Message-----
||From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
||Sent: Wednesday, July 17, 2013 12:07 PM
||To: Avasarala, Ranjit (NSN - IN/Bangalore)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Ranjit,
||
|||Many publicly available STUN servers or those that are part of Call serv=
ers
|||may not support this.
||
||Public STUN servers and call servers are not expected to support this.
||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough
|||signaling?
||
||No. That signaling between the JS and the web server could be anything (S=
IP or XMPP or proprietary or
||whatever) and the browser has no control over it.
||
||Muthu
||
|||-----Original Message-----
|||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@ns=
n.com]
|||Sent: Wednesday, July 17, 2013 11:19 AM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Hi Muthu
|||
|||Though using STUN request/response may be good for querying about consen=
t, I feel it is overloading
|||the functionality of STUN. Many publicly available STUN servers or those=
 that are part of Call
||servers
|||may not support this.
|||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough signaling?
|||
|||
|||Regards
|||Ranjit
|||
|||
|||
|||-----Original Message-----
|||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf=
 Of ext Simon Perreault
|||Sent: Tuesday, July 16, 2013 6:34 PM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|||> |> |>    Liveness timer: If no packets have been received on the local=
 port in
|||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript tha=
t
|||> |> |>    connectivity has been lost.  The JavaScript application will =
use this
|||> |> |>    notification however it desires (e.g., cease transmitting to =
the
|||> |> |>    remote peer, provide a notification to the user, etc.).
|||> |> |
|||> |> |This seems to me like it will not fulfill the goal set in the abst=
ract:
|||> |> |"to ensure that a malicious JavaScript cannot use the browser as a
|||> |> |platform for launching attacks". If the JavaScript is free to igno=
re the
|||> |> |notification from the browser, then it has no security benefits. I=
f you
|||> |> |want to reach that goal, the browser needs to forcefully stop tran=
smitting.
|||> |>
|||> |> That goal is fulfilled by the consent checks -- the browser would s=
top transmitting everything
||on
|||> |that candidate pair, including liveness checks, if there is a consent=
 failure.
|||> |
|||> |That's not what the draft says. It says that the browser "notifies" t=
he
|||> |JS app. It needs to say that the browser MUST stop sending.
|||>
|||> No. That section is about liveness check and its intention is just not=
ify the JavaScript of a
|||potential connectivity loss. It is when the consent check fails the brow=
ser actually stops sending
|||everything. Does the draft need more text on the distinction between con=
sent and liveness tests?
|||
|||Ah! No, you're right, and the text is already perfectly clear about
|||this. No need to change. I was just confused.
|||
|||> |> |>    When not actively sending traffic on a nominated candidate pa=
ir,
|||> |> |>    performing consent freshness does not serve any purpose from =
a
|||> |> |>    security perspective.
|||> |> |
|||> |> |I don't understand what this means. Why is the "security perspecti=
ve"
|||> |> |important here? Aren't we concerned about keepalives?
|||> |>
|||> |> You mean one could use keepalives (Binding indications) for launchi=
ng attacks, so consent
|||freshness
|||> |would be required for sending them as well?
|||> |
|||> |No.
|||> |
|||> |This is a section about keepalives. I just don't understand this
|||> |sentence, and I don't understand why it talks about security.
|||>
|||> Ok, let me elaborate:
|||> - Consent freshness is not necessary when the browser is not sending a=
ny
|||>   traffic on a candidate pair.
|||> - If the browser is not performing consent freshness on a candidate pa=
ir
|||>   for the above reason, it performs ICE keepalives (or RTP keepalives)=
 to
|||>   refresh NAT bindings.
|||>
|||> Of course, the browser could continue performing consent freshness eve=
n when it is not sending any
|||other traffic on that candidate pair and skip ICE keepalives.
|||
|||Ah, ok I understand with your explanation. It makes sense. There should
|||be a way to reformulate the text to make it clearer.
|||
|||Thanks,
|||Simon
|||_______________________________________________
|||rtcweb mailing list
|||rtcweb@ietf.org
|||https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Date: Thu, 18 Jul 2013 10:28:50 +0000
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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Thursday, July 18, 2013 2:51 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|One more question - Would consent freshness be an additional feature as pa=
rt of ICE functionality?

Yes.

|
|Regards
|Ranjit
|
|
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 7:48 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 2:36 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||My comments inline <Ranjit>
||
||Regards
||Ranjit
||
||-----Original Message-----
||From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
||Sent: Wednesday, July 17, 2013 2:16 PM
||To: Avasarala, Ranjit (NSN - IN/Bangalore)
||Cc: rtcweb@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
|||If many public STUN servers and call servers that implement STUN or ICEl=
ite
|||functionalities, are not expected to support
||
||I should have said they have no role to play. It is similar to peer-to-pe=
er ICE connectivity checks -
|-
||those checks don't go through public STUN or call servers at all.
||[Ranjit] But there can be scenarios where RTCWebBrowsers do talk to WebRT=
C Gateways which could be
|co-
||located with call handling servers. Then it won't make sense to have this=
 feature.
|
|An RTCWeb gateway wouldn't have to do anything fancier than what it alread=
y does for ICE connectivity
|checks.
|
||
||The difference between ICE connectivity checks and consent freshness chec=
ks is that the former is
||performed before session establishment and the later is performed after s=
ession establishment.
||[Ranjit] Why do you need to check consent of the peer after session is es=
tablished? Ideally I should
||check consent before session establishment. If the peer does not want to =
receive traffic, then
|session
||itself should not be established.
|
|See draft-ietf-rtcweb-security-arch:
|
|<snip>
|4.4.  Communications and Consent Freshness
|
|   From a security perspective, everything from here on in is a little
|   anticlimactic:  Alice and Bob exchange data protected by the keys
|   negotiated by DTLS.  Because of the security guarantees discussed in
|   the previous sections, they know that the communications are
|   encrypted and authenticated.
|
|   The one remaining security property we need to establish is "consent
|   freshness", i.e., allowing Alice to verify that Bob is still prepared
|   to receive her communications so that Alice does not continue to send
|   large traffic volumes to entities which went abruptly offline.  ICE
|   specifies periodic STUN keepalizes but only if media is not flowing.
|   Because the consent issue is more difficult here, we require RTCWeb
|   implementations to periodically send keepalives.  As described in
|   Section 5.3, these keepalives MUST be based on the consent freshness
|   mechanism specified in [I-D.muthu-behave-consent-freshness].  If a
|   keepalive fails and no new ICE channels can be established, then the
|   session is terminated.
|</snip>
|
||
|||then who would have interest to support this functionality?
||
||RTCWEB browsers and browser-like platforms.
||[Ranjit] browser may implement, but it would be very customized one. I do=
 not think the intermediate
||entities would want to support.
|
|You seem to be completely missing the very purpose of consent.
|
||
|||Ideally getting consent for receiving traffic or call should be part of =
signaling.
|||I do not think using STUN is appropriate.
||
||ICE already does that. You think it is inappropriate?
||[Ranjit] I am saying that using STUN for consent check is not appropriate=
 as STUN has other dedicated
||functionality to do - like connectivity checks, keep alives, etc.
|
|See draft-ietf-rtcweb-security and draft-ietf-rtcweb-security.
|
||Also in cases of Symmetric NATs where STUN fails, TURN is needed. Then th=
e proposed feature would
|anyway fail.
|
|The solution is built on top of ICE, so should work across these.
|
|Muthu
|
||
||Muthu
||
|||-----Original Message-----
|||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@ns=
n.com]
|||Sent: Wednesday, July 17, 2013 12:16 PM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org
|||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Hi Muthu
|||
|||If many public STUN servers and call servers that implement STUN or ICEl=
ite functionalities, are not
|||expected to support, then who would have interest to support this functi=
onality? Ideally getting
|||consent for receiving traffic or call should be part of signaling. I do =
not think using STUN is
|||appropriate.
|||
|||Regards
|||Ranjit
|||
|||
|||
|||
|||-----Original Message-----
|||From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com=
]
|||Sent: Wednesday, July 17, 2013 12:07 PM
|||To: Avasarala, Ranjit (NSN - IN/Bangalore)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Ranjit,
|||
||||Many publicly available STUN servers or those that are part of Call ser=
vers
||||may not support this.
|||
|||Public STUN servers and call servers are not expected to support this.
|||
||||Can't there be some other neutral mechanism to query peer's consent - t=
hrough
||||signaling?
|||
|||No. That signaling between the JS and the web server could be anything (=
SIP or XMPP or proprietary
|or
|||whatever) and the browser has no control over it.
|||
|||Muthu
|||
||||-----Original Message-----
||||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@n=
sn.com]
||||Sent: Wednesday, July 17, 2013 11:19 AM
||||To: Muthu Arul Mozhi Perumal (mperumal)
||||Cc: rtcweb@ietf.org; behave@ietf.org
||||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-conse=
nt-freshness-04.txt
||||
||||Hi Muthu
||||
||||Though using STUN request/response may be good for querying about conse=
nt, I feel it is overloading
||||the functionality of STUN. Many publicly available STUN servers or thos=
e that are part of Call
|||servers
||||may not support this.
||||
||||Can't there be some other neutral mechanism to query peer's consent - t=
hrough signaling?
||||
||||
||||Regards
||||Ranjit
||||
||||
||||
||||-----Original Message-----
||||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behal=
f Of ext Simon Perreault
||||Sent: Tuesday, July 16, 2013 6:34 PM
||||To: Muthu Arul Mozhi Perumal (mperumal)
||||Cc: rtcweb@ietf.org; behave@ietf.org
||||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-conse=
nt-freshness-04.txt
||||
||||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
||||> |> |>    Liveness timer: If no packets have been received on the loca=
l port in
||||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript th=
at
||||> |> |>    connectivity has been lost.  The JavaScript application will=
 use this
||||> |> |>    notification however it desires (e.g., cease transmitting to=
 the
||||> |> |>    remote peer, provide a notification to the user, etc.).
||||> |> |
||||> |> |This seems to me like it will not fulfill the goal set in the abs=
tract:
||||> |> |"to ensure that a malicious JavaScript cannot use the browser as =
a
||||> |> |platform for launching attacks". If the JavaScript is free to ign=
ore the
||||> |> |notification from the browser, then it has no security benefits. =
If you
||||> |> |want to reach that goal, the browser needs to forcefully stop tra=
nsmitting.
||||> |>
||||> |> That goal is fulfilled by the consent checks -- the browser would =
stop transmitting everything
|||on
||||> |that candidate pair, including liveness checks, if there is a consen=
t failure.
||||> |
||||> |That's not what the draft says. It says that the browser "notifies" =
the
||||> |JS app. It needs to say that the browser MUST stop sending.
||||>
||||> No. That section is about liveness check and its intention is just no=
tify the JavaScript of a
||||potential connectivity loss. It is when the consent check fails the bro=
wser actually stops sending
||||everything. Does the draft need more text on the distinction between co=
nsent and liveness tests?
||||
||||Ah! No, you're right, and the text is already perfectly clear about
||||this. No need to change. I was just confused.
||||
||||> |> |>    When not actively sending traffic on a nominated candidate p=
air,
||||> |> |>    performing consent freshness does not serve any purpose from=
 a
||||> |> |>    security perspective.
||||> |> |
||||> |> |I don't understand what this means. Why is the "security perspect=
ive"
||||> |> |important here? Aren't we concerned about keepalives?
||||> |>
||||> |> You mean one could use keepalives (Binding indications) for launch=
ing attacks, so consent
||||freshness
||||> |would be required for sending them as well?
||||> |
||||> |No.
||||> |
||||> |This is a section about keepalives. I just don't understand this
||||> |sentence, and I don't understand why it talks about security.
||||>
||||> Ok, let me elaborate:
||||> - Consent freshness is not necessary when the browser is not sending =
any
||||>   traffic on a candidate pair.
||||> - If the browser is not performing consent freshness on a candidate p=
air
||||>   for the above reason, it performs ICE keepalives (or RTP keepalives=
) to
||||>   refresh NAT bindings.
||||>
||||> Of course, the browser could continue performing consent freshness ev=
en when it is not sending
|any
||||other traffic on that candidate pair and skip ICE keepalives.
||||
||||Ah, ok I understand with your explanation. It makes sense. There should
||||be a way to reformulate the text to make it clearer.
||||
||||Thanks,
||||Simon
||||_______________________________________________
||||rtcweb mailing list
||||rtcweb@ietf.org
||||https://www.ietf.org/mailman/listinfo/rtcweb

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From: Paul Coverdale <coverdale@sympatico.ca>
To: "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Bogineni, Kalyani'" <Kalyani.Bogineni@VerizonWireless.com>, "'Bo Burman'" <bo.burman@ericsson.com>, <rtcweb@ietf.org>
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Date: Thu, 18 Jul 2013 08:25:29 -0400
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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Yes, this text has been around for a while. I also support it.

...Paul

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>Of Hutton, Andrew
>Sent: Wednesday, July 17, 2013 4:46 AM
>To: Bogineni, Kalyani; 'Bo Burman'; rtcweb@ietf.org
>Subject: Re: [rtcweb] Some thoughts on optional audio codecs
>
>We appear to have been around this loop a number of times the text
>suggested here is exactly what was suggested by Andrew Allen back in
>January and I for one supported it them and still do - See
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg06121.html.
>
>Not sure there was a definitive conclusion to that particular consensus
>call.
>
>Andy
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Bogineni, Kalyani
>> Sent: 16 July 2013 18:02
>> To: 'Bo Burman'; rtcweb@ietf.org
>> Cc: Bogineni, Kalyani
>> Subject: Re: [rtcweb] Some thoughts on optional audio codecs
>>
>> We support the following wording proposal from Bo Burman.
>>
>> "If other suitable audio codecs are available to the browser to use,
>> it is recommended that they are also included in the offer in order =
to
>> maximize the possibility to establish the session without the need =
for
>> audio transcoding".
>>
>> Regards,
>> Kalyani Bogineni
>> Verizon
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Bo Burman
>> Sent: Monday, July 15, 2013 11:15 AM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Some thoughts on optional audio codecs
>>
>> Regarding the previous discussion on optional audio codecs in the
>> (currently expired) draft on RTCWEB audio codecs
>> (https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/):
>>
>> I think most parties involved in WebRTC work, myself included, hope
>> and believe that it will be ubiquitous and easy to include real-time
>> media conversation functionality in nearly any web context. Since it
>> will be that easy, it can be expected that most web developers need
>> not be, and thus will not be, media specialists or very knowledgeable
>about codecs.
>>
>> The definition of RTCWEB MTI codecs ensures that communication is
>> possible since at least one codec will always be found, but it is not
>> possible to claim the resulting communication to be optimum for every
>> possible context.
>>
>> Even if WebRTC will be close to ubiquitous, there will for quite some
>> time likely be a desire to reach real-time media domains and devices
>> that were not originally designed for and thus are not optimized for
>> use with WebRTC. A communication device that is not designed solely
>> for WebRTC use will likely include functionality and codecs also for
>> its "native" domain.
>>
>> Any added cost of not being able to use existing "native" codecs will
>> vary both in amount and where the cost has to be taken. Eliminating =
it
>> is indeed an optimization, but the total cost savings may still be
>> significant.
>>
>> With the current design and to my understanding, it will be the
>> browser vendor's choice to add optional codecs, including any =
"native"
>> domain codecs.  The choice may possibly be delegated to individual =
web
>> developers making use of WebRTC functionality. A browser vendor will
>> arguably have to know each target platform to some extent, but it can
>> hardly be assumed that a web developer knows the capabilities of all
>> devices that will use the WebRTC-enabled site unless the browser can
>> provide the needed information. There is a risk that "native" codecs
>> in devices are not well handled, unless the motivations and methods =
to
>> make use of them are better specified.
>>
>> While any audio codecs besides the MTI ones are clearly optional, I
>> believe the suggested text addition on optional audio codecs to the
>> RTCWEB audio draft in Ticket #12
>> (http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/12#) to be too =
brief
>> considering the above.
>>
>> In that draft, I would prefer something more in line with:
>>
>> "If other suitable audio codecs are available to the browser to use,
>> it is recommended that they are also included in the offer in order =
to
>> maximize the possibility to establish the session without the need =
for
>> audio transcoding".
>>
>> Assuming that the browser vendor (or web developer) is sufficiently
>> concerned with codecs to read the audio codecs draft (or the
>> corresponding RFC to-be), the above text may, as a start, give some
>> added guidance why non-MTI codecs may be desirable to consider in
>> addition to the MTI ones.
>>
>> Cheers,
>> Bo
>>
>> Multimedia Technologies
>> Ericsson Research
>> F=E4r=F6gatan 6
>> SE-164 80, Kista, Sweden
>> www.ericsson.com
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


From rajmohanbanavi@gmail.com  Thu Jul 18 05:28:18 2013
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From: Rajmohan Banavi <rajmohanbanavi@gmail.com>
To: Justin Uberti <juberti@google.com>
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--047d7bdc119a41cf0c04e1c85826
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Hi  Justin, comments inline with additional comments at the end.

Correct, but that is a significant advantage, especially when the web and
> TURN servers are separate entities.
>

IMHO, the standard spec should not be based on any specific implementation
method. Now I agree that the TURN server validation becomes easier using
this method because it does not have to do some sort of lookup. However,
some another TURN server implementer can achieve the same functionality by
providing REST API but using randomly generated pasword (rather than use
HMAC) and using some sort of lookup for TURN username when TURN ALLOCATE
request is received.

Now lets look at the collateral affects of using this proposed
implementation way
- TURN server needs to perform MESSAGE-INTEGRITY validation across two
secrets (worst case due to key rotation) when TURN ALLOCATE request is
received
- TURN username needs to consist of entire state (user identifier + expiry
time). And upon receiving the ALLOCATE request, TURN server must check if
the username is still valid based on expiry time part (of username)
- Time sync needs to be maintained across the web server and the TURN
server.

All of the above are not needed if one does not use the implementation
method specified in the draft but still provide REST API to generate
ephemeral credentials. How these ephemeral credentials are generated and
how the TURN requests are validated is an implementation decision. And it
is not desirable to have standards which caters to each implementation
method.


> The WebRTC client application doesn't see the secret key, this is only
> shared between the web and TURN servers.
>

What does the web server do with the secret key? This is not clear from the
text.

Additional comments.

   1. Sec 4.2 Server - "Note that the REALM value supplied by the server is
   not meaningful in this context, and can be set to any valid value". Which
   realm value is being referred here? The REALM attribute value in the 401
   challenge response sent by the TURN server (in response to initial ALLOCATE
   request)?
   2. The implementation method proposed is implicit and not clear from the
   draft text for the reader. The text states that username must be so and so,
   password must be HMAC processed etc, but it is not clear why these need to
   be done. It must be made clear somewhere in the draft that these are being
   proposed to ensure that the TURN server can validate the TURN ALLOCATE
   requests by performing HMAC on the USERNAME attribute received. And that
   the TURN server does not have to maintain/store ephemeral credentials. This
   will help the reader to comprehend the draft in a better way.

Hope it is clear.

Thanks,
Rajmohan Banavi

--047d7bdc119a41cf0c04e1c85826
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<div dir=3D"ltr">Hi =A0Justin, comments inline with additional comments at =
the end.<div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-widt=
h:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-le=
ft:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>Correct, but that is a significant advantage, especially when the web and =
TURN servers are separate entities.=A0</div></div></div></div></blockquote>=
<div>

<br></div><div>IMHO, the standard spec should not be based on any specific =
implementation method. Now I agree that the TURN server validation becomes =
easier using this method because it does not have to do some sort of lookup=
. However, some another TURN server implementer can achieve the same functi=
onality by providing REST API but using randomly generated pasword (rather =
than use HMAC) and using some sort of lookup for TURN username when TURN AL=
LOCATE request is received.</div>

<div><br></div><div>Now lets look at the collateral affects of using this p=
roposed implementation way</div><div>- TURN server needs to perform MESSAGE=
-INTEGRITY validation across two secrets (worst case due to key rotation) w=
hen TURN ALLOCATE request is received</div>

<div>- TURN username needs to consist of entire state (user identifier + ex=
piry time). And upon receiving the ALLOCATE request, TURN server must check=
 if the username is still valid based on expiry time part (of username)</di=
v>

<div>- Time sync needs to be maintained across the web server and the TURN =
server.</div><div><br></div><div>All of the above are not needed if one doe=
s not use the implementation method specified in the draft but still provid=
e REST API to generate ephemeral credentials. How these ephemeral credentia=
ls are generated and how the TURN requests are validated is an implementati=
on decision. And it is not desirable to have standards which caters to each=
 implementation method.=A0</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>The WebRTC client application doesn&#39;t see the secret key, this is only=
 shared between the web and TURN servers.</div></div></div></div>
</blockquote><div><br></div><div>What does the web server do with the secre=
t key? This is not clear from the text.</div><div><br></div><div>Additional=
 comments.</div><div><ol><li>Sec 4.2 Server - &quot;Note that the REALM val=
ue supplied by the server is not meaningful in this context, and can be set=
 to any valid value&quot;. Which realm value is being referred here? The RE=
ALM attribute value in the 401 challenge response sent by the TURN server (=
in response to initial ALLOCATE request)?<br>

</li><li>The implementation method proposed is implicit and not clear from =
the draft text for the reader. The text states that username must be so and=
 so, password must be HMAC processed etc, but it is not clear why these nee=
d to be done. It must be made clear somewhere in the draft that these are b=
eing proposed to ensure that the TURN server can validate the TURN ALLOCATE=
 requests by performing HMAC on the USERNAME attribute received. And that t=
he TURN server does not have to maintain/store ephemeral credentials. This =
will help the reader to comprehend the draft in a better way.</li>
</ol></div><div>Hope it is clear.</div><div><br></div><div>Thanks,</div><di=
v>Rajmohan Banavi</div><div><br></div></div></div></div>

--047d7bdc119a41cf0c04e1c85826--

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From: Peter Thatcher <pthatcher@google.com>
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On Wed, Jul 17, 2013 at 9:47 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

> Matthew, thanks for sending 12 messages in 3 hours to the mailing list.
>
>
Given how many active threads and subthreads are going on at once, 12
messages doesn't seem like that many.  I've lost track of how many
centithreads there are now.



> Just one point on which I corrected you before, but since I see you're
> repeating it:
>
> On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:
> > - What if someone adds an r= or p= or e=?
>
> The code for handling an e= line has been in Chrome since a very early
> release.
> You claimed in Lyon that you'd tested it and failed to have it parse,
> but I never saw the test case; you may have inserted it into the wrong
> place, rendering the SDP invalid.
>
> The SDP specification is quite strict about the order of these fields,
> exactly to make it simpler to write parsers.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">On Wed, Jul 17, 2013 at 9:47 PM, Harald Alvestrand <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">ha=
rald@alvestrand.no</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div=
 class=3D"gmail_quote">


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Matthew, thanks for sending 12 messages in 3=
 hours to the mailing list.<br>
<br></blockquote><div><br></div><div>Given how many active threads and subt=
hreads are going on at once, 12 messages doesn&#39;t seem like that many. =
=C2=A0I&#39;ve lost track of how many centithreads there are now.</div><div=
>

=C2=A0</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

Just one point on which I corrected you before, but since I see you&#39;re<=
br>
repeating it:<br>
<br>
On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:<br>
&gt; - What if someone adds an r=3D or p=3D or e=3D?<br>
<br>
The code for handling an e=3D line has been in Chrome since a very early<br=
>
release.<br>
You claimed in Lyon that you&#39;d tested it and failed to have it parse,<b=
r>
but I never saw the test case; you may have inserted it into the wrong<br>
place, rendering the SDP invalid.<br>
<br>
The SDP specification is quite strict about the order of these fields,<br>
exactly to make it simpler to write parsers.<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--bcaec51f99e5253bb004e1ca7ee9--

From matthew.kaufman@skype.net  Thu Jul 18 08:07:46 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Peter Thatcher <pthatcher@google.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] e= lines (Re: Summary of Application Developers' opinions ..)
Thread-Index: AQHOg3H6C1QS7WP740Kwrfdnz1pKNZlqiHAAgAABWc0=
Date: Thu, 18 Jul 2013 15:06:46 +0000
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] e= lines (Re: Summary of Application Developers' opinions ..)
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By waiting, I reduced what would have been 30 or 40 messages to "only" 12. =
Just pretend that I spaced them out over the course of the discussion and b=
e happy.

Matthew Kaufman

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Peter =
Thatcher [pthatcher@google.com]
Sent: Thursday, July 18, 2013 8:01 AM
To: Harald Alvestrand
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] e=3D lines (Re: Summary of Application Developers' op=
inions ..)

On Wed, Jul 17, 2013 at 9:47 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
Matthew, thanks for sending 12 messages in 3 hours to the mailing list.


Given how many active threads and subthreads are going on at once, 12 messa=
ges doesn't seem like that many.  I've lost track of how many centithreads =
there are now.


Just one point on which I corrected you before, but since I see you're
repeating it:

On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:
> - What if someone adds an r=3D or p=3D or e=3D?

The code for handling an e=3D line has been in Chrome since a very early
release.
You claimed in Lyon that you'd tested it and failed to have it parse,
but I never saw the test case; you may have inserted it into the wrong
place, rendering the SDP invalid.

The SDP specification is quite strict about the order of these fields,
exactly to make it simpler to write parsers.

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


--_000_AE1A6B5FD507DC4FB3C5166F3A05A484237147CBTK5EX14MBXC266r_
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1">
<style type=3D"text/css" id=3D"owaParaStyle"></style>
</head>
<body fpstyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">By waiting, I reduced what would have been 30 or 40 messages to &quo=
t;only&quot; 12. Just pretend that I spaced them out over the course of the=
 discussion and be happy.
<div><br>
</div>
<div>Matthew Kaufman</div>
<div><br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF536001" style=3D"direction: ltr;"><font face=3D"Tahoma" si=
ze=3D"2" color=3D"#000000"><b>From:</b> rtcweb-bounces@ietf.org [rtcweb-bou=
nces@ietf.org] on behalf of Peter Thatcher [pthatcher@google.com]<br>
<b>Sent:</b> Thursday, July 18, 2013 8:01 AM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] e=3D lines (Re: Summary of Application Develop=
ers' opinions ..)<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">On Wed, Jul 17, 2013 at 9:47 PM, Harald Alvestrand <span d=
ir=3D"ltr">
&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvest=
rand.no</a>&gt;</span> wrote:<br>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
Matthew, thanks for sending 12 messages in 3 hours to the mailing list.<br>
<br>
</blockquote>
<div><br>
</div>
<div>Given how many active threads and subthreads are going on at once, 12 =
messages doesn't seem like that many. &nbsp;I've lost track of how many cen=
tithreads there are now.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
Just one point on which I corrected you before, but since I see you're<br>
repeating it:<br>
<br>
On 07/17/2013 11:58 PM, Matthew Kaufman (SKYPE) wrote:<br>
&gt; - What if someone adds an r=3D or p=3D or e=3D?<br>
<br>
The code for handling an e=3D line has been in Chrome since a very early<br=
>
release.<br>
You claimed in Lyon that you'd tested it and failed to have it parse,<br>
but I never saw the test case; you may have inserted it into the wrong<br>
place, rendering the SDP invalid.<br>
<br>
The SDP specification is quite strict about the order of these fields,<br>
exactly to make it simpler to write parsers.<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_AE1A6B5FD507DC4FB3C5166F3A05A484237147CBTK5EX14MBXC266r_--

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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 18 Jul 2013 08:12:04 -0700
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Wed, Jul 17, 2013 at 2:58 PM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

> Bernard Aboba:
> >
> > The problem was that we never defined what mangling browsers had to
> > support. Given all the SDP specs that is actually a huge work item.
>
>
> This is exactly what my slides that weren't presented last November
> started to touch upon. It only takes a few minutes with RFC3264 and its
> references to start documenting cases that are clearly "valid SDP
> offer/answer" and yet for which I cannot for the life of me figure out what
> a browser should do if they're presented.
>

I believe I began paying attention to the mailing lists after you sent out
theses slides that you didn't present.  I'm interested in seeing them, and
while I could dig through archives to find them, if convenient, could you
please give me a link to the slides?  Thanks.


>
> Just off the top of my head:
>
> Can I...
> - Change the t= line to be something other than t=0 0?
> - Change the rtpmap associations before calling setLocal?
> - Change a=sendrecv to a=recvonly before calling setLocal?
> - What do you do when you see a=content:sl ?
> - What if someone adds an r= or p= or e=?
> - What is the RFC that describes a=group:BUNDLE (as seen in some browser
> implementations)?
> - Can I remove a=group:BUNDLE (or add it) before calling setLocal?
> - How about removing a=rtcp-mux?
> - Should I do something special at my end if you set
> a=ice-options:google-ice ?
> - If you put a=ssrc lines in there, can I change the ssrc before passing
> it back to setLocal?
> - Can I delete the a=ssrc lines?
> - Is a=rtcp:1 IN IP4 0.0.0.0 valid or not?
> - What about 0.0.0.0 in the o= line?
> - If I get a bunch of a=candidate lines, can I swap them around to change
> the priority before calling setLocal?
> - What if someone claims SAVP instead of SAVPF but gives me rtcp info?
>
> At the *very least* for each and every line that comes from createOffer
> and createAnswer you must be able to answer the following:
> - Can I delete it?
> - Duplicate it?
> - Change it?
> - If not, how are violation handled? (Both when passed to another browser
> at the far end and when these modifications happen before calling setLocal)
>  - And can I add additional valid SDP to what came from createOffer or
> createAnswer and pass it back to setLocal or not?
>
> We appear to be nowhere near a document which explains the answers to
> these questions, certainly not in the W3C WG, and not in any IETF RFC or
> draft I can locate.
>



> Matthew Kaufman
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 17, 2013 at 2:58 PM, Matthew Kaufman (SKYPE) <span dir=
=3D"ltr">&lt;<a href=3D"mailto:matthew.kaufman@skype.net" target=3D"_blank"=
>matthew.kaufman@skype.net</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Bernard Aboba:<br>
<div class=3D"im">&gt;<br>
&gt; The problem was that we never defined what mangling browsers had to<br=
>
&gt; support. Given all the SDP specs that is actually a huge work item.<br=
>
<br>
<br>
</div>This is exactly what my slides that weren&#39;t presented last Novemb=
er started to touch upon. It only takes a few minutes with RFC3264 and its =
references to start documenting cases that are clearly &quot;valid SDP offe=
r/answer&quot; and yet for which I cannot for the life of me figure out wha=
t a browser should do if they&#39;re presented.<br>

</blockquote><div><br></div><div>I believe I began paying attention to the =
mailing lists after you sent out theses slides that you didn&#39;t present.=
 =C2=A0I&#39;m interested in seeing them, and while I could dig through arc=
hives to find them, if convenient, could you please give me a link to the s=
lides? =C2=A0Thanks.=C2=A0</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Just off the top of my head:<br>
<br>
Can I...<br>
- Change the t=3D line to be something other than t=3D0 0?<br>
- Change the rtpmap associations before calling setLocal?<br>
- Change a=3Dsendrecv to a=3Drecvonly before calling setLocal?<br>
- What do you do when you see a=3Dcontent:sl ?<br>
- What if someone adds an r=3D or p=3D or e=3D?<br>
- What is the RFC that describes a=3Dgroup:BUNDLE (as seen in some browser =
implementations)?<br>
- Can I remove a=3Dgroup:BUNDLE (or add it) before calling setLocal?<br>
- How about removing a=3Drtcp-mux?<br>
- Should I do something special at my end if you set a=3Dice-options:google=
-ice ?<br>
- If you put a=3Dssrc lines in there, can I change the ssrc before passing =
it back to setLocal?<br>
- Can I delete the a=3Dssrc lines?<br>
- Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?<br>
- What about 0.0.0.0 in the o=3D line?<br>
- If I get a bunch of a=3Dcandidate lines, can I swap them around to change=
 the priority before calling setLocal?<br>
- What if someone claims SAVP instead of SAVPF but gives me rtcp info?<br>
<br>
At the *very least* for each and every line that comes from createOffer and=
 createAnswer you must be able to answer the following:<br>
- Can I delete it?<br>
- Duplicate it?<br>
- Change it?<br>
- If not, how are violation handled? (Both when passed to another browser a=
t the far end and when these modifications happen before calling setLocal)<=
br>
=C2=A0- And can I add additional valid SDP to what came from createOffer or=
 createAnswer and pass it back to setLocal or not?<br>
<br>
We appear to be nowhere near a document which explains the answers to these=
 questions, certainly not in the W3C WG, and not in any IETF RFC or draft I=
 can locate.<br></blockquote><div><br></div><div><br></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">


<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Matthew Kaufman<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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References: <20130715203130.5677.31855.idtracker@ietfa.amsl.com> <CAJrXDUFGM9+bw7KcYxwK9s_MbzW2L_1xjshTocgTKxpur345Nw@mail.gmail.com> <DD542B20-8EA3-45F8-B9F1-7991F207F2A1@lurchi.franken.de>
From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 18 Jul 2013 08:19:18 -0700
Message-ID: <CAJrXDUG5sB63w3p6kDDuNT=_+Pq_m7ZmH+HdX_9c0sVXYBk2gw@mail.gmail.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-00.txt
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On Thu, Jul 18, 2013 at 2:11 AM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> On Jul 16, 2013, at 2:43 AM, Peter Thatcher <pthatcher@google.com> wrote:
>
> > Hey Randell, can you give a short summary of the changes, if any, that I
> would or other implementors would need to know about?
> Hi Peter,
>
> using
>
> http://tools.ietf.org/rfcdiff?url1=http://tools.ietf.org/id/draft-jesup-rtcweb-data-protocol-04.txt&url2=http://tools.ietf.org/id/draft-ietf-rtcweb-data-protocol-00.txt
> you see that in addition to clarifications and cleanups
>

Oh, now that's a slick tool.  Thanks for the link.


> * external negotiation of data channels is described
>
* the packet format of the DATA_CHANNEL_OPEN message as been changed
>

I just wrote the code to serialize and deserialize that message.  Please
tell me we're not going to change it again :).


> Both of these changes require code changes.
>
> Best regards
> Michael
> >
> >
> > On Mon, Jul 15, 2013 at 1:31 PM, <internet-drafts@ietf.org> wrote:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Real-Time Communication in
> WEB-browsers Working Group of the IETF.
> >
> >         Title           : WebRTC Data Channel Protocol
> >         Author(s)       : Randell Jesup
> >                           Salvatore Loreto
> >                           Michael Tuexen
> >         Filename        : draft-ietf-rtcweb-data-protocol-00.txt
> >         Pages           : 11
> >         Date            : 2013-07-15
> >
> > Abstract:
> >    The Web Real-Time Communication (WebRTC) working group is charged to
> >    provide protocols to support for direct interactive rich
> >    communication using audio, video, and data between two peers' web-
> >    browsers.  This document specifies an actual (minor) protocol for how
> >    the JS-layer DataChannel objects provide the data channels between
> >    the peers.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Jul 18, 2013 at 2:11 AM, Michael Tuexen <span dir=3D"ltr">&=
lt;<a href=3D"mailto:Michael.Tuexen@lurchi.franken.de" target=3D"_blank">Mi=
chael.Tuexen@lurchi.franken.de</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>On Jul 16, 2013, at 2:43 AM, Peter That=
cher &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatche=
r@google.com</a>&gt; wrote:<br>



<br>
&gt; Hey Randell, can you give a short summary of the changes, if any, that=
 I would or other implementors would need to know about?<br>
</div>Hi Peter,<br>
<br>
using<br>
<a href=3D"http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-jesup-rtcweb-data-protocol-04.txt&amp;url2=3Dhttp://tools.ietf.org/id/d=
raft-ietf-rtcweb-data-protocol-00.txt" target=3D"_blank">http://tools.ietf.=
org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-jesup-rtcweb-data-protoco=
l-04.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-ietf-rtcweb-data-protoco=
l-00.txt</a><br>



you see that in addition to clarifications and cleanups<br></blockquote><di=
v><br></div><div>Oh, now that&#39;s a slick tool. =C2=A0Thanks for the link=
.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


* external negotiation of data channels is described<br></blockquote><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">* the packet format of the DATA_CHANNEL_OPEN messa=
ge as been changed<br>

</blockquote><div><br></div><div>I just wrote the code to serialize and des=
erialize that message. =C2=A0Please tell me we&#39;re not going to change i=
t again :).</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">



Both of these changes require code changes.<br>
<br>
Best regards<br>
<span><font color=3D"#888888">Michael<br>
</font></span><div><div>&gt;<br>
&gt;<br>
&gt; On Mon, Jul 15, 2013 at 1:31 PM, &lt;<a href=3D"mailto:internet-drafts=
@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt; wrote:<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt; =C2=A0This draft is a work item of the Real-Time Communication in WEB-=
browsers Working Group of the IETF.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 WebRTC Data Channel Protocol<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Author(s) =C2=A0 =C2=A0 =C2=A0 : Randell J=
esup<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 Salvatore Loreto<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 Michael Tuexen<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draf=
t-ietf-rtcweb-data-protocol-00.txt<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 11<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0: 2013-07-15<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =C2=A0 =C2=A0The Web Real-Time Communication (WebRTC) working group is=
 charged to<br>
&gt; =C2=A0 =C2=A0provide protocols to support for direct interactive rich<=
br>
&gt; =C2=A0 =C2=A0communication using audio, video, and data between two pe=
ers&#39; web-<br>
&gt; =C2=A0 =C2=A0browsers. =C2=A0This document specifies an actual (minor)=
 protocol for how<br>
&gt; =C2=A0 =C2=A0the JS-layer DataChannel objects provide the data channel=
s between<br>
&gt; =C2=A0 =C2=A0the peers.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-pro=
tocol" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb=
-data-protocol</a><br>
&gt;<br>
&gt; There&#39;s also a htmlized version available at:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-=
00" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-pro=
tocol-00</a><br>
&gt;<br>
&gt;<br>
&gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp:=
//ftp.ietf.org/internet-drafts/</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</div></div></blockquote></div><br></div></div>

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From fineberg@vline.me  Thu Jul 18 08:43:21 2013
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Date: Thu, 18 Jul 2013 08:43:07 -0700
From: Adam Fineberg <fineberg@vline.me>
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Yes, I apologize for not adding avtext as a cc.

Regards,
Adam

On 7/18/13 1:16 AM, Harald Alvestrand wrote:
> On 07/18/2013 02:09 AM, Adam Fineberg wrote:
>> Hi,
>>
>> I'm working on WebRTC services and have found that while developing 
>> services that forward VP8 video streams if we want to take advantage 
>> of the VP8 temporal scaling we must get the temporal layer 
>> information from the RTP header which requires us to decrypt the SRTP 
>> packets. This is undesirable both because the middle-box needs to 
>> have access to the keys as well as the because of the added overhead 
>> of the decrypt/encrypt cycle. This draft proposes an RTP header 
>> extension that will allow us to use the VP8 temporal layer 
>> information included in the header extension and therefore do 
>> forwarding without SRTP decryption. Comments welcome.
>
> Just checking - you named your draft with the -avtext- convention; 
> this means that you want comments to go to avtext, yes?
>
>
>>
>> Regards,
>> Adam Fineberg
>> fineberg at vline.com
>>
>> -------- Original Message --------
>> Subject: 	New Version Notification for 
>> draft-fineberg-avtext-temporal-layer-ext-00.txt
>> Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>> From: 	internet-drafts at ietf.org
>> To: 	Adam Fineberg<fineberg at vline.com>
>>
>>
>>
>> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>> has been successfully submitted by Adam Fineberg and posted to the
>> IETF repository.
>>
>> Filename:	 draft-fineberg-avtext-temporal-layer-ext
>> Revision:	 00
>> Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>> Creation date:	 2013-07-08
>> Group:		 Individual Submission
>> Number of pages: 6
>> URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>> Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>> Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>
>>
>> Abstract:
>>     This document defines a mechanism by which packets of Real-Time
>>     Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>     indicate, in an RTP header extension, the temporal layer information
>>     about the frame encoded in the RTP packet.  This information can be
>>     used in a middlebox performing bandwidth management of streams
>>     without requiring it to decrypt the streams.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>

-- 
Regards,
Adam


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    Yes, I apologize for not adding avtext as a cc.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/18/13 1:16 AM, Harald Alvestrand
      wrote:<br>
    </div>
    <blockquote cite="mid:51E7A472.3010206@alvestrand.no" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 07/18/2013 02:09 AM, Adam Fineberg
        wrote:<br>
      </div>
      <blockquote cite="mid:51E7324A.3090405@vline.me" type="cite">
        <meta http-equiv="content-type" content="text/html;
          charset=ISO-8859-1">
        <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255); display: inline !important; float: none;">Hi,</span><br
          style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255);">
        <br style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255);">
        <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255); display: inline !important; float: none;">I'm
          working on WebRTC services and have found that while
          developing services that forward VP8 video streams if we want
          to take advantage of the VP8 temporal scaling we must get the
          temporal layer information from the RTP header which requires
          us to decrypt the SRTP packets. This is undesirable both
          because the middle-box needs to have access to the keys as
          well as the because of the added overhead of the
          decrypt/encrypt cycle. This draft proposes an RTP header
          extension that will allow us to use the VP8 temporal layer
          information included in the header extension and therefore do
          forwarding without SRTP decryption. Comments welcome.</span><br
          style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255);">
      </blockquote>
      <br>
      Just checking - you named your draft with the -avtext- convention;
      this means that you want comments to go to avtext, yes?<br>
      <br>
      <br>
      <blockquote cite="mid:51E7324A.3090405@vline.me" type="cite"> <br
          style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255);">
        <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255); display: inline !important; float: none;">Regards,</span><br
          style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255);">
        <span style="color: rgb(0, 0, 0); font-family: Times; font-size:
          medium; font-style: normal; font-variant: normal; font-weight:
          normal; letter-spacing: normal; line-height: normal; orphans:
          auto; text-align: start; text-indent: 0px; text-transform:
          none; white-space: normal; widows: auto; word-spacing: 0px;
          -webkit-text-stroke-width: 0px; background-color: rgb(255,
          255, 255); display: inline !important; float: none;">Adam
          Fineberg</span><br style="color: rgb(0, 0, 0); font-family:
          Times; font-size: medium; font-style: normal; font-variant:
          normal; font-weight: normal; letter-spacing: normal;
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          0px; background-color: rgb(255, 255, 255);">
        <div class="moz-forward-container" style="color: rgb(0, 0, 0);
          font-family: Times; font-size: medium; font-style: normal;
          font-variant: normal; font-weight: normal; letter-spacing:
          normal; line-height: normal; orphans: auto; text-align: start;
          text-indent: 0px; text-transform: none; white-space: normal;
          widows: auto; word-spacing: 0px; -webkit-text-stroke-width:
          0px; background-color: rgb(255, 255, 255);"><a
            moz-do-not-send="true" rel="nofollow"
            class="moz-txt-link-abbreviated"
            href="mailto:fineberg%20at%20vline.com">fineberg at
            vline.com</a><br>
          <br>
          -------- Original Message --------
          <table class="moz-email-headers-table" border="0"
            cellpadding="0" cellspacing="0">
            <tbody>
              <tr>
                <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
                <td>New Version Notification for
                  draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
              </tr>
              <tr>
                <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
                <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
              </tr>
              <tr>
                <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
                <td><a moz-do-not-send="true" rel="nofollow"
                    class="moz-txt-link-abbreviated"
                    href="mailto:internet-drafts%20at%20ietf.org">internet-drafts



                    at ietf.org</a></td>
              </tr>
              <tr>
                <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
                <td>Adam Fineberg<span class="Apple-converted-space">&nbsp;</span><a
                    moz-do-not-send="true" rel="nofollow"
                    class="moz-txt-link-rfc2396E"
                    href="mailto:fineberg%20at%20vline.com">&lt;fineberg
                    at vline.com&gt;</a></td>
              </tr>
            </tbody>
          </table>
          <br>
          <br>
          <pre style="white-space: pre-wrap; word-wrap: break-word; width: 939.5999755859375px;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" rel="nofollow" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
        </div>
        <br>
        <fieldset class="mimeAttachmentHeader"></fieldset>
        <br>
        <pre wrap="">_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
      <br>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
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Date: Thu, 18 Jul 2013 08:45:38 -0700
From: Adam Fineberg <fineberg@vline.me>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Bernard,

Good question.  I'm not familiar enough with the parameter requirements 
of all other scalable codecs to be able to generalize. If you'd like to 
help specify them, I'd be fine revising the draft to generalize.

Regards,
Adam

On 7/17/13 8:26 PM, Bernard Aboba wrote:
> Since the need is not codec specific (e.g. it arises with any codec 
> supporting temporal, spatial and quality scalability), why
>  a VP8-specific RTP extension?
>
> ------------------------------------------------------------------------
> Date: Wed, 17 Jul 2013 17:09:46 -0700
> From: fineberg@vline.me
> To: rtcweb@ietf.org
> Subject: [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Hi,
>
> I'm working on WebRTC services and have found that while developing 
> services that forward VP8 video streams if we want to take advantage 
> of the VP8 temporal scaling we must get the temporal layer information 
> from the RTP header which requires us to decrypt the SRTP packets. 
> This is undesirable both because the middle-box needs to have access 
> to the keys as well as the because of the added overhead of the 
> decrypt/encrypt cycle. This draft proposes an RTP header extension 
> that will allow us to use the VP8 temporal layer information included 
> in the header extension and therefore do forwarding without SRTP 
> decryption. Comments welcome.
>
> Regards,
> Adam Fineberg
> fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>
> -------- Original Message --------
> Subject: 	New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
> Date: 	Tue, 09 Jul 2013 10:02:05 -0700
> From: 	internet-drafts at ietf.org 
> <mailto:internet-drafts%20at%20ietf.org>
> To: 	Adam Fineberg<fineberg at vline.com> 
> <mailto:fineberg%20at%20vline.com>
>
>
>
> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
> has been successfully submitted by Adam Fineberg and posted to the
> IETF repository.
>
> Filename:	 draft-fineberg-avtext-temporal-layer-ext
> Revision:	 00
> Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
> Creation date:	 2013-07-08
> Group:		 Individual Submission
> Number of pages: 6
> URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
> Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
> Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>
> Abstract:
>     This document defines a mechanism by which packets of Real-Time
>     Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>     indicate, in an RTP header extension, the temporal layer information
>     about the frame encoded in the RTP packet.  This information can be
>     used in a middlebox performing bandwidth management of streams
>     without requiring it to decrypt the streams.
>
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

-- 
Regards,
Adam


--------------080806000200010002090007
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      http-equiv="Content-Type">
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    Bernard,<br>
    <br>
    Good question.&nbsp; I'm not familiar enough with the parameter
    requirements of all other scalable codecs to be able to generalize.&nbsp;
    If you'd like to help specify them, I'd be fine revising the draft
    to generalize.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/17/13 8:26 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style>
      <div dir="ltr">Since the need is not codec specific (e.g. it
        arises with any codec supporting temporal, spatial and quality
        scalability), why <br>
        &nbsp;a VP8-specific RTP extension? <br>
        &nbsp;<br>
        <div>
          <hr id="stopSpelling">Date: Wed, 17 Jul 2013 17:09:46 -0700<br>
          From: <a class="moz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
          To: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          Subject: [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          <span style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            display: inline !important; white-space: normal;
            background-color: rgb(255, 255, 255);">Hi,</span><br
            style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            white-space: normal; background-color: rgb(255, 255, 255);">
          <br style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            white-space: normal; background-color: rgb(255, 255, 255);">
          <span style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            display: inline !important; white-space: normal;
            background-color: rgb(255, 255, 255);">I'm working on WebRTC
            services and have found that while developing services that
            forward VP8 video streams if we want to take advantage of
            the VP8 temporal scaling we must get the temporal layer
            information from the RTP header which requires us to decrypt
            the SRTP packets. This is undesirable both because the
            middle-box needs to have access to the keys as well as the
            because of the added overhead of the decrypt/encrypt cycle.
            This draft proposes an RTP header extension that will allow
            us to use the VP8 temporal layer information included in the
            header extension and therefore do forwarding without SRTP
            decryption. Comments welcome.</span><br style="color: rgb(0,
            0, 0); text-transform: none; text-indent: 0px;
            letter-spacing: normal; word-spacing: 0px; white-space:
            normal; background-color: rgb(255, 255, 255);">
          <br style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            white-space: normal; background-color: rgb(255, 255, 255);">
          <span style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            display: inline !important; white-space: normal;
            background-color: rgb(255, 255, 255);">Regards,</span><br
            style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            white-space: normal; background-color: rgb(255, 255, 255);">
          <span style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            display: inline !important; white-space: normal;
            background-color: rgb(255, 255, 255);">Adam Fineberg</span><br
            style="color: rgb(0, 0, 0); text-transform: none;
            text-indent: 0px; letter-spacing: normal; word-spacing: 0px;
            white-space: normal; background-color: rgb(255, 255, 255);">
          <div class="ecxmoz-forward-container" style="color: rgb(0, 0,
            0); text-transform: none; text-indent: 0px; letter-spacing:
            normal; word-spacing: 0px; white-space: normal;
            background-color: rgb(255, 255, 255);"><a
              moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated"
              href="mailto:fineberg%20at%20vline.com" rel="nofollow">fineberg
              at vline.com</a><br>
            <br>
            -------- Original Message --------
            <table class="ecxmoz-email-headers-table" border="0"
              cellpadding="0" cellspacing="0">
              <tbody>
                <tr>
                  <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
                  <td>New Version Notification for
                    draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                </tr>
                <tr>
                  <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
                  <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                </tr>
                <tr>
                  <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
                  <td><a moz-do-not-send="true"
                      class="ecxmoz-txt-link-abbreviated"
                      href="mailto:internet-drafts%20at%20ietf.org"
                      rel="nofollow">internet-drafts at ietf.org</a></td>
                </tr>
                <tr>
                  <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
                  <td>Adam Fineberg<span
                      class="ecxApple-converted-space">&nbsp;</span><a
                      moz-do-not-send="true"
                      class="ecxmoz-txt-link-rfc2396E"
                      href="mailto:fineberg%20at%20vline.com"
                      rel="nofollow">&lt;fineberg at vline.com&gt;</a></td>
                </tr>
              </tbody>
            </table>
            <br>
            <br>
            <pre style="width: 939.59px; white-space: pre-wrap; -ms-word-wrap: break-word;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank" rel="nofollow">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank" rel="nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank" rel="nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
          </div>
          <br>
          _______________________________________________
          rtcweb mailing list
          <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
          <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
  </body>
</html>

--------------080806000200010002090007--

From hadriel.kaplan@oracle.com  Thu Jul 18 09:25:02 2013
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On Jul 17, 2013, at 4:48 PM, Matthew Kaufman (SKYPE) =
<matthew.kaufman@skype.net> wrote:
>> Given those personal feelings, I didn't much care either way, so long =
as a
>> decision was made - although I still preferred having SDES to avoid =
the early
>> media clipping problem.  So when I was going to present on this topic =
back in
>> the Vancouver meeting, my last slide basically said: "If we support =
SDES and it
>> turns out later we were wrong, we can always rescind it".  This was =
based on
>> the notion that browsers get updated all the time, especially for =
security
>> issues.
>>=20
>> But later it occurred to me that's actually the Achilles' heel of =
supporting SDES
>> in WebRTC, for those of us wanting to gateway this stuff to SIP.  =
Imagine if it
>> turns out we were wrong, and some expected-or-unexpected =
vulnerability is
>> exploited against some large WebRTC service provider, and makes the
>> news/slashdot.  Browser vendors would instantly have new code =
removing
>> SDES support, and browsers in devices would be updated virtually =
overnight -
>> some users may take a long time to upgrade their specific browsers on =
their
>> specific devices - but many of them would do so within days.  That =
would be
>> untenable for a WebRTC service provider.=20
>=20
> I can't imagine such a scenario that couldn't also happen with =
DTLS-SRTP, especially DTLS-SRTP + EKT.

Yes, I considered that as well - but if there's only one key-exchange =
mechanism, and it's the one used between browsers and everyone, then =
even the browser vendors wouldn't want to update the browsers in such a =
way that it's removed overnight.  They'd be braking media interop even =
between browsers.

For example, assume we only have DTLS-SRTP as MTI and some security =
vulnerability is discovered for it, which some new thing called =
"ETLS-SRTP" fixes.  They couldn't just remove DTLS-SRTP in favor of =
ETLS-SRTP overnight; they'd have to support both for some transition =
period - a period only constrained by the rate at which most browsers =
get updated.  After that it could be disabled by default and require the =
user to enable, or pop up some warning thingy to approve; and ultimately =
removed altogether someday.

But with SDES, we're basically asking the browsers to implement =
something they don't need to implement for browser-to-browser uses. (in =
fact, we'd define it in such a way that SDES wouldn't be used for =
browser-to-browser cases)  Unless some major service like Skype uses =
SDES, browsers vendors have no real market force keeping them supporting =
it.  The ones involved in this WG have already said they don't consider =
the WebRTC-SIP use case very important in the grand scheme of things.


> Plus there's lots of counterexamples where major sites have been =
compromised in ways that could have been fixed browser-side, and yet =
browsers didn't bother to update.

Some browsers won't update, for sure - but that's the problem with these =
things: it's a lowest-common-denominator model, because having media =
work is not an optional feature.  If you want to support all (major) =
browsers, you need to do what all of them can do.  You'd have to support =
DTLS-SRTP somehow, if even one of the major browser vendors only do =
DTLS-SRTP.

Sure you can save some cost by tracking how many people use what =
browsers, and deploying for the ratio of DTLS-SRTP vs. SDES you expect, =
etc.  And that will work for some operators of WebRTC sites/gateways =
(though not all).  But it still detracts from the arguments I and others =
have been making about what the benefits of having SDES would be.


> And finally, what makes you think that the browser vendors would do =
something like that if the service really was popular? I guess if Google =
and Mozilla wanted everyone using Skype to switch to Internet Explorer =
they could pull such a stunt, but why? (In a hypothetical future where =
IE has webrtc support and Skype supports webrtc-equipped browsers as a =
platform for calling, of course)

If there's a really popular service using SDES, like let's say Skype, I =
would expect the browser vendors to keep SDES support optional with a =
configuration thingy; or to keep it by default but only until Skype =
switched away from SDES.[1]  And I bet Skype would switch away quickly, =
if a security vulnerability made the press/slashdot. =20

My point wasn't that it would literally happen in 24 hours - it was more =
that such a change is uncontrollable by all the operators of WebRTC =
sites/gateways.  Skype might be big enough to keep browsers from =
removing SDES, but that just moves the problem around: now all the other =
WebRTC sites/gateways have to change to using DTLS-SRTP when Skype =
changes.  Either way, they can't control their own destiny.



>> Or if the WG prefers, we could even write a separate doc on what a =
web-
>> based framework maker should consider supporting/not-supporting. (I =
can
>> imagine a few other things they might want to offer that a browser =
can't)
>=20
> They might be able to turn ICE off, even!

Yup.  That would be #2 on the list of things an app should be able to =
do, right after using SDES. (or maybe the priority should even be =
swapped)

-hadriel
[1] having said that, there have been cases where the market power of a =
browser overcame the market power of a really popular service - like =
Apple's iOS-based Safari's lack of Flash support, with respect to =
YouTube service.



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On Jul 18, 2013, at 5:19 PM, Peter Thatcher <pthatcher@google.com> =
wrote:

>=20
>=20
>=20
> On Thu, Jul 18, 2013 at 2:11 AM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> On Jul 16, 2013, at 2:43 AM, Peter Thatcher <pthatcher@google.com> =
wrote:
>=20
> > Hey Randell, can you give a short summary of the changes, if any, =
that I would or other implementors would need to know about?
> Hi Peter,
>=20
> using
> =
http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-jesup-=
rtcweb-data-protocol-04.txt&url2=3Dhttp://tools.ietf.org/id/draft-ietf-rtc=
web-data-protocol-00.txt
> you see that in addition to clarifications and cleanups
>=20
> Oh, now that's a slick tool.  Thanks for the link.
> =20
> * external negotiation of data channels is described
> * the packet format of the DATA_CHANNEL_OPEN message as been changed
>=20
> I just wrote the code to serialize and deserialize that message.  =
Please tell me we're not going to change it again :).
I think the format is now pretty good. I don't expect it to change.

Best regards
Michael
> =20
> Both of these changes require code changes.
>=20
> Best regards
> Michael
> >
> >
> > On Mon, Jul 15, 2013 at 1:31 PM, <internet-drafts@ietf.org> wrote:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> >  This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
> >
> >         Title           : WebRTC Data Channel Protocol
> >         Author(s)       : Randell Jesup
> >                           Salvatore Loreto
> >                           Michael Tuexen
> >         Filename        : draft-ietf-rtcweb-data-protocol-00.txt
> >         Pages           : 11
> >         Date            : 2013-07-15
> >
> > Abstract:
> >    The Web Real-Time Communication (WebRTC) working group is charged =
to
> >    provide protocols to support for direct interactive rich
> >    communication using audio, video, and data between two peers' =
web-
> >    browsers.  This document specifies an actual (minor) protocol for =
how
> >    the JS-layer DataChannel objects provide the data channels =
between
> >    the peers.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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From: Martin Thomson <martin.thomson@gmail.com>
To: Peter Thatcher <pthatcher@google.com>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 18 July 2013 08:12, Peter Thatcher <pthatcher@google.com> wrote:
> I believe I began paying attention to the mailing lists after you sent out
> theses slides that you didn't present.  I'm interested in seeing them, and
> while I could dig through archives to find them, if convenient, could you
> please give me a link to the slides?  Thanks.

It wasn't actually November, it was October, which made this harder to
find than I had expected.

http://lists.w3.org/Archives/Public/public-webrtc/2012Oct/0148.html

From randell-ietf@jesup.org  Thu Jul 18 14:16:48 2013
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Cc: Randell Jesup <randell-ietf@jesup.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-00.txt
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This is a multi-part message in MIME format.
--------------030000000104000300020200
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On 7/18/2013 11:19 AM, Peter Thatcher wrote:
>
>     using
>     http://tools.ietf.org/rfcdiff?url1=http://tools.ietf.org/id/draft-jesup-rtcweb-data-protocol-04.txt&url2=http://tools.ietf.org/id/draft-ietf-rtcweb-data-protocol-00.txt
>     you see that in addition to clarifications and cleanups
>
>
> Oh, now that's a slick tool.  Thanks for the link.
>
>     * external negotiation of data channels is described
>
>     * the packet format of the DATA_CHANNEL_OPEN message as been changed
>
>
> I just wrote the code to serialize and deserialize that message. 
>  Please tell me we're not going to change it again :).

No, that's the last time.  :-)  That was in response to the "why only 16 
bits?" questions I got.  We changed that internally quite a while ago.

-- 
Randell Jesup
randell-ietf@jesup.org


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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 7/18/2013 11:19 AM, Peter Thatcher
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrXDUG5sB63w3p6kDDuNT=_+Pq_m7ZmH+HdX_9c0sVXYBk2gw@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra">
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              using<br>
              <a moz-do-not-send="true"
href="http://tools.ietf.org/rfcdiff?url1=http://tools.ietf.org/id/draft-jesup-rtcweb-data-protocol-04.txt&amp;url2=http://tools.ietf.org/id/draft-ietf-rtcweb-data-protocol-00.txt"
                target="_blank">http://tools.ietf.org/rfcdiff?url1=http://tools.ietf.org/id/draft-jesup-rtcweb-data-protocol-04.txt&amp;url2=http://tools.ietf.org/id/draft-ietf-rtcweb-data-protocol-00.txt</a><br>
              you see that in addition to clarifications and cleanups<br>
            </blockquote>
            <div><br>
            </div>
            <div>Oh, now that's a slick tool. Â Thanks for the link.</div>
            <div>Â </div>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              * external negotiation of data channels is described<br>
            </blockquote>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">* the
              packet format of the DATA_CHANNEL_OPEN message as been
              changed<br>
            </blockquote>
            <div><br>
            </div>
            <div>I just wrote the code to serialize and deserialize that
              message. Â Please tell me we're not going to change it
              again :).</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    No, that's the last time.Â  :-)Â  That was in response to the "why
    only 16 bits?" questions I got.Â  We changed that internally quite a
    while ago.<br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup
<a class="moz-txt-link-abbreviated" href="mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a></pre>
  </body>
</html>

--------------030000000104000300020200--

From csp@csperkins.org  Thu Jul 18 14:19:15 2013
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Hi,

I submitted an update to the RTCWEB RTP usage draft. This version tries =
to address the comments that have been made to the list over the last =
couple of months. The changes in this version are as follows:

- In Section 4.1, expand discussion of multiple simultaneous SSRC values =
in a single RTP session, referencing draft-ietf-avtcore-rtp-multi-stream =
and draft-ietf-avtcore-rtp-multi-stream-optimisation. Add a note =
suggesting that draft-westerlund-mmusic-max-ssrc might be a useful =
addition.

- In Section 4.3, expand discussion of RTP payload type assignment, and =
briefly explain how the RTP payload type can be used to associate an RTP =
media stream with a signalling context (e.g., an SDP "m=3D" line).

- In Section 4.3, clarify that this memo does not specify mandatory to =
implement codecs or RTP payload formats. Also clarify that any codec =
where there is an RTP payload format and SDP offer/answer procedures =
defined can be used with WebRTC, provided it is negotiated.

- In Section 4.4, clarify that a single RTP session is to be used for =
all RTP media streams. Note that there is no consensus to use a =
shim-based approach.

- In Section 4.8, clarify that implementations MUST be prepared to =
accept RTP and RTCP packets using SSRCs that have not been explicitly =
signalled. Briefly explain how RTP media streams can be associated with =
a signalling context, using either a signalled SSRC or the RTP payload =
type.

- In Section 5.2.1, clarify that the rapid synchronisation extensions =
are in addition to RTCP SR-based synchronisation, and do not replace it.

- In Section 6.1, add a reference to RFC 3550 to explain how to do the =
RTT calculation from SR/RR packets.

- Remove the first paragraph of Section 7.2, since the issue is covered =
by the text in the last paragraph of Section 7.1.

- Move the remaining content from Section 7.2 to the end of Section 7.4, =
where it fits better.

- In what was Section 7.4, and is now Section 7.3, add some words about =
interoperability between sender- and receiver-driven congestion control. =
This issue is not solvable in this draft, since we don't yet have any =
standardised congestion control algorithms of either type, however it is =
important to note that it needs to be addressed when future congestion =
control algorithms are defined.

- Rewrite Section 8, on Performance Monitoring, to reflect the =
discussion at IETF 86.

- Add a note to Section 12.2 about the issue #20 in the RTCWEB issue =
tracker

- Editorial fixes to Sections 1, 2, 4.2, 4.5, 4.6, 4.7, 5.1, 6, 7, 10, =
and 11.

- Update list of open issues.

The draft contains a list of open issues in Section 15. Also, note that =
Sections 11, 12, and Appendix A have yet to be updated, due to running =
out of time before the submission deadline.

Comments are appreciated.=20

Colin



On 15 Jul 2013, at 22:57, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
> 	Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-07.txt
> 	Pages           : 61
> 	Date            : 2013-07-15
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-07
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-07
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Colin Perkins
http://csperkins.org/




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From: Martin Thomson <martin.thomson@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] data channel reset
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This statement, or something like it, is made in several places in the draft:

"Channels are closed by an SCTP reset of the Stream."

I assume that this uses the "Outgoing SSN Reset Request Parameter",
http://tools.ietf.org/html/rfc6525#section-4.1

Should it say that explicitly?

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Martin Thomson <martin.thomson@gmail.com>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Date: Thu, 18 Jul 2013 22:40:14 +0000
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Thanks for digging this up... in the meantime of course there's been work o=
ngoing in the MMUSIC IETF WG that adds even more questions than it answers.=
 See messages from as recent as today about msid, bundle, "unified plan", e=
tc. (All useful work that is needed for other types of SDP-using applicatio=
ns, like SIP-based videoconferencing systems, but work that will take some =
time and which as long as we stick with SDP as an API will be holding up th=
e WEBRTC specification)=0A=
=0A=
As long as it is allowed for the SDP to be modified between createOffer and=
 setLocalDescription, there must be a W3C specification instructing browser=
 vendors as to what changes are and are not allowed. That is *in addition* =
to knowing what SDP should be accepted (and what it means) when it comes fr=
om the other side. And the more "special" stuff ends up in SDP as a result =
of this WEBRTC-driven MMUSIC work, the longer it will be before such a spec=
ification can possibly be complete. (And note that it isn't sufficient to j=
ust wait for the MMUSIC work to be done and point to it... it will still be=
 necessary to *write down what the browser must do* in a the W3C specificat=
ion.)=0A=
=0A=
Or, as I've pointed out before, we could define a browser API that is entir=
ely independent of SDP, and allow enough control that JavaScript programmer=
s can arrange objects and set their parameters such that useful SRTP media =
streams can be sent and received, and then they can write whatever JavaScri=
pt they want to create the SDP that is getting defined over in MMUSIC and o=
ther places, or not use those specs if they so desire.=0A=
=0A=
Matthew Kaufman=0A=
=0A=
________________________________________=0A=
From: Martin Thomson [martin.thomson@gmail.com]=0A=
Sent: Thursday, July 18, 2013 9:53 AM=0A=
To: Peter Thatcher=0A=
Cc: Matthew Kaufman (SKYPE); <rtcweb@ietf.org>; public-webrtc@w3.org=0A=
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the cu=
rrent WebRTC API and SDP as a control surface=0A=
=0A=
On 18 July 2013 08:12, Peter Thatcher <pthatcher@google.com> wrote:=0A=
> I believe I began paying attention to the mailing lists after you sent ou=
t=0A=
> theses slides that you didn't present.  I'm interested in seeing them, an=
d=0A=
> while I could dig through archives to find them, if convenient, could you=
=0A=
> please give me a link to the slides?  Thanks.=0A=
=0A=
It wasn't actually November, it was October, which made this harder to=0A=
find than I had expected.=0A=
=0A=
http://lists.w3.org/Archives/Public/public-webrtc/2012Oct/0148.html=0A=
=0A=


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Martin Thomson <martin.thomson@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] data channel reset
Thread-Index: AQHOhAev2XlnXDUyeEypQSr01Z6yC5lrB5RR
Date: Thu, 18 Jul 2013 22:40:59 +0000
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Only if you think that RTCWEB should be using SCTP, I suppose.=0A=
=0A=
Matthew Kaufman=0A=
=0A=
________________________________________=0A=
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Martin=
 Thomson [martin.thomson@gmail.com]=0A=
Sent: Thursday, July 18, 2013 3:39 PM=0A=
To: rtcweb@ietf.org=0A=
Subject: [rtcweb] data channel reset=0A=
=0A=
This statement, or something like it, is made in several places in the draf=
t:=0A=
=0A=
"Channels are closed by an SCTP reset of the Stream."=0A=
=0A=
I assume that this uses the "Outgoing SSN Reset Request Parameter",=0A=
http://tools.ietf.org/html/rfc6525#section-4.1=0A=
=0A=
Should it say that explicitly?=0A=
_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb=0A=
=0A=


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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Fri, 19 Jul 2013 10:46:48 +1000
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To: Martin Thomson <martin.thomson@gmail.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Fri, Jul 19, 2013 at 2:53 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 18 July 2013 08:12, Peter Thatcher <pthatcher@google.com> wrote:
>> I believe I began paying attention to the mailing lists after you sent out
>> theses slides that you didn't present.  I'm interested in seeing them, and
>> while I could dig through archives to find them, if convenient, could you
>> please give me a link to the slides?  Thanks.
>
> It wasn't actually November, it was October, which made this harder to
> find than I had expected.
>
> http://lists.w3.org/Archives/Public/public-webrtc/2012Oct/0148.html


This captures exactly the kind of questions and concerns I had. Excellent work!

However, I don't fully agree with the conclusion of the slide deck.
I'd prefer we extended the constraints and other browser APIs that set
the SDP parameters rather than (or: in preference to) fully specifying
what SDP the browser has to support. I do like the ability to get the
low-level access to mangle the SDP as a means of experimenting with
new functionality or as a means to try and connect to devices that
don't have a WebRTC API. But I see the full definition of the SDP
parameters that browsers have to support as less important and
potentially just a by product of creating the higher level APIs.

What does it take for us to get focused on defining such an API that
is independent of SDP for the JS developer, and for now requires
browsers to do the mapping to SDP for them? Is the extension of the
constraints that a JS dev can manipulate enough for this?

Silvia.

From bernard_aboba@hotmail.com  Thu Jul 18 19:40:13 2013
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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: Adam Fineberg <fineberg@vline.me>
Date: Thu, 18 Jul 2013 19:40:07 -0700
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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--_e40724f7-758d-4c47-8861-18af1b356bd4_
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I think it may be possible to generalize this.  For example=2C for H.264/SV=
C which can support temporal=2C spatial and quality scalability=2C you woul=
d need the quality_id and dependency_id in addition to the temporal_id (wha=
t you call the temporal layer index).   =20

Date: Thu=2C 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me
To: bernard_aboba@hotmail.com
CC: rtcweb@ietf.org=3B avtext@ietf.org
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

=0A=
  =0A=
    =0A=
  =0A=
  =0A=
    Bernard=2C
=0A=
   =20
=0A=
    Good question.  I'm not familiar enough with the parameter=0A=
    requirements of all other scalable codecs to be able to generalize. =0A=
    If you'd like to help specify them=2C I'd be fine revising the draft=0A=
    to generalize.
=0A=
   =20
=0A=
    Regards=2C
=0A=
    Adam
=0A=
   =20
=0A=
    On 7/17/13 8:26 PM=2C Bernard Aboba=0A=
      wrote:
=0A=
    =0A=
    =0A=
      =0A=
      Since the need is not codec specific (e.g. it=0A=
        arises with any codec supporting temporal=2C spatial and quality=0A=
        scalability)=2C why=20
=0A=
         a VP8-specific RTP extension?=20
=0A=
        =20
=0A=
        =0A=
          Date: Wed=2C 17 Jul 2013 17:09:46 -0700
=0A=
          From: fineberg@vline.me
=0A=
          To: rtcweb@ietf.org
=0A=
          Subject: [rtcweb] Fwd: New Version Notification for=0A=
          draft-fineberg-avtext-temporal-layer-ext-00.txt
=0A=
         =20
=0A=
          Hi=2C=0A=
          =0A=
          I'm working on WebRTC=0A=
            services and have found that while developing services that=0A=
            forward VP8 video streams if we want to take advantage of=0A=
            the VP8 temporal scaling we must get the temporal layer=0A=
            information from the RTP header which requires us to decrypt=0A=
            the SRTP packets. This is undesirable both because the=0A=
            middle-box needs to have access to the keys as well as the=0A=
            because of the added overhead of the decrypt/encrypt cycle.=0A=
            This draft proposes an RTP header extension that will allow=0A=
            us to use the VP8 temporal layer information included in the=0A=
            header extension and therefore do forwarding without SRTP=0A=
            decryption. Comments welcome.=0A=
          =0A=
          Regards=2C=0A=
          Adam Fineberg=0A=
          fineberg=0A=
              at vline.com
=0A=
           =20
=0A=
            -------- Original Message --------=0A=
            =0A=
              =0A=
                =0A=
                  Subject:=0A=
                  New Version Notification for=0A=
                    draft-fineberg-avtext-temporal-layer-ext-00.txt=0A=
                =0A=
                =0A=
                  Date:=0A=
                  Tue=2C 09 Jul 2013 10:02:05 -0700=0A=
                =0A=
                =0A=
                  From:=0A=
                  internet-drafts at ietf.org=0A=
                =0A=
                =0A=
                  To:=0A=
                  Adam Fineberg <fineberg at vline.com>=0A=
                =0A=
              =0A=
            =0A=
           =20
=0A=
           =20
=0A=
            A new version of I-D=2C draft-fineberg-avtext-temporal-layer-ex=
t-00.txt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt=0A=
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext=0A=
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
=0A=
          =0A=
         =20
=0A=
          _______________________________________________=0A=
          rtcweb mailing list=0A=
          rtcweb@ietf.org=0A=
          https://www.ietf.org/mailman/listinfo/rtcweb=0A=
      =0A=
    =0A=
   =20
=0A=
    -- =0A=
Regards=2C=0A=
Adam 		 	   		  =

--_e40724f7-758d-4c47-8861-18af1b356bd4_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
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body.hmmessage
{
font-size: 12pt=3B
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>I think it may be possible to ge=
neralize this.&nbsp=3B For example=2C for H.264/SVC which can support tempo=
ral=2C spatial and quality scalability=2C you would need the quality_id and=
 dependency_id in addition to the temporal_id (what you call the temporal l=
ayer index). &nbsp=3B&nbsp=3B <br><br><div><hr id=3D"stopSpelling">Date: Th=
u=2C 18 Jul 2013 08:45:38 -0700<br>From: fineberg@vline.me<br>To: bernard_a=
boba@hotmail.com<br>CC: rtcweb@ietf.org=3B avtext@ietf.org<br>Subject: Re: =
[rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-l=
ayer-ext-00.txt<br><br>=0A=
  =0A=
    =0A=
  =0A=
  =0A=
    Bernard=2C<br>=0A=
    <br>=0A=
    Good question.&nbsp=3B I'm not familiar enough with the parameter=0A=
    requirements of all other scalable codecs to be able to generalize.&nbs=
p=3B=0A=
    If you'd like to help specify them=2C I'd be fine revising the draft=0A=
    to generalize.<br>=0A=
    <br>=0A=
    Regards=2C<br>=0A=
    Adam<br>=0A=
    <br>=0A=
    <div class=3D"ecxmoz-cite-prefix">On 7/17/13 8:26 PM=2C Bernard Aboba=
=0A=
      wrote:<br>=0A=
    </div>=0A=
    <blockquote cite=3D"mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl">=
=0A=
      <style><!--=0A=
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padding:0px=3B=0A=
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.ExternalClass body.ecxhmmessage {=0A=
font-size:12pt=3B=0A=
font-family:Calibri=3B=0A=
}=0A=
=0A=
--></style>=0A=
      <div dir=3D"ltr">Since the need is not codec specific (e.g. it=0A=
        arises with any codec supporting temporal=2C spatial and quality=0A=
        scalability)=2C why <br>=0A=
        &nbsp=3Ba VP8-specific RTP extension? <br>=0A=
        &nbsp=3B<br>=0A=
        <div>=0A=
          <hr id=3D"ecxstopSpelling">Date: Wed=2C 17 Jul 2013 17:09:46 -070=
0<br>=0A=
          From: <a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:fin=
eberg@vline.me">fineberg@vline.me</a><br>=0A=
          To: <a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:rtcwe=
b@ietf.org">rtcweb@ietf.org</a><br>=0A=
          Subject: [rtcweb] Fwd: New Version Notification for=0A=
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>=0A=
          <br>=0A=
          <span style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btex=
t-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bdisplay:inline !i=
mportant=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B"=
>Hi=2C</span><br style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Bte=
xt-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bwhite-space:norm=
al=3Bbackground-color:rgb(255=2C 255=2C 255)=3B">=0A=
          <br style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btext-=
indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bwhite-space:normal=
=3Bbackground-color:rgb(255=2C 255=2C 255)=3B">=0A=
          <span style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btex=
t-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bdisplay:inline !i=
mportant=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B"=
>I'm working on WebRTC=0A=
            services and have found that while developing services that=0A=
            forward VP8 video streams if we want to take advantage of=0A=
            the VP8 temporal scaling we must get the temporal layer=0A=
            information from the RTP header which requires us to decrypt=0A=
            the SRTP packets. This is undesirable both because the=0A=
            middle-box needs to have access to the keys as well as the=0A=
            because of the added overhead of the decrypt/encrypt cycle.=0A=
            This draft proposes an RTP header extension that will allow=0A=
            us to use the VP8 temporal layer information included in the=0A=
            header extension and therefore do forwarding without SRTP=0A=
            decryption. Comments welcome.</span><br style=3D"color:rgb(0=2C=
 0=2C 0)=3Btext-transform:none=3Btext-indent:0px=3Bletter-spacing:normal=3B=
word-spacing:0px=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C =
255)=3B">=0A=
          <br style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btext-=
indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bwhite-space:normal=
=3Bbackground-color:rgb(255=2C 255=2C 255)=3B">=0A=
          <span style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btex=
t-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bdisplay:inline !i=
mportant=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B"=
>Regards=2C</span><br style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=
=3Btext-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bwhite-space=
:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B">=0A=
          <span style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:none=3Btex=
t-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bdisplay:inline !i=
mportant=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B"=
>Adam Fineberg</span><br style=3D"color:rgb(0=2C 0=2C 0)=3Btext-transform:n=
one=3Btext-indent:0px=3Bletter-spacing:normal=3Bword-spacing:0px=3Bwhite-sp=
ace:normal=3Bbackground-color:rgb(255=2C 255=2C 255)=3B">=0A=
          <div class=3D"ecxmoz-forward-container" style=3D"color:rgb(0=2C 0=
=2C 0)=3Btext-transform:none=3Btext-indent:0px=3Bletter-spacing:normal=3Bwo=
rd-spacing:0px=3Bwhite-space:normal=3Bbackground-color:rgb(255=2C 255=2C 25=
5)=3B"><a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:fineberg%20a=
t%20vline.com" rel=3D"nofollow">fineberg=0A=
              at vline.com</a><br>=0A=
            <br>=0A=
            -------- Original Message --------=0A=
            <table class=3D"ecxmoz-email-headers-table" border=3D"0" cellpa=
dding=3D"0" cellspacing=3D"0">=0A=
              <tbody>=0A=
                <tr>=0A=
                  <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE"=
>Subject:</th>=0A=
                  <td>New Version Notification for=0A=
                    draft-fineberg-avtext-temporal-layer-ext-00.txt</td>=0A=
                </tr>=0A=
                <tr>=0A=
                  <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE"=
>Date:</th>=0A=
                  <td>Tue=2C 09 Jul 2013 10:02:05 -0700</td>=0A=
                </tr>=0A=
                <tr>=0A=
                  <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE"=
>From:</th>=0A=
                  <td><a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mail=
to:internet-drafts%20at%20ietf.org" rel=3D"nofollow">internet-drafts at iet=
f.org</a></td>=0A=
                </tr>=0A=
                <tr>=0A=
                  <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE"=
>To:</th>=0A=
                  <td>Adam Fineberg<span class=3D"ecxApple-converted-space"=
>&nbsp=3B</span><a class=3D"ecxmoz-txt-link-rfc2396E" href=3D"mailto:finebe=
rg%20at%20vline.com" rel=3D"nofollow">&lt=3Bfineberg at vline.com&gt=3B</a>=
</td>=0A=
                </tr>=0A=
              </tbody>=0A=
            </table>=0A=
            <br>=0A=
            <br>=0A=
            <pre style=3D"width:939.59px=3Bwhite-space:pre-wrap=3B-ms-word-=
wrap:break-word=3B">A new version of I-D=2C draft-fineberg-avtext-temporal-=
layer-ext-00.txt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://www.i=
etf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" ta=
rget=3D"_blank" rel=3D"nofollow">http://www.ietf.org/internet-drafts/draft-=
fineberg-avtext-temporal-layer-ext-00.txt</a>=0A=
Status:          <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://datat=
racker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target=3D"_bl=
ank" rel=3D"nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext=
-temporal-layer-ext</a>=0A=
Htmlized:        <a class=3D"ecxmoz-txt-link-freetext" href=3D"http://tools=
.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target=3D"_blan=
k" rel=3D"nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-tempor=
al-layer-ext-00</a>=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
</pre>=0A=
          </div>=0A=
          <br>=0A=
          _______________________________________________=0A=
          rtcweb mailing list=0A=
          <a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:rtcweb@ie=
tf.org">rtcweb@ietf.org</a>=0A=
          <a class=3D"ecxmoz-txt-link-freetext" href=3D"https://www.ietf.or=
g/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/l=
istinfo/rtcweb</a></div>=0A=
      </div>=0A=
    </blockquote>=0A=
    <br>=0A=
    <pre class=3D"ecxmoz-signature">-- =0A=
Regards=2C=0A=
Adam</pre></div> 		 	   		  </div></body>
</html>=

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From: Justin Uberti <juberti@google.com>
Date: Thu, 18 Jul 2013 22:40:20 -0400
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Well, for the past 6+ months we haven't been able to agree on exactly how a
MediaStreamTrack should be represented in SDP, so it would have been
premature to try to write this stuff down.

That said, I agree this stuff needs to be fully defined, so that
applications can get predictable behavior. I also agree it will be a large
undertaking.


On Tue, Jul 16, 2013 at 8:30 PM, Bernard Aboba <bernard_aboba@hotmail.com>w=
rote:

> I would agree. Just pointing out that this issue was raised in December
> 2012, and it is now 6+ months later with little progress and more
> disgruntlement. So if there is motivation to fix this it probably would
> have manifested itself by now.
>
> On Jul 16, 2013, at 5:27 PM, "Silvia Pfeiffer" <silviapfeiffer1@gmail.com=
>
> wrote:
>
> > The whole point of standardisation at the W3C is for browser to agree
> > on the knobs. So, I do think it's a valuable effort.
> > Cheers,
> > Silvia.
> >
> >
> > On Wed, Jul 17, 2013 at 10:15 AM, Bernard Aboba
> > <bernard_aboba@hotmail.com> wrote:
> >> The problem was that we never defined what mangling browsers had to
> support. Given all the SDP specs that is actually a huge work item. And
> creating APIs to avoid mangling isn't actually that much simpler. What
> knobs do you provide and which ones do you ignore? Browsers might not agr=
ee
> on those either.
> >>
> >> On Jul 8, 2013, at 1:37 PM, "Stefan H=C3=A5kansson LK" <
> stefan.lk.hakansson@ericsson.com> wrote:
> >>
> >>> On 7/8/13 5:22 PM, Roman Shpount wrote:
> >>>
> >>>> So, was there ever a consensus that "developers MUST NOT
> >>>> touch SDP"?
> >>>
> >>> No, developers are free to touch the SDP (and probably must in certai=
n
> >>> cases).
> >>>
> >>>>
> >>>> _____________
> >>>> Roman Shpount
> >>>
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Well, for the past 6+ months we haven&#39;t been able to a=
gree on exactly how a MediaStreamTrack should be represented in SDP, so it =
would have been premature to try to write this stuff down.<div><br></div><d=
iv>

That said, I agree this stuff needs to be fully defined, so that applicatio=
ns can get predictable behavior. I also agree it will be a large undertakin=
g.<div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue=
, Jul 16, 2013 at 8:30 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_aboba@hotmail.co=
m</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I would agree. Just pointing out that this i=
ssue was raised in December 2012, and it is now 6+ months later with little=
 progress and more disgruntlement. So if there is motivation to fix this it=
 probably would have manifested itself by now.<br>


<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Jul 16, 2013, at 5:27 PM, &quot;Silvia Pfeiffer&quot; &lt;<a href=3D"mai=
lto:silviapfeiffer1@gmail.com">silviapfeiffer1@gmail.com</a>&gt; wrote:<br>
<br>
&gt; The whole point of standardisation at the W3C is for browser to agree<=
br>
&gt; on the knobs. So, I do think it&#39;s a valuable effort.<br>
&gt; Cheers,<br>
&gt; Silvia.<br>
&gt;<br>
&gt;<br>
&gt; On Wed, Jul 17, 2013 at 10:15 AM, Bernard Aboba<br>
&gt; &lt;<a href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail=
.com</a>&gt; wrote:<br>
&gt;&gt; The problem was that we never defined what mangling browsers had t=
o support. Given all the SDP specs that is actually a huge work item. And c=
reating APIs to avoid mangling isn&#39;t actually that much simpler. What k=
nobs do you provide and which ones do you ignore? Browsers might not agree =
on those either.<br>


&gt;&gt;<br>
&gt;&gt; On Jul 8, 2013, at 1:37 PM, &quot;Stefan H=C3=A5kansson LK&quot; &=
lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@=
ericsson.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; On 7/8/13 5:22 PM, Roman Shpount wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; So, was there ever a consensus that &quot;developers MUST =
NOT<br>
&gt;&gt;&gt;&gt; touch SDP&quot;?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; No, developers are free to touch the SDP (and probably must in=
 certain<br>
&gt;&gt;&gt; cases).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; _____________<br>
&gt;&gt;&gt;&gt; Roman Shpount<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" targe=
t=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div></div></div>

--e89a8f2353a7a98cad04e1d440ea--

From fineberg@vline.me  Thu Jul 18 20:34:21 2013
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Date: Thu, 18 Jul 2013 20:34:16 -0700
From: Adam Fineberg <fineberg@vline.me>
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Cc: "avtext@ietf.org" <avtext@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Bernard,

Are there other codecs you are thinking should be supported?  If it's 
generalized I would think we want to be able to cover all known scalable 
codecs. I'll look into the H264/SVC fields to see how to encode them in 
a generalized header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
> I think it may be possible to generalize this.  For example, for 
> H.264/SVC which can support temporal, spatial and quality scalability, 
> you would need the quality_id and dependency_id in addition to the 
> temporal_id (what you call the temporal layer index).
>
> ------------------------------------------------------------------------
> Date: Thu, 18 Jul 2013 08:45:38 -0700
> From: fineberg@vline.me
> To: bernard_aboba@hotmail.com
> CC: rtcweb@ietf.org; avtext@ietf.org
> Subject: Re: [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Bernard,
>
> Good question.  I'm not familiar enough with the parameter 
> requirements of all other scalable codecs to be able to generalize.  
> If you'd like to help specify them, I'd be fine revising the draft to 
> generalize.
>
> Regards,
> Adam
>
> On 7/17/13 8:26 PM, Bernard Aboba wrote:
>
>     Since the need is not codec specific (e.g. it arises with any
>     codec supporting temporal, spatial and quality scalability), why
>      a VP8-specific RTP extension?
>
>     ------------------------------------------------------------------------
>     Date: Wed, 17 Jul 2013 17:09:46 -0700
>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>     To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     Subject: [rtcweb] Fwd: New Version Notification for
>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>     Hi,
>
>     I'm working on WebRTC services and have found that while
>     developing services that forward VP8 video streams if we want to
>     take advantage of the VP8 temporal scaling we must get the
>     temporal layer information from the RTP header which requires us
>     to decrypt the SRTP packets. This is undesirable both because the
>     middle-box needs to have access to the keys as well as the because
>     of the added overhead of the decrypt/encrypt cycle. This draft
>     proposes an RTP header extension that will allow us to use the VP8
>     temporal layer information included in the header extension and
>     therefore do forwarding without SRTP decryption. Comments welcome.
>
>     Regards,
>     Adam Fineberg
>     fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>
>     -------- Original Message --------
>     Subject: 	New Version Notification for
>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>     Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>     From: 	internet-drafts at ietf.org
>     <mailto:internet-drafts%20at%20ietf.org>
>     To: 	Adam Fineberg<fineberg at vline.com>
>     <mailto:fineberg%20at%20vline.com>
>
>
>
>     A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>     has been successfully submitted by Adam Fineberg and posted to the
>     IETF repository.
>
>     Filename:	 draft-fineberg-avtext-temporal-layer-ext
>     Revision:	 00
>     Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>     Creation date:	 2013-07-08
>     Group:		 Individual Submission
>     Number of pages: 6
>     URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>     Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>     Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>
>     Abstract:
>         This document defines a mechanism by which packets of Real-Time
>         Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>         indicate, in an RTP header extension, the temporal layer information
>         about the frame encoded in the RTP packet.  This information can be
>         used in a middlebox performing bandwidth management of streams
>         without requiring it to decrypt the streams.
>
>
>     _______________________________________________ rtcweb mailing
>     list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> -- 
> Regards,
> Adam


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    <div class="moz-cite-prefix">Bernard,<br>
      <br>
      Are there other codecs you are thinking should be supported?&nbsp; If
      it's generalized I would think we want to be able to cover all
      known scalable codecs. I'll look into the H264/SVC fields to see
      how to encode them in a generalized header.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W107EA5AE4A7E9D3DB6AFD0893630@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">I think it may be possible to generalize this.&nbsp; For
        example, for H.264/SVC which can support temporal, spatial and
        quality scalability, you would need the quality_id and
        dependency_id in addition to the temporal_id (what you call the
        temporal layer index). &nbsp;&nbsp; <br>
        <br>
        <div>
          <hr id="stopSpelling">Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
          From: <a class="moz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
          To: <a class="moz-txt-link-abbreviated" href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a><br>
          CC: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><br>
          Subject: Re: [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          Bernard,<br>
          <br>
          Good question.&nbsp; I'm not familiar enough with the parameter
          requirements of all other scalable codecs to be able to
          generalize.&nbsp; If you'd like to help specify them, I'd be fine
          revising the draft to generalize.<br>
          <br>
          Regards,<br>
          Adam<br>
          <br>
          <div class="ecxmoz-cite-prefix">On 7/17/13 8:26 PM, Bernard
            Aboba wrote:<br>
          </div>
          <blockquote
            cite="mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl">
            <style><!--
.ExternalClass .ecxhmmessage P {
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--></style>
            <div dir="ltr">Since the need is not codec specific (e.g. it
              arises with any codec supporting temporal, spatial and
              quality scalability), why <br>
              &nbsp;a VP8-specific RTP extension? <br>
              &nbsp;<br>
              <div>
                <hr id="ecxstopSpelling">Date: Wed, 17 Jul 2013 17:09:46
                -0700<br>
                From: <a moz-do-not-send="true"
                  class="ecxmoz-txt-link-abbreviated"
                  href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                To: <a moz-do-not-send="true"
                  class="ecxmoz-txt-link-abbreviated"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                Subject: [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                <br>
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Hi,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">I'm working on WebRTC services and have
                  found that while developing services that forward VP8
                  video streams if we want to take advantage of the VP8
                  temporal scaling we must get the temporal layer
                  information from the RTP header which requires us to
                  decrypt the SRTP packets. This is undesirable both
                  because the middle-box needs to have access to the
                  keys as well as the because of the added overhead of
                  the decrypt/encrypt cycle. This draft proposes an RTP
                  header extension that will allow us to use the VP8
                  temporal layer information included in the header
                  extension and therefore do forwarding without SRTP
                  decryption. Comments welcome.</span><br
                  style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Regards,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Adam Fineberg</span><br
                  style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <div class="ecxmoz-forward-container"
                  style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);"><a moz-do-not-send="true"
                    class="ecxmoz-txt-link-abbreviated"
                    href="mailto:fineberg%20at%20vline.com"
                    rel="nofollow">fineberg at vline.com</a><br>
                  <br>
                  -------- Original Message --------
                  <table class="ecxmoz-email-headers-table" border="0"
                    cellpadding="0" cellspacing="0">
                    <tbody>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap"
                          valign="BASELINE">Subject:</th>
                        <td>New Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap"
                          valign="BASELINE">Date:</th>
                        <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap"
                          valign="BASELINE">From:</th>
                        <td><a moz-do-not-send="true"
                            class="ecxmoz-txt-link-abbreviated"
                            href="mailto:internet-drafts%20at%20ietf.org"
                            rel="nofollow">internet-drafts at ietf.org</a></td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap"
                          valign="BASELINE">To:</th>
                        <td>Adam Fineberg<span
                            class="ecxApple-converted-space">&nbsp;</span><a
                            moz-do-not-send="true"
                            class="ecxmoz-txt-link-rfc2396E"
                            href="mailto:fineberg%20at%20vline.com"
                            rel="nofollow">&lt;fineberg at vline.com&gt;</a></td>
                      </tr>
                    </tbody>
                  </table>
                  <br>
                  <br>
                  <pre style="width:939.59px;white-space:pre-wrap;-ms-word-wrap:break-word;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank" rel="nofollow">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank" rel="nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank" rel="nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                </div>
                <br>
                _______________________________________________ rtcweb
                mailing list <a moz-do-not-send="true"
                  class="ecxmoz-txt-link-abbreviated"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a
                  moz-do-not-send="true"
                  class="ecxmoz-txt-link-freetext"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
            </div>
          </blockquote>
          <br>
          <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------050600000506060503030109--

From denglingli@chinamobile.com  Thu Jul 18 22:34:04 2013
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From: =?UTF-8?B?6YKT54G16I6JL0xpbmdsaSBEZW5n?= <denglingli@chinamobile.com>
To: "'Paul Coverdale'" <coverdale@sympatico.ca>, "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Bogineni, Kalyani'" <Kalyani.Bogineni@VerizonWireless.com>, "'Bo Burman'" <bo.burman@ericsson.com>, <rtcweb@ietf.org>
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We also support this proposal.

In particular, I think it makes sense to allow the codec choice be =
possibly delegated to individual web developers making use of WebRTC =
functionality.
Therefore, it would be better for them to be informed of available =
options and make the decision accordingly.
While it is not realistic to assume that a browser could always =
implement a best choice for every potential setting, to include other =
available codec in the offer would open the door to a win-win outcome.

BR
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 Paul Coverdale
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2013=E5=B9=B47=E6=9C=8818=E6=97=A5 =
20:25
=E6=94=B6=E4=BB=B6=E4=BA=BA: 'Hutton, Andrew'; 'Bogineni, Kalyani'; 'Bo =
Burman'; rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: Re: [rtcweb] Some thoughts on optional audio codecs

Yes, this text has been around for a while. I also support it.

...Paul

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>Of Hutton, Andrew
>Sent: Wednesday, July 17, 2013 4:46 AM
>To: Bogineni, Kalyani; 'Bo Burman'; rtcweb@ietf.org
>Subject: Re: [rtcweb] Some thoughts on optional audio codecs
>
>We appear to have been around this loop a number of times the text
>suggested here is exactly what was suggested by Andrew Allen back in
>January and I for one supported it them and still do - See
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg06121.html.
>
>Not sure there was a definitive conclusion to that particular consensus
>call.
>
>Andy
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Bogineni, Kalyani
>> Sent: 16 July 2013 18:02
>> To: 'Bo Burman'; rtcweb@ietf.org
>> Cc: Bogineni, Kalyani
>> Subject: Re: [rtcweb] Some thoughts on optional audio codecs
>>
>> We support the following wording proposal from Bo Burman.
>>
>> "If other suitable audio codecs are available to the browser to use,
>> it is recommended that they are also included in the offer in order =
to
>> maximize the possibility to establish the session without the need =
for
>> audio transcoding".
>>
>> Regards,
>> Kalyani Bogineni
>> Verizon
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Bo Burman
>> Sent: Monday, July 15, 2013 11:15 AM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Some thoughts on optional audio codecs
>>
>> Regarding the previous discussion on optional audio codecs in the
>> (currently expired) draft on RTCWEB audio codecs
>> (https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/):
>>
>> I think most parties involved in WebRTC work, myself included, hope
>> and believe that it will be ubiquitous and easy to include real-time
>> media conversation functionality in nearly any web context. Since it
>> will be that easy, it can be expected that most web developers need
>> not be, and thus will not be, media specialists or very knowledgeable
>about codecs.
>>
>> The definition of RTCWEB MTI codecs ensures that communication is
>> possible since at least one codec will always be found, but it is not
>> possible to claim the resulting communication to be optimum for every
>> possible context.
>>
>> Even if WebRTC will be close to ubiquitous, there will for quite some
>> time likely be a desire to reach real-time media domains and devices
>> that were not originally designed for and thus are not optimized for
>> use with WebRTC. A communication device that is not designed solely
>> for WebRTC use will likely include functionality and codecs also for
>> its "native" domain.
>>
>> Any added cost of not being able to use existing "native" codecs will
>> vary both in amount and where the cost has to be taken. Eliminating =
it
>> is indeed an optimization, but the total cost savings may still be
>> significant.
>>
>> With the current design and to my understanding, it will be the
>> browser vendor's choice to add optional codecs, including any =
"native"
>> domain codecs.  The choice may possibly be delegated to individual =
web
>> developers making use of WebRTC functionality. A browser vendor will
>> arguably have to know each target platform to some extent, but it can
>> hardly be assumed that a web developer knows the capabilities of all
>> devices that will use the WebRTC-enabled site unless the browser can
>> provide the needed information. There is a risk that "native" codecs
>> in devices are not well handled, unless the motivations and methods =
to
>> make use of them are better specified.
>>
>> While any audio codecs besides the MTI ones are clearly optional, I
>> believe the suggested text addition on optional audio codecs to the
>> RTCWEB audio draft in Ticket #12
>> (http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/12#) to be too =
brief
>> considering the above.
>>
>> In that draft, I would prefer something more in line with:
>>
>> "If other suitable audio codecs are available to the browser to use,
>> it is recommended that they are also included in the offer in order =
to
>> maximize the possibility to establish the session without the need =
for
>> audio transcoding".
>>
>> Assuming that the browser vendor (or web developer) is sufficiently
>> concerned with codecs to read the audio codecs draft (or the
>> corresponding RFC to-be), the above text may, as a start, give some
>> added guidance why non-MTI codecs may be desirable to consider in
>> addition to the MTI ones.
>>
>> Cheers,
>> Bo
>>
>> Multimedia Technologies
>> Ericsson Research
>> F=C3=A4r=C3=B6gatan 6
>> SE-164 80, Kista, Sweden
>> www.ericsson.com
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb




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If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me> wrote:

> Bernard,
>=20
> Are there other codecs you are thinking should be supported?  If it's gene=
ralized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generaliz=
ed header.
>=20
> Regards,
> Adam
>=20
> On 7/18/13 7:40 PM, Bernard Aboba wrote:
>> I think it may be possible to generalize this.  For example, for H.264/SV=
C which can support temporal, spatial and quality scalability, you would nee=
d the quality_id and dependency_id in addition to the temporal_id (what you c=
all the temporal layer index).   =20
>>=20
>> Date: Thu, 18 Jul 2013 08:45:38 -0700
>> From: fineberg@vline.me
>> To: bernard_aboba@hotmail.com
>> CC: rtcweb@ietf.org; avtext@ietf.org
>> Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-av=
text-temporal-layer-ext-00.txt
>>=20
>> Bernard,
>>=20
>> Good question.  I'm not familiar enough with the parameter requirements o=
f all other scalable codecs to be able to generalize.  If you'd like to help=
 specify them, I'd be fine revising the draft to generalize.
>>=20
>> Regards,
>> Adam
>>=20
>> On 7/17/13 8:26 PM, Bernard Aboba wrote:
>> Since the need is not codec specific (e.g. it arises with any codec suppo=
rting temporal, spatial and quality scalability), why=20
>>  a VP8-specific RTP extension?=20
>> =20
>> Date: Wed, 17 Jul 2013 17:09:46 -0700
>> From: fineberg@vline.me
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext=
-temporal-layer-ext-00.txt
>>=20
>> Hi,
>>=20
>> I'm working on WebRTC services and have found that while developing servi=
ces that forward VP8 video streams if we want to take advantage of the VP8 t=
emporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both b=
ecause the middle-box needs to have access to the keys as well as the becaus=
e of the added overhead of the decrypt/encrypt cycle. This draft proposes an=
 RTP header extension that will allow us to use the VP8 temporal layer infor=
mation included in the header extension and therefore do forwarding without S=
RTP decryption. Comments welcome.
>>=20
>> Regards,
>> Adam Fineberg
>> fineberg at vline.com
>>=20
>> -------- Original Message --------
>> Subject:	New Version Notification for draft-fineberg-avtext-temporal=
-layer-ext-00.txt
>> Date:	Tue, 09 Jul 2013 10:02:05 -0700
>> From:	internet-drafts at ietf.org
>> To:	Adam Fineberg <fineberg at vline.com>
>>=20
>> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>> has been successfully submitted by Adam Fineberg and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-fineberg-avtext-temporal-layer-ext
>> Revision:	 00
>> Title:		 A Real-Time Transport Protocol (RTP) Header Extens=
ion for VP8 Temporal Layer Information
>> Creation date:	 2013-07-08
>> Group:		 Individual Submission
>> Number of pages: 6
>> URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtex=
t-temporal-layer-ext-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-te=
mporal-layer-ext
>> Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-tempora=
l-layer-ext-00
>>=20
>>=20
>> Abstract:
>>    This document defines a mechanism by which packets of Real-Time
>>    Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>    indicate, in an RTP header extension, the temporal layer information
>>    about the frame encoded in the RTP packet.  This information can be
>>    used in a middlebox performing bandwidth management of streams
>>    without requiring it to decrypt the streams.
>>=20
>> _______________________________________________ rtcweb mailing list rtcwe=
b@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>> --=20
>> Regards,
>> Adam
>=20

--Apple-Mail-C7668BC4-5989-40CD-888A-D7A5292B2C95
Content-Type: text/html; charset="utf-8"
Content-Transfer-Encoding: 7bit

<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>If we can support VP8/9 as well as H.264/5 SVC</div><div>that would be a start. It seems doable to me.</div><div><br>On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" &lt;<a href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt; wrote:<br><br></div><blockquote type="cite"><div>
  
    <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
  
  
    <div class="moz-cite-prefix">Bernard,<br>
      <br>
      Are there other codecs you are thinking should be supported?&nbsp; If
      it's generalized I would think we want to be able to cover all
      known scalable codecs. I'll look into the H264/SVC fields to see
      how to encode them in a generalized header.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W107EA5AE4A7E9D3DB6AFD0893630@phx.gbl" type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style>
      <div dir="ltr">I think it may be possible to generalize this.&nbsp; For
        example, for H.264/SVC which can support temporal, spatial and
        quality scalability, you would need the quality_id and
        dependency_id in addition to the temporal_id (what you call the
        temporal layer index). &nbsp;&nbsp; <br>
        <br>
        <div>
          <hr id="stopSpelling">Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
          From: <a class="moz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
          To: <a class="moz-txt-link-abbreviated" href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a><br>
          CC: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><br>
          Subject: Re: [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          Bernard,<br>
          <br>
          Good question.&nbsp; I'm not familiar enough with the parameter
          requirements of all other scalable codecs to be able to
          generalize.&nbsp; If you'd like to help specify them, I'd be fine
          revising the draft to generalize.<br>
          <br>
          Regards,<br>
          Adam<br>
          <br>
          <div class="ecxmoz-cite-prefix">On 7/17/13 8:26 PM, Bernard
            Aboba wrote:<br>
          </div>
          <blockquote cite="mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl">
            <style><!--
.ExternalClass .ecxhmmessage P {
padding:0px;
}

.ExternalClass body.ecxhmmessage {
font-size:12pt;
font-family:Calibri;
}

--></style>
            <div dir="ltr">Since the need is not codec specific (e.g. it
              arises with any codec supporting temporal, spatial and
              quality scalability), why <br>
              &nbsp;a VP8-specific RTP extension? <br>
              &nbsp;<br>
              <div>
                <hr id="ecxstopSpelling">Date: Wed, 17 Jul 2013 17:09:46
                -0700<br>
                From: <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                To: <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                Subject: [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                <br>
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Hi,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">I'm working on WebRTC services and have
                  found that while developing services that forward VP8
                  video streams if we want to take advantage of the VP8
                  temporal scaling we must get the temporal layer
                  information from the RTP header which requires us to
                  decrypt the SRTP packets. This is undesirable both
                  because the middle-box needs to have access to the
                  keys as well as the because of the added overhead of
                  the decrypt/encrypt cycle. This draft proposes an RTP
                  header extension that will allow us to use the VP8
                  temporal layer information included in the header
                  extension and therefore do forwarding without SRTP
                  decryption. Comments welcome.</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Regards,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Adam Fineberg</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <div class="ecxmoz-forward-container" style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);"><a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:fineberg%20at%20vline.com" rel="nofollow">fineberg at vline.com</a><br>
                  <br>
                  -------- Original Message --------
                  <table class="ecxmoz-email-headers-table" border="0" cellpadding="0" cellspacing="0">
                    <tbody>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
                        <td>New Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
                        <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
                        <td><a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow">internet-drafts at ietf.org</a></td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
                        <td>Adam Fineberg<span class="ecxApple-converted-space">&nbsp;</span><a moz-do-not-send="true" class="ecxmoz-txt-link-rfc2396E" href="mailto:fineberg%20at%20vline.com" rel="nofollow">&lt;fineberg at vline.com&gt;</a></td>
                      </tr>
                    </tbody>
                  </table>
                  <br>
                  <br>
                  <pre style="width:939.59px;white-space:pre-wrap;-ms-word-wrap:break-word;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank" rel="nofollow">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank" rel="nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank" rel="nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                </div>
                <br>
                _______________________________________________ rtcweb
                mailing list <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
            </div>
          </blockquote>
          <br>
          <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
        </div>
      </div>
    </blockquote>
    <br>
  

</div></blockquote></body></html>
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Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On 7/19/13 12:40 AM, Matthew Kaufman (SKYPE) wrote:=0A=
> Thanks for digging this up... in the meantime of course there's been=0A=
> work ongoing in the MMUSIC IETF WG that adds even more questions than=0A=
> it answers. See messages from as recent as today about msid, bundle,=0A=
> "unified plan", etc. (All useful work that is needed for other types=0A=
> of SDP-using applications, like SIP-based videoconferencing systems,=0A=
> but work that will take some time and which as long as we stick with=0A=
> SDP as an API will be holding up the WEBRTC specification)=0A=
>=0A=
> As long as it is allowed for the SDP to be modified between=0A=
> createOffer and setLocalDescription, there must be a W3C=0A=
> specification instructing browser vendors as to what changes are and=0A=
> are not allowed.=0A=
=0A=
Perhaps we should, at least initially, consider Martin's proposal: =0A=
modifications are not allowed between createX and setLocal.=0A=
=0A=
>=0A=
> Or, as I've pointed out before, we could define a browser API that is=0A=
> entirely independent of SDP, and allow enough control that JavaScript=0A=
> programmers can arrange objects and set their parameters such that=0A=
> useful SRTP media streams can be sent and received, and then they can=0A=
> write whatever JavaScript they want to create the SDP that is getting=0A=
> defined over in MMUSIC and other places, or not use those specs if=0A=
> they so desire.=0A=
>=0A=
> Matthew Kaufman=0A=
>=0A=
> ________________________________________ From: Martin Thomson=0A=
> [martin.thomson@gmail.com] Sent: Thursday, July 18, 2013 9:53 AM To:=0A=
> Peter Thatcher Cc: Matthew Kaufman (SKYPE); <rtcweb@ietf.org>;=0A=
> public-webrtc@w3.org Subject: Re: [rtcweb] Summary of Application=0A=
> Developers' opinions of the current WebRTC API and SDP as a control=0A=
> surface=0A=
>=0A=
> On 18 July 2013 08:12, Peter Thatcher <pthatcher@google.com> wrote:=0A=
>> I believe I began paying attention to the mailing lists after you=0A=
>> sent out theses slides that you didn't present.  I'm interested in=0A=
>> seeing them, and while I could dig through archives to find them,=0A=
>> if convenient, could you please give me a link to the slides?=0A=
>> Thanks.=0A=
>=0A=
> It wasn't actually November, it was October, which made this harder=0A=
> to find than I had expected.=0A=
>=0A=
> http://lists.w3.org/Archives/Public/public-webrtc/2012Oct/0148.html=0A=
>=0A=
>=0A=
> _______________________________________________ rtcweb mailing list=0A=
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
Thread-Topic: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-freshness-04.txt
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Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action:	draft-muthu-behave-consent-freshness-04.txt
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Hi Muthu
So if I understand correctly, this functionality is on top of ICE functiona=
lity and is optional for the gateways to support it. Right?

Regards
Ranjit




-----Original Message-----
From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Wednesday, July 17, 2013 7:48 PM
To: Avasarala, Ranjit (NSN - IN/Bangalore)
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-f=
reshness-04.txt

|-----Original Message-----
|From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn.=
com]
|Sent: Wednesday, July 17, 2013 2:36 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
|My comments inline <Ranjit>
|
|Regards
|Ranjit
|
|-----Original Message-----
|From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
|Sent: Wednesday, July 17, 2013 2:16 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent-=
freshness-04.txt
|
||If many public STUN servers and call servers that implement STUN or ICEli=
te
||functionalities, are not expected to support
|
|I should have said they have no role to play. It is similar to peer-to-pee=
r ICE connectivity checks --
|those checks don't go through public STUN or call servers at all.
|[Ranjit] But there can be scenarios where RTCWebBrowsers do talk to WebRTC=
 Gateways which could be co-
|located with call handling servers. Then it won't make sense to have this =
feature.

An RTCWeb gateway wouldn't have to do anything fancier than what it already=
 does for ICE connectivity checks.=20

|
|The difference between ICE connectivity checks and consent freshness check=
s is that the former is
|performed before session establishment and the later is performed after se=
ssion establishment.
|[Ranjit] Why do you need to check consent of the peer after session is est=
ablished? Ideally I should
|check consent before session establishment. If the peer does not want to r=
eceive traffic, then session
|itself should not be established.

See draft-ietf-rtcweb-security-arch:

<snip>
4.4.  Communications and Consent Freshness

   From a security perspective, everything from here on in is a little
   anticlimactic:  Alice and Bob exchange data protected by the keys
   negotiated by DTLS.  Because of the security guarantees discussed in
   the previous sections, they know that the communications are
   encrypted and authenticated.

   The one remaining security property we need to establish is "consent
   freshness", i.e., allowing Alice to verify that Bob is still prepared
   to receive her communications so that Alice does not continue to send
   large traffic volumes to entities which went abruptly offline.  ICE
   specifies periodic STUN keepalizes but only if media is not flowing.
   Because the consent issue is more difficult here, we require RTCWeb
   implementations to periodically send keepalives.  As described in
   Section 5.3, these keepalives MUST be based on the consent freshness
   mechanism specified in [I-D.muthu-behave-consent-freshness].  If a
   keepalive fails and no new ICE channels can be established, then the
   session is terminated.
</snip>

|
||then who would have interest to support this functionality?
|
|RTCWEB browsers and browser-like platforms.
|[Ranjit] browser may implement, but it would be very customized one. I do =
not think the intermediate
|entities would want to support.

You seem to be completely missing the very purpose of consent.

|
||Ideally getting consent for receiving traffic or call should be part of s=
ignaling.
||I do not think using STUN is appropriate.
|
|ICE already does that. You think it is inappropriate?
|[Ranjit] I am saying that using STUN for consent check is not appropriate =
as STUN has other dedicated
|functionality to do - like connectivity checks, keep alives, etc.=20

See draft-ietf-rtcweb-security and draft-ietf-rtcweb-security.

|Also in cases of Symmetric NATs where STUN fails, TURN is needed. Then the=
 proposed feature would anyway fail.

The solution is built on top of ICE, so should work across these.

Muthu

|
|Muthu
|
||-----Original Message-----
||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@nsn=
.com]
||Sent: Wednesday, July 17, 2013 12:16 PM
||To: Muthu Arul Mozhi Perumal (mperumal)
||Cc: rtcweb@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Hi Muthu
||
||If many public STUN servers and call servers that implement STUN or ICEli=
te functionalities, are not
||expected to support, then who would have interest to support this functio=
nality? Ideally getting
||consent for receiving traffic or call should be part of signaling. I do n=
ot think using STUN is
||appropriate.
||
||Regards
||Ranjit
||
||
||
||
||-----Original Message-----
||From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
||Sent: Wednesday, July 17, 2013 12:07 PM
||To: Avasarala, Ranjit (NSN - IN/Bangalore)
||Cc: rtcweb@ietf.org; behave@ietf.org
||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consent=
-freshness-04.txt
||
||Ranjit,
||
|||Many publicly available STUN servers or those that are part of Call serv=
ers
|||may not support this.
||
||Public STUN servers and call servers are not expected to support this.
||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough
|||signaling?
||
||No. That signaling between the JS and the web server could be anything (S=
IP or XMPP or proprietary or
||whatever) and the browser has no control over it.
||
||Muthu
||
|||-----Original Message-----
|||From: Avasarala, Ranjit (NSN - IN/Bangalore) [mailto:ranjit.avasarala@ns=
n.com]
|||Sent: Wednesday, July 17, 2013 11:19 AM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: RE: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Hi Muthu
|||
|||Though using STUN request/response may be good for querying about consen=
t, I feel it is overloading
|||the functionality of STUN. Many publicly available STUN servers or those=
 that are part of Call
||servers
|||may not support this.
|||
|||Can't there be some other neutral mechanism to query peer's consent - th=
rough signaling?
|||
|||
|||Regards
|||Ranjit
|||
|||
|||
|||-----Original Message-----
|||From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf=
 Of ext Simon Perreault
|||Sent: Tuesday, July 16, 2013 6:34 PM
|||To: Muthu Arul Mozhi Perumal (mperumal)
|||Cc: rtcweb@ietf.org; behave@ietf.org
|||Subject: Re: [rtcweb] [BEHAVE] FW: I-D Action: draft-muthu-behave-consen=
t-freshness-04.txt
|||
|||Le 2013-07-16 14:43, Muthu Arul Mozhi Perumal (mperumal) a =E9crit :
|||> |> |>    Liveness timer: If no packets have been received on the local=
 port in
|||> |> |>    Tr seconds, the WebRTC browser MUST inform the JavaScript tha=
t
|||> |> |>    connectivity has been lost.  The JavaScript application will =
use this
|||> |> |>    notification however it desires (e.g., cease transmitting to =
the
|||> |> |>    remote peer, provide a notification to the user, etc.).
|||> |> |
|||> |> |This seems to me like it will not fulfill the goal set in the abst=
ract:
|||> |> |"to ensure that a malicious JavaScript cannot use the browser as a
|||> |> |platform for launching attacks". If the JavaScript is free to igno=
re the
|||> |> |notification from the browser, then it has no security benefits. I=
f you
|||> |> |want to reach that goal, the browser needs to forcefully stop tran=
smitting.
|||> |>
|||> |> That goal is fulfilled by the consent checks -- the browser would s=
top transmitting everything
||on
|||> |that candidate pair, including liveness checks, if there is a consent=
 failure.
|||> |
|||> |That's not what the draft says. It says that the browser "notifies" t=
he
|||> |JS app. It needs to say that the browser MUST stop sending.
|||>
|||> No. That section is about liveness check and its intention is just not=
ify the JavaScript of a
|||potential connectivity loss. It is when the consent check fails the brow=
ser actually stops sending
|||everything. Does the draft need more text on the distinction between con=
sent and liveness tests?
|||
|||Ah! No, you're right, and the text is already perfectly clear about
|||this. No need to change. I was just confused.
|||
|||> |> |>    When not actively sending traffic on a nominated candidate pa=
ir,
|||> |> |>    performing consent freshness does not serve any purpose from =
a
|||> |> |>    security perspective.
|||> |> |
|||> |> |I don't understand what this means. Why is the "security perspecti=
ve"
|||> |> |important here? Aren't we concerned about keepalives?
|||> |>
|||> |> You mean one could use keepalives (Binding indications) for launchi=
ng attacks, so consent
|||freshness
|||> |would be required for sending them as well?
|||> |
|||> |No.
|||> |
|||> |This is a section about keepalives. I just don't understand this
|||> |sentence, and I don't understand why it talks about security.
|||>
|||> Ok, let me elaborate:
|||> - Consent freshness is not necessary when the browser is not sending a=
ny
|||>   traffic on a candidate pair.
|||> - If the browser is not performing consent freshness on a candidate pa=
ir
|||>   for the above reason, it performs ICE keepalives (or RTP keepalives)=
 to
|||>   refresh NAT bindings.
|||>
|||> Of course, the browser could continue performing consent freshness eve=
n when it is not sending any
|||other traffic on that candidate pair and skip ICE keepalives.
|||
|||Ah, ok I understand with your explanation. It makes sense. There should
|||be a way to reformulate the text to make it clearer.
|||
|||Thanks,
|||Simon
|||_______________________________________________
|||rtcweb mailing list
|||rtcweb@ietf.org
|||https://www.ietf.org/mailman/listinfo/rtcweb

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Agree those are the right codecs to design for. Since in each case there
are fairly low limits on the number of supported layers (i.e. 3 spatial
layers for SVC), I think it should be possible to pack the temporal,
spatial, quality layer ids into 16 bits.


On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> If we can support VP8/9 as well as H.264/5 SVC
> that would be a start. It seems doable to me.
>
> On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me> wrote:
>
> Bernard,
>
> Are there other codecs you are thinking should be supported?  If it's
> generalized I would think we want to be able to cover all known scalable
> codecs. I'll look into the H264/SVC fields to see how to encode them in a
> generalized header.
>
> Regards,
> Adam
>
> On 7/18/13 7:40 PM, Bernard Aboba wrote:
>
> I think it may be possible to generalize this.  For example, for H.264/SVC
> which can support temporal, spatial and quality scalability, you would need
> the quality_id and dependency_id in addition to the temporal_id (what you
> call the temporal layer index).
>
>  ------------------------------
> Date: Thu, 18 Jul 2013 08:45:38 -0700
> From: fineberg@vline.me
> To: bernard_aboba@hotmail.com
> CC: rtcweb@ietf.org; avtext@ietf.org
> Subject: Re: [rtcweb] Fwd: New Version Notification for
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Bernard,
>
> Good question.  I'm not familiar enough with the parameter requirements of
> all other scalable codecs to be able to generalize.  If you'd like to help
> specify them, I'd be fine revising the draft to generalize.
>
> Regards,
> Adam
>
> On 7/17/13 8:26 PM, Bernard Aboba wrote:
>
> Since the need is not codec specific (e.g. it arises with any codec
> supporting temporal, spatial and quality scalability), why
>  a VP8-specific RTP extension?
>
>  ------------------------------
> Date: Wed, 17 Jul 2013 17:09:46 -0700
> From: fineberg@vline.me
> To: rtcweb@ietf.org
> Subject: [rtcweb] Fwd: New Version Notification for
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Hi,
>
> I'm working on WebRTC services and have found that while developing
> services that forward VP8 video streams if we want to take advantage of the
> VP8 temporal scaling we must get the temporal layer information from the
> RTP header which requires us to decrypt the SRTP packets. This is
> undesirable both because the middle-box needs to have access to the keys as
> well as the because of the added overhead of the decrypt/encrypt cycle.
> This draft proposes an RTP header extension that will allow us to use the
> VP8 temporal layer information included in the header extension and
> therefore do forwarding without SRTP decryption. Comments welcome.
>
> Regards,
> Adam Fineberg
> fineberg at vline.com
>
> -------- Original Message --------  Subject: New Version Notification for
> draft-fineberg-avtext-temporal-layer-ext-00.txt  Date: Tue, 09 Jul 2013
> 10:02:05 -0700  From: internet-drafts at ietf.org  To: Adam Fineberg <fineberg
> at vline.com> <fineberg%20at%20vline.com>
>
> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
> has been successfully submitted by Adam Fineberg and posted to the
> IETF repository.
>
> Filename:	 draft-fineberg-avtext-temporal-layer-ext
> Revision:	 00
> Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
> Creation date:	 2013-07-08
> Group:		 Individual Submission
> Number of pages: 6
> URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
> Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>
> Abstract:
>    This document defines a mechanism by which packets of Real-Time
>    Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>    indicate, in an RTP header extension, the temporal layer information
>    about the frame encoded in the RTP packet.  This information can be
>    used in a middlebox performing bandwidth management of streams
>    without requiring it to decrypt the streams.
>
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> --
> Regards,
> Adam
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--20cf303bf8be90697404e1dd5c94
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Agree those are the right codecs to design for. Since in e=
ach case there are fairly low limits on the number of supported layers (i.e=
. 3 spatial layers for SVC), I think it should be possible to pack the temp=
oral, spatial, quality layer ids into 16 bits.<div class=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:56 AM, Bernard=
 Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"auto"><div>If we can support VP8=
/9 as well as H.264/5 SVC</div><div>that would be a start. It seems doable =
to me.</div>


<div><div><div><br>On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &=
lt;<a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline.me=
</a>&gt; wrote:<br><br></div><blockquote type=3D"cite"><div>
 =20
   =20
 =20
 =20
    <div>Bernard,<br>
      <br>
      Are there other codecs you are thinking should be supported?=C2=A0 If
      it&#39;s generalized I would think we want to be able to cover all
      known scalable codecs. I&#39;ll look into the H264/SVC fields to see
      how to encode them in a generalized header.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <div dir=3D"ltr">I think it may be possible to generalize this.=C2=A0=
 For
        example, for H.264/SVC which can support temporal, spatial and
        quality scalability, you would need the quality_id and
        dependency_id in addition to the temporal_id (what you call the
        temporal layer index). =C2=A0=C2=A0 <br>
        <br>
        <div>
          <hr>Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
          From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fine=
berg@vline.me</a><br>
          To: <a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank=
">bernard_aboba@hotmail.com</a><br>
          CC: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@i=
etf.org</a>; <a href=3D"mailto:avtext@ietf.org" target=3D"_blank">avtext@ie=
tf.org</a><br>
          Subject: Re: [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          Bernard,<br>
          <br>
          Good question.=C2=A0 I&#39;m not familiar enough with the paramet=
er
          requirements of all other scalable codecs to be able to
          generalize.=C2=A0 If you&#39;d like to help specify them, I&#39;d=
 be fine
          revising the draft to generalize.<br>
          <br>
          Regards,<br>
          Adam<br>
          <br>
          <div>On 7/17/13 8:26 PM, Bernard
            Aboba wrote:<br>
          </div>
          <blockquote>
           =20
            <div dir=3D"ltr">Since the need is not codec specific (e.g. it
              arises with any codec supporting temporal, spatial and
              quality scalability), why <br>
              =C2=A0a VP8-specific RTP extension? <br>
              =C2=A0<br>
              <div>
                <hr>Date: Wed, 17 Jul 2013 17:09:46
                -0700<br>
                From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank=
">fineberg@vline.me</a><br>
                To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rt=
cweb@ietf.org</a><br>
                Subject: [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                <br>
                <span style=3D"text-indent:0px;letter-spacing:normal;text-t=
ransform:none;white-space:normal;display:inline!important;word-spacing:0px"=
>Hi,</span><br style=3D"text-indent:0px;letter-spacing:normal;text-transfor=
m:none;white-space:normal;word-spacing:0px">



                <br style=3D"text-indent:0px;letter-spacing:normal;text-tra=
nsform:none;white-space:normal;word-spacing:0px">
                <span style=3D"text-indent:0px;letter-spacing:normal;text-t=
ransform:none;white-space:normal;display:inline!important;word-spacing:0px"=
>I&#39;m working on WebRTC services and have
                  found that while developing services that forward VP8
                  video streams if we want to take advantage of the VP8
                  temporal scaling we must get the temporal layer
                  information from the RTP header which requires us to
                  decrypt the SRTP packets. This is undesirable both
                  because the middle-box needs to have access to the
                  keys as well as the because of the added overhead of
                  the decrypt/encrypt cycle. This draft proposes an RTP
                  header extension that will allow us to use the VP8
                  temporal layer information included in the header
                  extension and therefore do forwarding without SRTP
                  decryption. Comments welcome.</span><br style=3D"text-ind=
ent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-s=
pacing:0px">
                <br style=3D"text-indent:0px;letter-spacing:normal;text-tra=
nsform:none;white-space:normal;word-spacing:0px">
                <span style=3D"text-indent:0px;letter-spacing:normal;text-t=
ransform:none;white-space:normal;display:inline!important;word-spacing:0px"=
>Regards,</span><br style=3D"text-indent:0px;letter-spacing:normal;text-tra=
nsform:none;white-space:normal;word-spacing:0px">



                <span style=3D"text-indent:0px;letter-spacing:normal;text-t=
ransform:none;white-space:normal;display:inline!important;word-spacing:0px"=
>Adam Fineberg</span><br style=3D"text-indent:0px;letter-spacing:normal;tex=
t-transform:none;white-space:normal;word-spacing:0px">



                <div style=3D"text-indent:0px;letter-spacing:normal;text-tr=
ansform:none;white-space:normal;word-spacing:0px"><a href=3D"mailto:fineber=
g%20at%20vline.com" rel=3D"nofollow" target=3D"_blank">fineberg at vline.co=
m</a><br>



                  <br>
                  -------- Original Message --------
                  <table border=3D"0" cellpadding=3D"0" cellspacing=3D"0">
                    <tbody>
                      <tr>
                        <th align=3D"RIGHT" nowrap valign=3D"BASELINE">Subj=
ect:</th>
                        <td>New Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt</=
td>
                      </tr>
                      <tr>
                        <th align=3D"RIGHT" nowrap valign=3D"BASELINE">Date=
:</th>
                        <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                      </tr>
                      <tr>
                        <th align=3D"RIGHT" nowrap valign=3D"BASELINE">From=
:</th>
                        <td><a href=3D"mailto:internet-drafts%20at%20ietf.o=
rg" rel=3D"nofollow" target=3D"_blank">internet-drafts at ietf.org</a></td>
                      </tr>
                      <tr>
                        <th align=3D"RIGHT" nowrap valign=3D"BASELINE">To:<=
/th>
                        <td>Adam Fineberg<span>=C2=A0</span><a href=3D"mail=
to:fineberg%20at%20vline.com" rel=3D"nofollow" target=3D"_blank">&lt;finebe=
rg at vline.com&gt;</a></td>
                      </tr>
                    </tbody>
                  </table>
                  <br>
                  <br>
                  <pre style=3D"width:939.59px;white-space:pre-wrap">A new =
version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a href=3D"http://www.ietf.org/internet-drafts/draft-fineb=
erg-avtext-temporal-layer-ext-00.txt" rel=3D"nofollow" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-=
00.txt</a>
Status:          <a href=3D"http://datatracker.ietf.org/doc/draft-fineberg-=
avtext-temporal-layer-ext" rel=3D"nofollow" target=3D"_blank">http://datatr=
acker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a href=3D"http://tools.ietf.org/html/draft-fineberg-avtex=
t-temporal-layer-ext-00" rel=3D"nofollow" target=3D"_blank">http://tools.ie=
tf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                </div>
                <br>
                _______________________________________________ rtcweb
                mailing list <a href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a> <a href=3D"https://www.ietf.org/mailman/listinfo=
/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a>=
</div>



            </div>
          </blockquote>
          <br>
          <pre>--=20
Regards,
Adam</pre>
        </div>
      </div>
    </blockquote>
    <br>
 =20

</div></blockquote></div></div></div><br>__________________________________=
_____________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--20cf303bf8be90697404e1dd5c94--

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That's the set of codecs that seems interesting to most people right now

On Jul 18, 2013, at 10:56 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrot=
e:

> If we can support VP8/9 as well as H.264/5 SVC
> that would be a start. It seems doable to me.
>=20
> On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me> wrote:
>=20
>> Bernard,
>>=20
>> Are there other codecs you are thinking should be supported?  If it's gen=
eralized I would think we want to be able to cover all known scalable codecs=
. I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.
>>=20
>> Regards,
>> Adam
>>=20
>> On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>> I think it may be possible to generalize this.  For example, for H.264/S=
VC which can support temporal, spatial and quality scalability, you would ne=
ed the quality_id and dependency_id in addition to the temporal_id (what you=
 call the temporal layer index).   =20
>>>=20
>>> Date: Thu, 18 Jul 2013 08:45:38 -0700
>>> From: fineberg@vline.me
>>> To: bernard_aboba@hotmail.com
>>> CC: rtcweb@ietf.org; avtext@ietf.org
>>> Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-a=
vtext-temporal-layer-ext-00.txt
>>>=20
>>> Bernard,
>>>=20
>>> Good question.  I'm not familiar enough with the parameter requirements o=
f all other scalable codecs to be able to generalize.  If you'd like to help=
 specify them, I'd be fine revising the draft to generalize.
>>>=20
>>> Regards,
>>> Adam
>>>=20
>>> On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>> Since the need is not codec specific (e.g. it arises with any codec supp=
orting temporal, spatial and quality scalability), why=20
>>>  a VP8-specific RTP extension?=20
>>> =20
>>> Date: Wed, 17 Jul 2013 17:09:46 -0700
>>> From: fineberg@vline.me
>>> To: rtcweb@ietf.org
>>> Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtex=
t-temporal-layer-ext-00.txt
>>>=20
>>> Hi,
>>>=20
>>> I'm working on WebRTC services and have found that while developing serv=
ices that forward VP8 video streams if we want to take advantage of the VP8 t=
emporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both b=
ecause the middle-box needs to have access to the keys as well as the becaus=
e of the added overhead of the decrypt/encrypt cycle. This draft proposes an=
 RTP header extension that will allow us to use the VP8 temporal layer infor=
mation included in the header extension and therefore do forwarding without S=
RTP decryption. Comments welcome.
>>>=20
>>> Regards,
>>> Adam Fineberg
>>> fineberg at vline.com
>>>=20
>>> -------- Original Message --------
>>> Subject:	New Version Notification for draft-fineberg-avtext-temporal=
-layer-ext-00.txt
>>> Date:	Tue, 09 Jul 2013 10:02:05 -0700
>>> From:	internet-drafts at ietf.org
>>> To:	Adam Fineberg <fineberg at vline.com>
>>>=20
>>> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>> has been successfully submitted by Adam Fineberg and posted to the
>>> IETF repository.
>>>=20
>>> Filename:	 draft-fineberg-avtext-temporal-layer-ext
>>> Revision:	 00
>>> Title:		 A Real-Time Transport Protocol (RTP) Header Extens=
ion for VP8 Temporal Layer Information
>>> Creation date:	 2013-07-08
>>> Group:		 Individual Submission
>>> Number of pages: 6
>>> URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avte=
xt-temporal-layer-ext-00.txt
>>> Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-t=
emporal-layer-ext
>>> Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-tempor=
al-layer-ext-00
>>>=20
>>>=20
>>> Abstract:
>>>    This document defines a mechanism by which packets of Real-Time
>>>    Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>>    indicate, in an RTP header extension, the temporal layer information
>>>    about the frame encoded in the RTP packet.  This information can be
>>>    used in a middlebox performing bandwidth management of streams
>>>    without requiring it to decrypt the streams.
>>>=20
>>> _______________________________________________ rtcweb mailing list rtcw=
eb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>> --=20
>>> Regards,
>>> Adam
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

--Apple-Mail-120029B0-280A-471C-8749-E808561C9BC6
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	charset=utf-8
Content-Transfer-Encoding: 7bit

<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>That's the set of codecs that seems interesting to most people right now</div><div><br>On Jul 18, 2013, at 10:56 PM, Bernard Aboba &lt;<a href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div><meta http-equiv="content-type" content="text/html; charset=utf-8"><div>If we can support VP8/9 as well as H.264/5 SVC</div><div>that would be a start. It seems doable to me.</div><div><br>On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" &lt;<a href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt; wrote:<br><br></div><blockquote type="cite"><div>
  
    <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
  
  
    <div class="moz-cite-prefix">Bernard,<br>
      <br>
      Are there other codecs you are thinking should be supported?&nbsp; If
      it's generalized I would think we want to be able to cover all
      known scalable codecs. I'll look into the H264/SVC fields to see
      how to encode them in a generalized header.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W107EA5AE4A7E9D3DB6AFD0893630@phx.gbl" type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style>
      <div dir="ltr">I think it may be possible to generalize this.&nbsp; For
        example, for H.264/SVC which can support temporal, spatial and
        quality scalability, you would need the quality_id and
        dependency_id in addition to the temporal_id (what you call the
        temporal layer index). &nbsp;&nbsp; <br>
        <br>
        <div>
          <hr id="stopSpelling">Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
          From: <a class="moz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
          To: <a class="moz-txt-link-abbreviated" href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a><br>
          CC: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><br>
          Subject: Re: [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          Bernard,<br>
          <br>
          Good question.&nbsp; I'm not familiar enough with the parameter
          requirements of all other scalable codecs to be able to
          generalize.&nbsp; If you'd like to help specify them, I'd be fine
          revising the draft to generalize.<br>
          <br>
          Regards,<br>
          Adam<br>
          <br>
          <div class="ecxmoz-cite-prefix">On 7/17/13 8:26 PM, Bernard
            Aboba wrote:<br>
          </div>
          <blockquote cite="mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl">
            <style><!--
.ExternalClass .ecxhmmessage P {
padding:0px;
}

.ExternalClass body.ecxhmmessage {
font-size:12pt;
font-family:Calibri;
}

--></style>
            <div dir="ltr">Since the need is not codec specific (e.g. it
              arises with any codec supporting temporal, spatial and
              quality scalability), why <br>
              &nbsp;a VP8-specific RTP extension? <br>
              &nbsp;<br>
              <div>
                <hr id="ecxstopSpelling">Date: Wed, 17 Jul 2013 17:09:46
                -0700<br>
                From: <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                To: <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                Subject: [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                <br>
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Hi,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">I'm working on WebRTC services and have
                  found that while developing services that forward VP8
                  video streams if we want to take advantage of the VP8
                  temporal scaling we must get the temporal layer
                  information from the RTP header which requires us to
                  decrypt the SRTP packets. This is undesirable both
                  because the middle-box needs to have access to the
                  keys as well as the because of the added overhead of
                  the decrypt/encrypt cycle. This draft proposes an RTP
                  header extension that will allow us to use the VP8
                  temporal layer information included in the header
                  extension and therefore do forwarding without SRTP
                  decryption. Comments welcome.</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Regards,</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <span style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline
                  !important;white-space:normal;background-color:rgb(255,
                  255, 255);">Adam Fineberg</span><br style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);">
                <div class="ecxmoz-forward-container" style="color:rgb(0, 0,
                  0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,
                  255, 255);"><a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:fineberg%20at%20vline.com" rel="nofollow">fineberg at vline.com</a><br>
                  <br>
                  -------- Original Message --------
                  <table class="ecxmoz-email-headers-table" border="0" cellpadding="0" cellspacing="0">
                    <tbody>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
                        <td>New Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
                        <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
                        <td><a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow">internet-drafts at ietf.org</a></td>
                      </tr>
                      <tr>
                        <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
                        <td>Adam Fineberg<span class="ecxApple-converted-space">&nbsp;</span><a moz-do-not-send="true" class="ecxmoz-txt-link-rfc2396E" href="mailto:fineberg%20at%20vline.com" rel="nofollow">&lt;fineberg at vline.com&gt;</a></td>
                      </tr>
                    </tbody>
                  </table>
                  <br>
                  <br>
                  <pre style="width:939.59px;white-space:pre-wrap;-ms-word-wrap:break-word;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank" rel="nofollow">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank" rel="nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank" rel="nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                </div>
                <br>
                _______________________________________________ rtcweb
                mailing list <a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
            </div>
          </blockquote>
          <br>
          <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
        </div>
      </div>
    </blockquote>
    <br>
  

</div></blockquote></div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>avtext mailing list</span><br><span><a href="mailto:avtext@ietf.org">avtext@ietf.org</a></span><br><span><a href="https://www.ietf.org/mailman/listinfo/avtext">https://www.ietf.org/mailman/listinfo/avtext</a></span><br></div></blockquote></body></html>
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To: =?utf-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Please tell us the use case you have that requires this and it will make it e=
asier to figure out how to solve the pain.=20

On Jul 8, 2013, at 1:43 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/8 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
>>> So, was there ever a consensus that "developers MUST NOT
>>> touch SDP"?
>>=20
>> No, developers are free to touch the SDP (and probably must in certain
>> cases).
>=20
>=20
> Do you feel how much painful is "touching a browser generated SDP"? To be c=
lear:
>=20
> - JS requests "A" to browser.
> - Browser returns "A" (in the form of unmanageable blob).
> - JS modifies "A" to send "A-bis" in the wire.
> - Remote peer receives "A-bis" and replies "B-bis".
> - Both JS app are lying to their respective  browsers to get the
> desired behavior.
>=20
> This is really painful, really.
>=20
> In the other side, as I explained in other thread (may be in this
> one?) adding some methods to the current API (to avoid playing with
> the SDP) is not the way to go. A specification in which the JS app
> does not know what he is sending in-the-wire (a blob generated by the
> browser) is doomed to failure.
>=20
> Please remember that mandating SDP (plain SDP) means that no other
> media signaling protocol can be implemented in future WebRTC apps.
>=20
> SIP phones have *fixed* code and *fixed* logic. This is no longer true
> in the Web.
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Can someone please email a full copy of the spreadsheet to the list so it is=
 in the archives - thanks

On Jul 9, 2013, at 3:18 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/9 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
>> I want to be very clear and careful on what I say. So I am repeating:
>>=20
>> * My comment that I think Eric is right in that there is consensus on
>> providing APIs that allow most use-cases to be met without SDP mangling
>> is meant only in that context: SDP O/A is kept (and PeerConnection more
>> or less as is and so on).
>=20
> Hi Stefan,
>=20
> You insist that "there is consensus on keeping SDP O/A but providing a
> better API". Given current discussions IMHO it is clear there is not
> such a consensus (not at all). Or may be you are just talking about
> two years ago in some IETF meeting (if so I'm sorry).
>=20
> Please review the results in
> https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHl=
ZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1
>=20
>  Bad for advanced stuff: 94%
>  OK for 1.0: 40%
>=20
> IMHO such a result indicates all but "let's keep SDP O/A and improve a
> bit the API" (I mean now, in July 2013).
>=20
>=20
> Best regards.
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Cullen Jennings <fluffy@iii.ca>
Date: Fri, 19 Jul 2013 07:04:01 -0700
To: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the	current WebRTC API and SDP as a control surface
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Folks wanted to sort out how mindless works fist before getting to theses sm=
aller details but in general I think progress is being made on all of these

On Jul 17, 2013, at 2:58 PM, "Matthew Kaufman (SKYPE)" <matthew.kaufman@skyp=
e.net> wrote:

> Bernard Aboba:
>>=20
>> The problem was that we never defined what mangling browsers had to
>> support. Given all the SDP specs that is actually a huge work item.
>=20
>=20
> This is exactly what my slides that weren't presented last November starte=
d to touch upon. It only takes a few minutes with RFC3264 and its references=
 to start documenting cases that are clearly "valid SDP offer/answer" and ye=
t for which I cannot for the life of me figure out what a browser should do i=
f they're presented.
>=20
> Just off the top of my head:
>=20
> Can I...
> - Change the t=3D line to be something other than t=3D0 0?
> - Change the rtpmap associations before calling setLocal?
> - Change a=3Dsendrecv to a=3Drecvonly before calling setLocal?
> - What do you do when you see a=3Dcontent:sl ?
> - What if someone adds an r=3D or p=3D or e=3D?
> - What is the RFC that describes a=3Dgroup:BUNDLE (as seen in some browser=
 implementations)?
> - Can I remove a=3Dgroup:BUNDLE (or add it) before calling setLocal?
> - How about removing a=3Drtcp-mux?
> - Should I do something special at my end if you set a=3Dice-options:googl=
e-ice ?
> - If you put a=3Dssrc lines in there, can I change the ssrc before passing=
 it back to setLocal?
> - Can I delete the a=3Dssrc lines?
> - Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?
> - What about 0.0.0.0 in the o=3D line?
> - If I get a bunch of a=3Dcandidate lines, can I swap them around to chang=
e the priority before calling setLocal?
> - What if someone claims SAVP instead of SAVPF but gives me rtcp info?
>   =20
> At the *very least* for each and every line that comes from createOffer an=
d createAnswer you must be able to answer the following:
> - Can I delete it?
> - Duplicate it?
> - Change it?
> - If not, how are violation handled? (Both when passed to another browser a=
t the far end and when these modifications happen before calling setLocal)
> - And can I add additional valid SDP to what came from createOffer or crea=
teAnswer and pass it back to setLocal or not?
>=20
> We appear to be nowhere near a document which explains the answers to thes=
e questions, certainly not in the W3C WG, and not in any IETF RFC or draft I=
 can locate.
>=20
> Matthew Kaufman
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Cullen Jennings <fluffy@iii.ca>
Date: Fri, 19 Jul 2013 07:24:26 -0700
To: cowwoc <cowwoc@bbs.darktech.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Locus of API discussion
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On Jul 9, 2013, at 2:44 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:

>    Three weeks ago I posted a summary of discussion points that came up in=
 the WebRTC World conference (most of which had to do with the WebRTC API). T=
o date, I have not received a reply from any of the people you have listed. I=
 am unable to gather the necessary momentum to turn these points into action=
 items without your help. I was/am frustrated that the spec editors and vend=
ors are responsible to engage the community on these matters, but did not. I=
 hope this clarifies what I meant.

Many of us have been very busy updating drafts for ietf deadline so it as be=
en a hard time to respond.=20=

From matthew.kaufman@skype.net  Fri Jul 19 07:33:36 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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You say... "But I see the full definition of the SDP=0A=
parameters that browsers have to support as less important and=0A=
potentially just a by product of creating the higher level APIs."=0A=
=0A=
But I'm afraid that as a browser vendor, I will need exactly such a specifi=
cation in order to create an interoperable browser without looking at the o=
ther browser vendors' source code.... so even if your conclusion is correct=
, we can't avoid this step unless we also abandon SDP as one of the API sur=
faces.=0A=
=0A=
Matthew Kaufman=0A=
=0A=
________________________________________=0A=
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Silvia=
 Pfeiffer [silviapfeiffer1@gmail.com]=0A=
Sent: Thursday, July 18, 2013 5:46 PM=0A=
To: Martin Thomson=0A=
Cc: <rtcweb@ietf.org>; public-webrtc@w3.org=0A=
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the cu=
rrent WebRTC API and SDP as a control surface=0A=
=0A=
On Fri, Jul 19, 2013 at 2:53 AM, Martin Thomson=0A=
<martin.thomson@gmail.com> wrote:=0A=
> On 18 July 2013 08:12, Peter Thatcher <pthatcher@google.com> wrote:=0A=
>> I believe I began paying attention to the mailing lists after you sent o=
ut=0A=
>> theses slides that you didn't present.  I'm interested in seeing them, a=
nd=0A=
>> while I could dig through archives to find them, if convenient, could yo=
u=0A=
>> please give me a link to the slides?  Thanks.=0A=
>=0A=
> It wasn't actually November, it was October, which made this harder to=0A=
> find than I had expected.=0A=
>=0A=
> http://lists.w3.org/Archives/Public/public-webrtc/2012Oct/0148.html=0A=
=0A=
=0A=
This captures exactly the kind of questions and concerns I had. Excellent w=
ork!=0A=
=0A=
However, I don't fully agree with the conclusion of the slide deck.=0A=
I'd prefer we extended the constraints and other browser APIs that set=0A=
the SDP parameters rather than (or: in preference to) fully specifying=0A=
what SDP the browser has to support. I do like the ability to get the=0A=
low-level access to mangle the SDP as a means of experimenting with=0A=
new functionality or as a means to try and connect to devices that=0A=
don't have a WebRTC API. But I see the full definition of the SDP=0A=
parameters that browsers have to support as less important and=0A=
potentially just a by product of creating the higher level APIs.=0A=
=0A=
What does it take for us to get focused on defining such an API that=0A=
is independent of SDP for the JS developer, and for now requires=0A=
browsers to do the mapping to SDP for them? Is the extension of the=0A=
constraints that a JS dev can manipulate enough for this?=0A=
=0A=
Silvia.=0A=
_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb=0A=
=0A=


From pthatcher@google.com  Fri Jul 19 07:50:33 2013
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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 07:49:50 -0700
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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"mindless works fist"?  There's a deep meaning there, but I'm still trying
to figure it out...


On Fri, Jul 19, 2013 at 7:04 AM, Cullen Jennings <fluffy@iii.ca> wrote:

> Folks wanted to sort out how mindless works fist before getting to theses
> smaller details but in general I think progress is being made on all of
> these
>
> On Jul 17, 2013, at 2:58 PM, "Matthew Kaufman (SKYPE)" <
> matthew.kaufman@skype.net> wrote:
>
> > Bernard Aboba:
> >>
> >> The problem was that we never defined what mangling browsers had to
> >> support. Given all the SDP specs that is actually a huge work item.
> >
> >
> > This is exactly what my slides that weren't presented last November
> started to touch upon. It only takes a few minutes with RFC3264 and its
> references to start documenting cases that are clearly "valid SDP
> offer/answer" and yet for which I cannot for the life of me figure out what
> a browser should do if they're presented.
> >
> > Just off the top of my head:
> >
> > Can I...
> > - Change the t= line to be something other than t=0 0?
> > - Change the rtpmap associations before calling setLocal?
> > - Change a=sendrecv to a=recvonly before calling setLocal?
> > - What do you do when you see a=content:sl ?
> > - What if someone adds an r= or p= or e=?
> > - What is the RFC that describes a=group:BUNDLE (as seen in some browser
> implementations)?
> > - Can I remove a=group:BUNDLE (or add it) before calling setLocal?
> > - How about removing a=rtcp-mux?
> > - Should I do something special at my end if you set
> a=ice-options:google-ice ?
> > - If you put a=ssrc lines in there, can I change the ssrc before passing
> it back to setLocal?
> > - Can I delete the a=ssrc lines?
> > - Is a=rtcp:1 IN IP4 0.0.0.0 valid or not?
> > - What about 0.0.0.0 in the o= line?
> > - If I get a bunch of a=candidate lines, can I swap them around to
> change the priority before calling setLocal?
> > - What if someone claims SAVP instead of SAVPF but gives me rtcp info?
> >
> > At the *very least* for each and every line that comes from createOffer
> and createAnswer you must be able to answer the following:
> > - Can I delete it?
> > - Duplicate it?
> > - Change it?
> > - If not, how are violation handled? (Both when passed to another
> browser at the far end and when these modifications happen before calling
> setLocal)
> > - And can I add additional valid SDP to what came from createOffer or
> createAnswer and pass it back to setLocal or not?
> >
> > We appear to be nowhere near a document which explains the answers to
> these questions, certainly not in the W3C WG, and not in any IETF RFC or
> draft I can locate.
> >
> > Matthew Kaufman
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">&quot;<span style=3D"font-family:arial,sans-serif;font-siz=
e:12.727272033691406px">mindless works fist&quot;? =C2=A0There&#39;s a deep=
 meaning there, but I&#39;m still trying to figure it out...</span></div><d=
iv class=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 7:04 AM, Cullen =
Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_=
blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">

Folks wanted to sort out how mindless works fist before getting to theses s=
maller details but in general I think progress is being made on all of thes=
e<br>
<br>
On Jul 17, 2013, at 2:58 PM, &quot;Matthew Kaufman (SKYPE)&quot; &lt;<a hre=
f=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a>&gt; wr=
ote:<br>
<br>
&gt; Bernard Aboba:<br>
&gt;&gt;<br>
&gt;&gt; The problem was that we never defined what mangling browsers had t=
o<br>
&gt;&gt; support. Given all the SDP specs that is actually a huge work item=
.<br>
&gt;<br>
&gt;<br>
&gt; This is exactly what my slides that weren&#39;t presented last Novembe=
r started to touch upon. It only takes a few minutes with RFC3264 and its r=
eferences to start documenting cases that are clearly &quot;valid SDP offer=
/answer&quot; and yet for which I cannot for the life of me figure out what=
 a browser should do if they&#39;re presented.<br>


&gt;<br>
&gt; Just off the top of my head:<br>
&gt;<br>
&gt; Can I...<br>
&gt; - Change the t=3D line to be something other than t=3D0 0?<br>
&gt; - Change the rtpmap associations before calling setLocal?<br>
&gt; - Change a=3Dsendrecv to a=3Drecvonly before calling setLocal?<br>
&gt; - What do you do when you see a=3Dcontent:sl ?<br>
&gt; - What if someone adds an r=3D or p=3D or e=3D?<br>
&gt; - What is the RFC that describes a=3Dgroup:BUNDLE (as seen in some bro=
wser implementations)?<br>
&gt; - Can I remove a=3Dgroup:BUNDLE (or add it) before calling setLocal?<b=
r>
&gt; - How about removing a=3Drtcp-mux?<br>
&gt; - Should I do something special at my end if you set a=3Dice-options:g=
oogle-ice ?<br>
&gt; - If you put a=3Dssrc lines in there, can I change the ssrc before pas=
sing it back to setLocal?<br>
&gt; - Can I delete the a=3Dssrc lines?<br>
&gt; - Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?<br>
&gt; - What about 0.0.0.0 in the o=3D line?<br>
&gt; - If I get a bunch of a=3Dcandidate lines, can I swap them around to c=
hange the priority before calling setLocal?<br>
&gt; - What if someone claims SAVP instead of SAVPF but gives me rtcp info?=
<br>
&gt;<br>
&gt; At the *very least* for each and every line that comes from createOffe=
r and createAnswer you must be able to answer the following:<br>
&gt; - Can I delete it?<br>
&gt; - Duplicate it?<br>
&gt; - Change it?<br>
&gt; - If not, how are violation handled? (Both when passed to another brow=
ser at the far end and when these modifications happen before calling setLo=
cal)<br>
&gt; - And can I add additional valid SDP to what came from createOffer or =
createAnswer and pass it back to setLocal or not?<br>
&gt;<br>
&gt; We appear to be nowhere near a document which explains the answers to =
these questions, certainly not in the W3C WG, and not in any IETF RFC or dr=
aft I can locate.<br>
&gt;<br>
&gt; Matthew Kaufman<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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From: Peter Thatcher <pthatcher@google.com>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Here you go:




On Fri, Jul 19, 2013 at 7:00 AM, Cullen Jennings <fluffy@iii.ca> wrote:

> Can someone please email a full copy of the spreadsheet to the list so it
> is in the archives - thanks
>
> On Jul 9, 2013, at 3:18 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote=
:
>
> > 2013/7/9 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> >> I want to be very clear and careful on what I say. So I am repeating:
> >>
> >> * My comment that I think Eric is right in that there is consensus on
> >> providing APIs that allow most use-cases to be met without SDP manglin=
g
> >> is meant only in that context: SDP O/A is kept (and PeerConnection mor=
e
> >> or less as is and so on).
> >
> > Hi Stefan,
> >
> > You insist that "there is consensus on keeping SDP O/A but providing a
> > better API". Given current discussions IMHO it is clear there is not
> > such a consensus (not at all). Or may be you are just talking about
> > two years ago in some IETF meeting (if so I'm sorry).
> >
> > Please review the results in
> >
> https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdH=
lZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1
> >
> >  Bad for advanced stuff: 94%
> >  OK for 1.0: 40%
> >
> > IMHO such a result indicates all but "let's keep SDP O/A and improve a
> > bit the API" (I mean now, in July 2013).
> >
> >
> > Best regards.
> >
> >
> >
> > --
> > I=C3=B1aki Baz Castillo
> > <ibc@aliax.net>
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Here you go:<div><br></div><div><br></div></div><div class=
=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Jul 19, 2013 at=
 7:00 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@ii=
i.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Can someone please email a full copy of the =
spreadsheet to the list so it is in the archives - thanks<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Jul 9, 2013, at 3:18 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:i=
bc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
<br>
&gt; 2013/7/9 Stefan H=C3=A5kansson LK &lt;<a href=3D"mailto:stefan.lk.haka=
nsson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;:<br>
&gt;&gt; I want to be very clear and careful on what I say. So I am repeati=
ng:<br>
&gt;&gt;<br>
&gt;&gt; * My comment that I think Eric is right in that there is consensus=
 on<br>
&gt;&gt; providing APIs that allow most use-cases to be met without SDP man=
gling<br>
&gt;&gt; is meant only in that context: SDP O/A is kept (and PeerConnection=
 more<br>
&gt;&gt; or less as is and so on).<br>
&gt;<br>
&gt; Hi Stefan,<br>
&gt;<br>
&gt; You insist that &quot;there is consensus on keeping SDP O/A but provid=
ing a<br>
&gt; better API&quot;. Given current discussions IMHO it is clear there is =
not<br>
&gt; such a consensus (not at all). Or may be you are just talking about<br=
>
&gt; two years ago in some IETF meeting (if so I&#39;m sorry).<br>
&gt;<br>
&gt; Please review the results in<br>
&gt; <a href=3D"https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0=
AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1" target=3D"_blank">http=
s://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHlZdV9R=
N0xSWFhybVl4anJLRkVPV0E#gid=3D1</a><br>


&gt;<br>
&gt; =C2=A0Bad for advanced stuff: 94%<br>
&gt; =C2=A0OK for 1.0: 40%<br>
&gt;<br>
&gt; IMHO such a result indicates all but &quot;let&#39;s keep SDP O/A and =
improve a<br>
&gt; bit the API&quot; (I mean now, in July 2013).<br>
&gt;<br>
&gt;<br>
&gt; Best regards.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; I=C3=B1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 19 Jul 2013 17:03:37 +0200
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/19 Cullen Jennings <fluffy@iii.ca>:
> Folks wanted to sort out how mindless works fist before getting to theses=
 smaller details

Hi Cullen,

All these "smaller details" are in fact issues and problems we don't
need. Issues not needed at all in WebRTC, issues that WebRTC
developers and vendors should NOT care about. If those "smaller
details" do exist is due the mandate of SDP. And all the time the WG
(or WGs) will spent "fixing/defining" them is just wasted time, since
nothing useful is being done for WebRTC by "defining" those "minor
details".

Regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From cowwoc@bbs.darktech.org  Fri Jul 19 08:54:33 2013
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On 19/07/2013 10:24 AM, Cullen Jennings wrote:
> On Jul 9, 2013, at 2:44 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>
>>     Three weeks ago I posted a summary of discussion points that came up in the WebRTC World conference (most of which had to do with the WebRTC API). To date, I have not received a reply from any of the people you have listed. I am unable to gather the necessary momentum to turn these points into action items without your help. I was/am frustrated that the spec editors and vendors are responsible to engage the community on these matters, but did not. I hope this clarifies what I meant.
> Many of us have been very busy updating drafts for ietf deadline so it as been a hard time to respond.
Hi Cullen,

     Welcome back :)

     That's understandable but next time might I suggest you reply 
immediately with: "You bring up good points but I am busy updating 
drafts for the ietf deadline. I'll reply as soon as I get this done..." 
At least then there is an acknowledgement that my points are being taken 
into consideration and an expectation that a response will arrive in the 
near future. The current process is very broken. We have a lot of 
discussions on the mailing list without any concrete action items. When 
a discussion thread goes unanswered, there is no follow-up. On the one 
hand, most organizations don't conduct such long discussions over a bug 
tracker but on the other hand a bug tracker provides the assurance that 
an issue will remain open until it is resolved to everyone's mutual 
satisfaction. Issues don't close silently as happens on mailing lists. 
We need to find a way to fix our process.

     Another thing I don't understand: As far as I know, the Working 
Group consists of 
http://www.w3.org/2000/09/dbwg/details?group=47318&public=1&order=org. 
That's around 78 people, yet none of them followed up on this. Where is 
the list of Working Group members that have an obligation to follow-up 
on these kind of posts? Meaning, who could I reasonably expect to reply?

Thanks,
Gili

From stewe@stewe.org  Fri Jul 19 09:03:44 2013
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To: Justin Uberti <juberti@google.com>, Bernard Aboba <bernard_aboba@hotmail.com>
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--_000_CE0E9F569F89Bstewesteweorg_
Content-Type: text/plain; charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling=97as the encrypted in-band VPS would not be useful under the assu=
mption that the middle box does not have the key to the media=97which is th=
e motivation of the draft in the first place).  I personally don't mind at =
all the second mechanism, as I'm a big fan of out-of-band parameter set tra=
nsmission and any middle box must be in the signaling path anyway to meanin=
gfully manipulate RTP.  I do not like the first option due to its variable,=
 and possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.


On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:

Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).

________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam

On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:        New Version Notification for draft-fineberg-avtext-temporal=
-layer-ext-00.txt
Date:   Tue, 09 Jul 2013 10:02:05 -0700
From:   internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>
To:     Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.=
com>



A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:        draft-fineberg-avtext-temporal-layer-ext
Revision:        00
Title:           A Real-Time Transport Protocol (RTP) Header Extension for =
VP8 Temporal Layer Information
Creation date:   2013-07-08
Group:           Individual Submission
Number of pages: 6
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.


_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--
Regards,
Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb



--_000_CE0E9F569F89Bstewesteweorg_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <733E8B5C549C77459614F7BC7CA923A3@namprd07.prod.outlook.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>Hi,</div>
<div>I also believe that 16 bits should be enough. &nbsp;For H.264 and VP8 =
that has already been demonstrated. &nbsp;For H.265, some initial thoughts =
below. &nbsp;Apologies for the word-count.</div>
<div><br>
</div>
<div>The scalable version of H265 (called SHVC) is currently under developm=
ent. &nbsp;The current working draft can be found here:&nbsp;<a href=3D"htt=
p://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M10=
08-v3.zip">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/=
wg11/JCTVC-M1008-v3.zip</a>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.</div>
<div><br>
</div>
<div>That said, the ordering by the encoder is just a recommendation, and t=
here may well be cases where different pruning strategies may be advisable.=
 &nbsp;For example, a layering structure could be constructed that expands =
into two branches, one using 2D scalable
 tools only, the other including view_id for multi view coding. &nbsp;By lo=
oking at the six bit field alone, a middle box will not be able to meaningf=
ully remove NAL units belonging to one of the branches completely while pru=
ning the other branch. &nbsp;In order to meaningfully
 deal with that scenario, there would be two options: one to represent the =
dimension_id[][] (and associated control info) in the header extension, or =
require the middle box to have access to the VPS and be able to interpret i=
ts content. &nbsp;The further could take
 considerably more than 16 bits and we would be talking about a variable le=
ngth data structure. &nbsp;The latter requires the middle box to have state=
 and a mechanism to intercept the VPS (through signaling=97as the encrypted=
 in-band VPS would not be useful under
 the assumption that the middle box does not have the key to the media=97wh=
ich is the motivation of the draft in the first place). &nbsp;I personally =
don't mind at all the second mechanism, as I'm a big fan of out-of-band par=
ameter set transmission and any middle
 box must be in the signaling path anyway to meaningfully manipulate RTP. &=
nbsp;I do not like the first option due to its variable, and possibly subst=
antial, overhead.</div>
<div><br>
</div>
<div>Stephan &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Justin Uberti &lt;<a href=3D"=
mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 19 July, 2013 06:32 <=
br>
<span style=3D"font-weight:bold">To: </span>Bernard Aboba &lt;<a href=3D"ma=
ilto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:avtext@=
ietf.org">avtext@ietf.org</a>&quot; &lt;<a href=3D"mailto:avtext@ietf.org">=
avtext@ietf.org</a>&gt;, &quot;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ie=
tf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>=
&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Fwd: New Vers=
ion Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">Agree those are the right codecs to design for. Since in e=
ach case there are fairly low limits on the number of supported layers (i.e=
. 3 spatial layers for SVC), I think it should be possible to pack the temp=
oral, spatial, quality layer ids into
 16 bits.
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_=
aboba@hotmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>If we can support VP8/9 as well as H.264/5 SVC</div>
<div>that would be a start. It seems doable to me.</div>
<div>
<div>
<div><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a href=3D"mailt=
o:fineberg@vline.me" target=3D"_blank">fineberg@vline.me</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div>Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">I think it may be possible to generalize this.&nbsp; For e=
xample, for H.264/SVC which can support temporal, spatial and quality scala=
bility, you would need the quality_id and dependency_id in addition to the =
temporal_id (what you call the temporal
 layer index). &nbsp;&nbsp; <br>
<br>
<div>
<hr>
Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_=
aboba@hotmail.com</a><br>
CC: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
>; <a href=3D"mailto:avtext@ietf.org" target=3D"_blank">
avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam<br>
<br>
<div>On 7/17/13 8:26 PM, Bernard Aboba wrote:<br>
</div>
<blockquote>
<div dir=3D"ltr">Since the need is not codec specific (e.g. it arises with =
any codec supporting temporal, spatial and quality scalability), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;<br>
<div>
<hr>
Date: Wed, 17 Jul 2013 17:09:46 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br s=
tyle=3D"text-indent:0px;letter-spacing:normal;text-transform:none;white-spa=
ce:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">I'm working on =
WebRTC services and have found that while developing services that forward =
VP8 video streams if we want to take
 advantage of the VP8 temporal scaling we must get the temporal layer infor=
mation from the RTP header which requires us to decrypt the SRTP packets. T=
his is undesirable both because the middle-box needs to have access to the =
keys as well as the because of the
 added overhead of the decrypt/encrypt cycle. This draft proposes an RTP he=
ader extension that will allow us to use the VP8 temporal layer information=
 included in the header extension and therefore do forwarding without SRTP =
decryption. Comments welcome.</span><br style=3D"text-indent:0px;letter-spa=
cing:normal;text-transform:none;white-space:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Regards,</span>=
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Adam Fineberg</=
span><br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none=
;white-space:normal;word-spacing:0px">
<div style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whi=
te-space:normal;word-spacing:0px">
<a href=3D"mailto:fineberg%20at%20vline.com" rel=3D"nofollow" target=3D"_bl=
ank">fineberg at vline.com</a><br>
<br>
-------- Original Message --------
<table border=3D"0" cellpadding=3D"0" cellspacing=3D"0">
<tbody>
<tr>
<th align=3D"RIGHT" nowrap=3D"" valign=3D"BASELINE">Subject:</th>
<td>New Version Notification for draft-fineberg-avtext-temporal-layer-ext-0=
0.txt</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"" valign=3D"BASELINE">Date:</th>
<td>Tue, 09 Jul 2013 10:02:05 -0700</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"" valign=3D"BASELINE">From:</th>
<td><a href=3D"mailto:internet-drafts%20at%20ietf.org" rel=3D"nofollow" tar=
get=3D"_blank">internet-drafts at ietf.org</a></td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"" valign=3D"BASELINE">To:</th>
<td>Adam Fineberg<span>&nbsp;</span><a href=3D"mailto:fineberg%20at%20vline=
.com" rel=3D"nofollow" target=3D"_blank">&lt;fineberg at vline.com&gt;</a><=
/td>
</tr>
</tbody>
</table>
<br>
<br>
<pre style=3D"width:939.59px;white-space:pre-wrap">A new version of I-D, dr=
aft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a href=3D"http://www.ietf.org/internet-drafts/draft-fineb=
erg-avtext-temporal-layer-ext-00.txt" rel=3D"nofollow" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-=
00.txt</a>
Status:          <a href=3D"http://datatracker.ietf.org/doc/draft-fineberg-=
avtext-temporal-layer-ext" rel=3D"nofollow" target=3D"_blank">http://datatr=
acker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a href=3D"http://tools.ietf.org/html/draft-fineberg-avtex=
t-temporal-layer-ext-00" rel=3D"nofollow" target=3D"_blank">http://tools.ie=
tf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
</div>
<br>
_______________________________________________ rtcweb mailing list <a href=
=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb=
" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
</div>
</blockquote>
<br>
<pre>--=20
Regards,
Adam</pre>
</div>
</div>
</blockquote>
<br>
</div>
</blockquote>
</div>
</div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_CE0E9F569F89Bstewesteweorg_--

From salvatore.loreto@ericsson.com  Fri Jul 19 09:12:17 2013
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] data channel reset
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good catch!
we will clarify and make it explicitly in the next version of the draft!

thanks for the feedback

/Sal

On 7/19/13 12:39 AM, Martin Thomson wrote:
> This statement, or something like it, is made in several places in the draft:
>
> "Channels are closed by an SCTP reset of the Stream."
>
> I assume that this uses the "Outgoing SSN Reset Request Parameter",
> http://tools.ietf.org/html/rfc6525#section-4.1
>
> Should it say that explicitly?
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 10:03, IÃ±aki Baz Castillo wrote:
> All these "smaller details" are in fact issues and problems we don't
> need. Issues not needed at all in WebRTC, issues that WebRTC
> developers and vendors should NOT care about. If those "smaller
> details" do exist is due the mandate of SDP. And all the time the WG
> (or WGs) will spent "fixing/defining" them is just wasted time, since
> nothing useful is being done for WebRTC by "defining" those "minor
> details".
>

I think this is a hopelessly naÃ¯ve interpretation of the facts on the 
ground. Simply discarding SDP doesn't magically make the underlying 
issues go away. We would still need to settle a vast number of issues 
around things like simulcast, FEC, codec parameters, indication of 
supported codecs, correlation of RTP streams with MediaStreamTracks, 
attempts by both parties to operate on the same stream simultaneously 
[1], etc. Basically, with very rare exception, the same set of problems 
that we need to solve if we *don't* throw SDP out the window.

I understand the temptation to think that starting over makes all the 
problems go away. There's a mental trap in thinking that all you really 
need is to announce ports and codecs and get on with it. But then this 
person over here really needs simulcast. And that person over there 
insists that RTCP NACK feedback is critical for his application. And 
then I need to be able to tell you that your 1280x720 video stream is 
going to overwhelm my limited ability to decode and that you really need 
to turn it down to QCIF. And, before you know it, you've reinvented 
something approximately as complex as SDP that everyone is just going to 
shove into a JSON blob and send across the wire. As an added bonus, by 
deciding that legacy interop is of no value, you've limited the utility 
of the overall system by setting Metcalfe's law on fire and throwing it 
over the railing of the third-floor balcony.

Your pain point isn't SDP syntax. Yeah, it's ugly, but it's not hard. 
Your pain point isn't offer/answer. Two unilaterally declared sessions 
that are simply blasted out onto the wire only satisfies the simplest of 
use cases; you need a negotiation, and any attempt to define how that 
negotiation looks is going to arrive at something with enough rules that 
it is substantially as complicated as offer/answer.

No, your pain point here is that non-master-slave networked 
communications are not easy to get right, and it is the height of hubris 
to think that you inherently know better than everyone else who has 
worked in this space before you. Consider that TCP has far fewer moving 
parts than even a simple one-stream-audio-only call setup and took 85 
pages to specify.

I understand comment 22 at its core [2], but it has a corollary: any 
system that replaces SDP O/A will end up being similar in complexity 
once all interested parties' use cases have been factored in.

/a

____
[1] What we call "glare" in telecom and SIP, but a phenomenon that 
arises whenever you have connections made between peers without a 
master/slave arrangement; see RFC 793, page 32, for example.

[2] For the uninitiated, "comment 22" was a shorthand developed at last 
year's TPAC for the sentiment "SDP offer/answer is hard".

From martin.thomson@gmail.com  Fri Jul 19 09:54:22 2013
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On 19 July 2013 09:37, Adam Roach <adam@nostrum.com> wrote:
> [2] For the uninitiated, "comment 22" was a shorthand developed at last
> year's TPAC for the sentiment "SDP offer/answer is hard".

No quite, it's shorthand for "there is a proposal that has been
submitted in which the problem in question does not need to be
solved", with one of two options: "either the problem has been
solved", or "the problem simply does not need to be solved (by us,
with a standard)".

I reject your thesis for one simple reason: scope.

As it turns out, "does not need to be solved" doesn't go to 100%.
Some problems are deferred to applications.  Intentionally.  Because
a) they are better at that than we are, clearly, and b) they don't
necessarily want our crappy solutions.

How long have we talked about BUNDLE?  How long do you think that it
would take someone with a functioning RTP library to build something
that multiplexes and demultiplexes RTP streams?

Negotiation is a hole.  A vast, soul-sucking, waste of time.

From ted.ietf@gmail.com  Fri Jul 19 10:07:20 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson <martin.thomson@gmail.com>wrote:

>
> Negotiation is a hole.  A vast, soul-sucking, waste of time.
>
>
Even if you have the same javascript application downloaded, you will have
disparate capabilities in the environments into which it is downloaded
(browser/os/codecs/media sources/available network capacity).  Getting set
intersection and preference order for those capabilities is something that
applications actually want.  You may be able to move the pain of that
around, but it isn't a waste of time.

My personal opinion, with no IETF hats or company swag,

Ted

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On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gma=
il.com</a>&gt;</span> wrote:<br><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
<br>
Negotiation is a hole. =A0A vast, soul-sucking, waste of time.<br>
<br>
</blockquote></div><br>Even if you have the same javascript application dow=
nloaded, you will have disparate capabilities in the environments into whic=
h it is downloaded (browser/os/codecs/media sources/available network capac=
ity).=A0 Getting set intersection and preference order for those capabiliti=
es is something that applications actually want.=A0 You may be able to move=
 the pain of that around, but it isn&#39;t a waste of time.<br>
<br>My personal opinion, with no IETF hats or company swag,<br><br>Ted<br>

--089e015370969b50ad04e1e05bfc--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 11:54, Martin Thomson wrote:
> As it turns out, "does not need to be solved" doesn't go to 100%.
> Some problems are deferred to applications.  Intentionally.  Because
> a) they are better at that than we are, clearly, and b) they don't
> necessarily want our crappy solutions.
>
> How long have we talked about BUNDLE?  How long do you think that it
> would take someone with a functioning RTP library to build something
> that multiplexes and demultiplexes RTP streams?

Are you proposing that Firefox come up with its own multiplexing 
mechanism for RTP; Chrome its own; Opera, yet a third; IE, a fourth; and 
Safari, a fifth?  And then we just kind of pray that five proprietary 
solutions developed in a vacuum miraculously work together? I mean, 
yeah, if we can rely on miracle interop for independently-developed 
proprietary solutions, I guess that works.

Or are you envisioning a WebRTC API that requires javascript 
applications to supply their own RTP stacks?

/a

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 10:18:09 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 9:37 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 10:03, I=C3=B1aki Baz Castillo wrote:
>
>> All these "smaller details" are in fact issues and problems we don't
>> need. Issues not needed at all in WebRTC, issues that WebRTC
>> developers and vendors should NOT care about. If those "smaller
>> details" do exist is due the mandate of SDP. And all the time the WG
>> (or WGs) will spent "fixing/defining" them is just wasted time, since
>> nothing useful is being done for WebRTC by "defining" those "minor
>> details".
>>
>>
> I think this is a hopelessly na=C3=AFve interpretation of the facts on th=
e
> ground. Simply discarding SDP doesn't magically make the underlying issue=
s
> go away. We would still need to settle a vast number of issues around
> things like simulcast, FEC, codec parameters, indication of supported
> codecs, correlation of RTP streams with MediaStreamTracks, attempts by bo=
th
> parties to operate on the same stream simultaneously [1], etc. Basically,
> with very rare exception, the same set of problems that we need to solve =
if
> we *don't* throw SDP out the window.
>

It's interesting that most or your list of things that needed to be solved
without SDP (simulcast, FEC, correlation of RTP streams with
MediaStreamTracks, glare) still haven't been solved for WebRTC even with
SDP, despite many months (years?) of effort.

I don't think anyone is saying "without SDP, there would be issues".  I
think they're saying "without SDP, the issues are much easier/faster to
solve, and the API would be much more usable and implementable".   And I
don't think that's a naive opinion.




>
> I understand the temptation to think that starting over makes all the
> problems go away. There's a mental trap in thinking that all you really
> need is to announce ports and codecs and get on with it. But then this
> person over here really needs simulcast. And that person over there insis=
ts
> that RTCP NACK feedback is critical for his application. And then I need =
to
> be able to tell you that your 1280x720 video stream is going to overwhelm
> my limited ability to decode and that you really need to turn it down to
> QCIF.  And, before you know it, you've reinvented something approximately
> as complex as SDP


I think you are conflating signalling with API surface.  If we focus on
making a good API rather than on signalling, we can make something much
more simple and leave signalling up to the application.


> that everyone is just going to shove into a JSON blob and send across the
> wire.  As an added bonus, by deciding that legacy interop is of no value,
> you've limited the utility of the overall system by setting Metcalfe's la=
w
> on fire and throwing it over the railing of the third-floor balcony.


Again, you are conflating API surface and signalling.  An API that doesn't
have SDP in it doesn't preclude web apps from using SDP for signalling.
 There's no loss here for those app that choose to use SDP.  There is only
gain for web developers chooses not to use SDP for the signalling in their
web application.  And if they choose not to use SDP and use JSON instead,
why are you so opposed?  There is a world outside of legacy interop, you
know.




>
>



> Your pain point isn't SDP syntax. Yeah, it's ugly, but it's not hard. You=
r
> pain point isn't offer/answer. Two unilaterally declared sessions that ar=
e
> simply blasted out onto the wire only satisfies the simplest of use cases=
;
> you need a negotiation, and any attempt to define how that negotiation
> looks is going to arrive at something with enough rules that it is
> substantially as complicated as offer/answer.
>

You are underestimating how painful it is to deal with the current API,
both because you are very familiar with it (with many years of experience)
and because the use cases you care about are the ones the API was defined
for (legacy interop).  But as soon as you ask someone who is not familiar
with SDP (a lot of web devleopers), or someone who is doing more advanced
or different things than what you have thought of, they will give you a
different answer.   They feel pain, and it is because of the poor API
surface.



>
> No, your pain point here is that non-master-slave networked communication=
s
> are not easy to get right, and it is the height of hubris to think that y=
ou
> inherently know better than everyone else who has worked in this space
> before you. Consider that TCP has far fewer moving parts than even a simp=
le
> one-stream-audio-only call setup and took 85 pages to specify.
>

Please try to consider what you sound like to a web developer who is trying
to use WebRTC and finding SDP to be difficult API surface.  Maybe I'm
wrong, but I'm guessing what you're saying will sound like.  "my use case
(legacy interop) is the most important and my solution (SDP) is the best
solution for everyone and I know better because I've been working on it for
longer".  Hubris?


>
> I understand comment 22 at its core [2], but it has a corollary: any
> system that replaces SDP O/A will end up being similar in complexity once
> all interested parties' use cases have been factored in.
>

There is a vast difference that you are not taking into account.  The SDP
O/A, if in JS, is a complexity cost only paid by those web applications
that need it.  But if baked into the API, every app must pay the cost of
the complexity.

In general, this emails seems to be arguing that  SDP is the optimal API
surface for expressing all these complex things, and  yet in other
occasions, I've heard you admit it's a terrible API surface.  So does that
mean the optimal API surface is terrible?  I don't think so.  It will be
work, but I think the work will be worth it in the end.

That said, the chairs have asked us to delay discussing a better API, so
we'll have to wait before we can discuss details of how an API could be
done without SDP.


>
> /a
>
> ____
> [1] What we call "glare" in telecom and SIP, but a phenomenon that arises
> whenever you have connections made between peers without a master/slave
> arrangement; see RFC 793, page 32, for example.
>
> [2] For the uninitiated, "comment 22" was a shorthand developed at last
> year's TPAC for the sentiment "SDP offer/answer is hard".
>
>

--047d7b5d438c26fb5104e1e085c4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 9:37 AM, Adam Roach <span dir=3D"ltr">&lt;<=
a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&g=
t;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">On 7/19/13 10:03, I=C3=B1aki Baz Castillo wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
All these &quot;smaller details&quot; are in fact issues and problems we do=
n&#39;t<br>
need. Issues not needed at all in WebRTC, issues that WebRTC<br>
developers and vendors should NOT care about. If those &quot;smaller<br>
details&quot; do exist is due the mandate of SDP. And all the time the WG<b=
r>
(or WGs) will spent &quot;fixing/defining&quot; them is just wasted time, s=
ince<br>
nothing useful is being done for WebRTC by &quot;defining&quot; those &quot=
;minor<br>
details&quot;.<br>
<br>
</blockquote>
<br>
I think this is a hopelessly na=C3=AFve interpretation of the facts on the =
ground. Simply discarding SDP doesn&#39;t magically make the underlying iss=
ues go away. We would still need to settle a vast number of issues around t=
hings like simulcast, FEC, codec parameters, indication of supported codecs=
, correlation of RTP streams with MediaStreamTracks, attempts by both parti=
es to operate on the same stream simultaneously [1], etc. Basically, with v=
ery rare exception, the same set of problems that we need to solve if we *d=
on&#39;t* throw SDP out the window.<br>

</blockquote><div><br></div><div>It&#39;s interesting that most or your lis=
t of things that needed to be solved without SDP (simulcast, FEC, correlati=
on of RTP streams with MediaStreamTracks, glare) still haven&#39;t been sol=
ved for WebRTC even with SDP, despite many months (years?) of effort. =C2=
=A0</div>

<div><br></div><div>I don&#39;t think anyone is saying &quot;without SDP, t=
here would be issues&quot;. =C2=A0I think they&#39;re saying &quot;without =
SDP, the issues are much easier/faster to solve, and the API would be much =
more usable and implementable&quot;. =C2=A0 And I don&#39;t think that&#39;=
s a naive opinion.</div>

<div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<br>
I understand the temptation to think that starting over makes all the probl=
ems go away. There&#39;s a mental trap in thinking that all you really need=
 is to announce ports and codecs and get on with it. But then this person o=
ver here really needs simulcast. And that person over there insists that RT=
CP NACK feedback is critical for his application. And then I need to be abl=
e to tell you that your 1280x720 video stream is going to overwhelm my limi=
ted ability to decode and that you really need to turn it down to QCIF. =C2=
=A0And, before you know it, you&#39;ve reinvented something approximately a=
s complex as SDP </blockquote>

<div><br></div><div>I think you are conflating signalling with API surface.=
 =C2=A0If we focus on making a good API rather than on signalling, we can m=
ake something much more simple and leave signalling up to the application.<=
/div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">that everyone is just going to shove into=
 a JSON blob and send across the wire. =C2=A0As an added bonus, by deciding=
 that legacy interop is of no value, you&#39;ve limited the utility of the =
overall system by setting Metcalfe&#39;s law on fire and throwing it over t=
he railing of the third-floor balcony.</blockquote>

<div><br></div><div>Again, you are conflating API surface and signalling. =
=C2=A0An API that doesn&#39;t have SDP in it doesn&#39;t preclude web apps =
from using SDP for signalling. =C2=A0There&#39;s no loss here for those app=
 that choose to use SDP. =C2=A0There is only gain for web developers choose=
s not to use SDP for the signalling in their web application. =C2=A0And if =
they choose not to use SDP and use JSON instead, why are you so opposed? =
=C2=A0There is a world outside of legacy interop, you know. =C2=A0</div>

<div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<br></blockquote><div><br></div><div><br></div><div>=C2=A0</div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:=
1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left=
:1ex">


Your pain point isn&#39;t SDP syntax. Yeah, it&#39;s ugly, but it&#39;s not=
 hard. Your pain point isn&#39;t offer/answer. Two unilaterally declared se=
ssions that are simply blasted out onto the wire only satisfies the simples=
t of use cases; you need a negotiation, and any attempt to define how that =
negotiation looks is going to arrive at something with enough rules that it=
 is substantially as complicated as offer/answer.<br>

</blockquote><div><br></div><div>You are underestimating how painful it is =
to deal with the current API, both because you are very familiar with it (w=
ith many years of experience) and because the use cases you care about are =
the ones the API was defined for (legacy interop). =C2=A0But as soon as you=
 ask someone who is not familiar with SDP (a lot of web devleopers), or som=
eone who is doing more advanced or different things than what you have thou=
ght of, they will give you a different answer. =C2=A0 They feel pain, and i=
t is because of the poor API surface.</div>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex">
<br>
No, your pain point here is that non-master-slave networked communications =
are not easy to get right, and it is the height of hubris to think that you=
 inherently know better than everyone else who has worked in this space bef=
ore you. Consider that TCP has far fewer moving parts than even a simple on=
e-stream-audio-only call setup and took 85 pages to specify.<br>

</blockquote><div><br></div><div>Please try to consider what you sound like=
 to a web developer who is trying to use WebRTC and finding SDP to be diffi=
cult API surface. =C2=A0Maybe I&#39;m wrong, but I&#39;m guessing what you&=
#39;re saying will sound like. =C2=A0&quot;my use case (legacy interop) is =
the most important and my solution (SDP) is the best solution for everyone =
and I know better because I&#39;ve been working on it for longer&quot;. =C2=
=A0Hubris? =C2=A0</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">
<br>
I understand comment 22 at its core [2], but it has a corollary: any system=
 that replaces SDP O/A will end up being similar in complexity once all int=
erested parties&#39; use cases have been factored in.<br></blockquote>

<div><br></div><div>There is a vast difference that you are not taking into=
 account. =C2=A0The SDP O/A, if in JS, is a complexity cost only paid by th=
ose web applications that need it. =C2=A0But if baked into the API, every a=
pp must pay the cost of the complexity. =C2=A0</div>

<div><br></div><div>In general, this emails seems to be arguing that =C2=A0=
SDP is the optimal API surface for expressing all these complex things, and=
 =C2=A0yet in other occasions, I&#39;ve heard you admit it&#39;s a terrible=
 API surface. =C2=A0So does that mean the optimal API surface is terrible? =
=C2=A0I don&#39;t think so. =C2=A0It will be work, but I think the work wil=
l be worth it in the end. =C2=A0</div>

<div><br></div><div>That said, the chairs have asked us to delay discussing=
 a better API, so we&#39;ll have to wait before we can discuss details of h=
ow an API could be done without SDP.=C2=A0</div><div>=C2=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width=
:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-lef=
t:1ex">


<br>
/a<br>
<br>
____<br>
[1] What we call &quot;glare&quot; in telecom and SIP, but a phenomenon tha=
t arises whenever you have connections made between peers without a master/=
slave arrangement; see RFC 793, page 32, for example.<br>
<br>
[2] For the uninitiated, &quot;comment 22&quot; was a shorthand developed a=
t last year&#39;s TPAC for the sentiment &quot;SDP offer/answer is hard&quo=
t;.<br>
<br>
</blockquote></div><br></div></div>

--047d7b5d438c26fb5104e1e085c4--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 19 July 2013 10:07, Ted Hardie <ted.ietf@gmail.com> wrote:
> Even if you have the same javascript application downloaded, you will have
> disparate capabilities in the environments into which it is downloaded
> (browser/os/codecs/media sources/available network capacity).  Getting set
> intersection and preference order for those capabilities is something that
> applications actually want.  You may be able to move the pain of that
> around, but it isn't a waste of time.

There's a vast difference between "it needs to be done" and "we need
to define a standard for it".

I have nothing against providing tools that make it easier, but it's a
real leap of logic to conclude that the one and only mechanism for
interaction *requires* negotiation and that SDP O/A is *the*
negotiation method.

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 19 July 2013 10:09, Adam Roach <adam@nostrum.com> wrote:
> Are you proposing that Firefox come up with its own multiplexing mechanism
> for RTP; Chrome its own; Opera, yet a third; IE, a fourth; and Safari, a
> fifth?  And then we just kind of pray that five proprietary solutions
> developed in a vacuum miraculously work together? I mean, yeah, if we can
> rely on miracle interop for independently-developed proprietary solutions, I
> guess that works.

I'm saying nothing of the sort.  We definitely need to define what
multiplexing looks like when it happens, but there's a vast gulf
between that and defining how to negotiate it.  Discussions on RTP
shims or SSRC mux did consume some time, but I believe them to be
done.  For quite a while now.

The negotiation part looks like it's nearing completion, but only
after a fairly significant amount of wrangling and pain.

From pthatcher@google.com  Fri Jul 19 10:24:40 2013
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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 10:23:59 -0700
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 10:09 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 11:54, Martin Thomson wrote:
>
>> As it turns out, "does not need to be solved" doesn't go to 100%.
>> Some problems are deferred to applications.  Intentionally.  Because
>> a) they are better at that than we are, clearly, and b) they don't
>> necessarily want our crappy solutions.
>>
>> How long have we talked about BUNDLE?  How long do you think that it
>> would take someone with a functioning RTP library to build something
>> that multiplexes and demultiplexes RTP streams?
>>
>
> Are you proposing that Firefox come up with its own multiplexing mechanism
> for RTP; Chrome its own; Opera, yet a third; IE, a fourth; and Safari, a
> fifth?  And then we just kind of pray that five proprietary solutions
> developed in a vacuum miraculously work together? I mean, yeah, if we can
> rely on miracle interop for independently-developed proprietary solutions,
> I guess that works.
>
>
Again, you are conflating signalling and API surface.  We could define a
simple API that leaves the control up the application, and different
applications can signal different multiplexing techniques in different
ways.


> Or are you envisioning a WebRTC API that requires javascript applications
> to supply their own RTP stacks?


SDP != RTP.   It is possible to have an RTP stack without an SDP stack.
 You realize this, right?



>
>
> /a
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--047d7b111cff05e44404e1e09a8e
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:09 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/19/13 11:54, Martin T=
homson wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
As it turns out, &quot;does not need to be solved&quot; doesn&#39;t go to 1=
00%.<br>
Some problems are deferred to applications. =C2=A0Intentionally. =C2=A0Beca=
use<br>
a) they are better at that than we are, clearly, and b) they don&#39;t<br>
necessarily want our crappy solutions.<br>
<br>
How long have we talked about BUNDLE? =C2=A0How long do you think that it<b=
r>
would take someone with a functioning RTP library to build something<br>
that multiplexes and demultiplexes RTP streams?<br>
</blockquote>
<br></div>
Are you proposing that Firefox come up with its own multiplexing mechanism =
for RTP; Chrome its own; Opera, yet a third; IE, a fourth; and Safari, a fi=
fth? =C2=A0And then we just kind of pray that five proprietary solutions de=
veloped in a vacuum miraculously work together? I mean, yeah, if we can rel=
y on miracle interop for independently-developed proprietary solutions, I g=
uess that works.<br>


<br></blockquote><div><br></div><div>Again, you are conflating signalling a=
nd API surface. =C2=A0We could define a simple API that leaves the control =
up the application, and different applications can signal different multipl=
exing techniques in different ways.=C2=A0</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
Or are you envisioning a WebRTC API that requires javascript applications t=
o supply their own RTP stacks?</blockquote><div><br></div><div>SDP !=3D RTP=
. =C2=A0 It is possible to have an RTP stack without an SDP stack. =C2=A0Yo=
u realize this, right?</div>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=
=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
/a</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--047d7b111cff05e44404e1e09a8e--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 12:07, Ted Hardie wrote:
> On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson 
> <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
>
>
>     Negotiation is a hole.  A vast, soul-sucking, waste of time.
>
>
> Even if you have the same javascript application downloaded, you will 
> have disparate capabilities in the environments into which it is 
> downloaded (browser/os/codecs/media sources/available network 
> capacity).  Getting set intersection and preference order for those 
> capabilities is something that applications actually want. You may be 
> able to move the pain of that around, but it isn't a waste of time.

I can't +1 this hard enough. I certainly don't want every javascript 
application that makes use of the WebRTC API to independently discover 
that mobile terminal Foocom A1 runnning MobileOS 3.1.7(a) bogs down to 
unusable if you try to send it more than 320x200 video, and then try to 
solve that problem.

Again and again, for every permutation of phone and operating system 
version.

Yeah, someone has to do this kind of characterization, and some of it 
can be done real-time if you're interacting directly with the operating 
system. So... maybe we could add yet another API to WebRTC to allow 
applications to build this functionality themselves rather than counting 
on them characterizing the systems they care about and blowing up on the 
ones they don't.

But, honestly, any course of action that relegates this to the 
applications seems to have the dual properties of forcing it to be 
implemented hundreds of thousands of times while making the actual user 
experience worse.

/a

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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 7/19/13 12:07, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMCjt2UHFynLqwns0J0f5ZtnxtMX3ppzR66e5q_rJ9D5Ug@mail.gmail.com"
      type="cite">On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson <span
        dir="ltr">&lt;<a moz-do-not-send="true"
          href="mailto:martin.thomson@gmail.com" target="_blank">martin.thomson@gmail.com</a>&gt;</span>
      wrote:<br>
      <div class="gmail_quote">
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <br>
          Negotiation is a hole. &nbsp;A vast, soul-sucking, waste of time.<br>
          <br>
        </blockquote>
      </div>
      <br>
      Even if you have the same javascript application downloaded, you
      will have disparate capabilities in the environments into which it
      is downloaded (browser/os/codecs/media sources/available network
      capacity).&nbsp; Getting set intersection and preference order for
      those capabilities is something that applications actually want.&nbsp;
      You may be able to move the pain of that around, but it isn't a
      waste of time.<br>
    </blockquote>
    <br>
    I can't +1 this hard enough. I certainly don't want every javascript
    application that makes use of the WebRTC API to independently
    discover that mobile terminal Foocom A1 runnning MobileOS 3.1.7(a)
    bogs down to unusable if you try to send it more than 320x200 video,
    and then try to solve that problem.<br>
    <br>
    Again and again, for every permutation of phone and operating system
    version.<br>
    <br>
    Yeah, someone has to do this kind of characterization, and some of
    it can be done real-time if you're interacting directly with the
    operating system. So... maybe we could add yet another API to WebRTC
    to allow applications to build this functionality themselves rather
    than counting on them characterizing the systems they care about and
    blowing up on the ones they don't.<br>
    <br>
    But, honestly, any course of action that relegates this to the
    applications seems to have the dual properties of forcing it to be
    implemented hundreds of thousands of times while making the actual
    user experience worse.<br>
    <br>
    /a<br>
  </body>
</html>

--------------090605080100040702020800--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 10:27:16 -0700
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 10:25 AM, Adam Roach <adam@nostrum.com> wrote:

>  On 7/19/13 12:07, Ted Hardie wrote:
>
> On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson <martin.thomson@gmail.com>wrote:
>
>>
>> Negotiation is a hole.  A vast, soul-sucking, waste of time.
>>
>>
> Even if you have the same javascript application downloaded, you will have
> disparate capabilities in the environments into which it is downloaded
> (browser/os/codecs/media sources/available network capacity).  Getting set
> intersection and preference order for those capabilities is something that
> applications actually want.  You may be able to move the pain of that
> around, but it isn't a waste of time.
>
>
> I can't +1 this hard enough. I certainly don't want every javascript
> application that makes use of the WebRTC API to independently discover that
> mobile terminal Foocom A1 runnning MobileOS 3.1.7(a) bogs down to unusable
> if you try to send it more than 320x200 video, and then try to solve that
> problem.
>
> Again and again, for every permutation of phone and operating system
> version.
>
> Yeah, someone has to do this kind of characterization, and some of it can
> be done real-time if you're interacting directly with the operating system.
> So... maybe we could add yet another API to WebRTC to allow applications to
> build this functionality themselves rather than counting on them
> characterizing the systems they care about and blowing up on the ones they
> don't.
>
> But, honestly, any course of action that relegates this to the
> applications seems to have the dual properties of forcing it to be
> implemented hundreds of thousands of times while making the actual user
> experience worse.
>

Hundreds of thousands of times?


>
>
> /a
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:25 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div class=3D"h5">
    <div>On 7/19/13 12:07, Ted Hardie wrote:<br>
    </div>
    <blockquote type=3D"cite">On Fri, Jul 19, 2013 at 9:54 AM, Martin Thoms=
on <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=
=3D"_blank">martin.thomson@gmail.com</a>&gt;</span>
      wrote:<br>
      <div class=3D"gmail_quote">
        <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex">
          <br>
          Negotiation is a hole. =C2=A0A vast, soul-sucking, waste of time.=
<br>
          <br>
        </blockquote>
      </div>
      <br>
      Even if you have the same javascript application downloaded, you
      will have disparate capabilities in the environments into which it
      is downloaded (browser/os/codecs/media sources/available network
      capacity).=C2=A0 Getting set intersection and preference order for
      those capabilities is something that applications actually want.=C2=
=A0
      You may be able to move the pain of that around, but it isn&#39;t a
      waste of time.<br>
    </blockquote>
    <br></div></div>
    I can&#39;t +1 this hard enough. I certainly don&#39;t want every javas=
cript
    application that makes use of the WebRTC API to independently
    discover that mobile terminal Foocom A1 runnning MobileOS 3.1.7(a)
    bogs down to unusable if you try to send it more than 320x200 video,
    and then try to solve that problem.<br>
    <br>
    Again and again, for every permutation of phone and operating system
    version.<br>
    <br>
    Yeah, someone has to do this kind of characterization, and some of
    it can be done real-time if you&#39;re interacting directly with the
    operating system. So... maybe we could add yet another API to WebRTC
    to allow applications to build this functionality themselves rather
    than counting on them characterizing the systems they care about and
    blowing up on the ones they don&#39;t.<br>
    <br>
    But, honestly, any course of action that relegates this to the
    applications seems to have the dual properties of forcing it to be
    implemented hundreds of thousands of times while making the actual
    user experience worse.</div></blockquote><div><br></div><div>Hundreds o=
f thousands of times?</div><div>=C2=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
<div bgcolor=3D"#FFFFFF" text=3D"#000000">

<span class=3D"HOEnZb"><font color=3D"#888888"><br>
    <br>
    /a<br>
  </font></span></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--bcaec51f99e5bd0b8404e1e0a559--

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From: Roman Shpount <roman@telurix.com>
To: Adam Roach <adam@nostrum.com>
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 12:37 PM, Adam Roach <adam@nostrum.com> wrote:

> I think this is a hopelessly na=EFve interpretation of the facts on the
> ground. Simply discarding SDP doesn't magically make the underlying issue=
s
> go away. We would still need to settle a vast number of issues around
> things like simulcast, FEC, codec parameters, indication of supported
> codecs, correlation of RTP streams with MediaStreamTracks, attempts by bo=
th
> parties to operate on the same stream simultaneously [1], etc. Basically,
> with very rare exception, the same set of problems that we need to solve =
if
> we *don't* throw SDP out the window.
>

I would argue that we are not as na=EFve as you think.

First of all, a lot of the issues you are talking about are not yet solved.
At the same time we have something already working and useful with current
browser implementation. So, what we want is to have these features released
as version 1.0. The problem that we face, that when you implement all the
above mentioned features, even though the API will stay the same, SDP that
it will produce will change, and that will require all, except the simplest
application, and all external components interfacing WebRTC clients to
change. And then when more features are implemented and more different
versions are in use more and more interop for applications and external
components will be required. What we want is an API that is not based on
blob with unpredictable data.

Second, offer/answer and SDP introduce a number of negotiation constraints
that really should not be there. They cause unrelated things to be
negotiated together. It should be possible to negotiate a data connection
and then add and remove media streams without renegotiating the underlying
data connection. To force selection of single codec you need to do two
offer/answer exchanges, which caused by no other reason then framework
limitations. Resolution of glare is restricted to offer back off.
Furthermore, even from the point of view of offer/answer the current API is
broken. It does not allow forking. It introduces a new PRANSWER which is
not a part of normal offer/answer specification. It does not allow change
send media parameters (ie change send codec or send codec parameters)
without doing an offer/answer exchange, which is not necessary with normal
offer/answer framework.

So, what we are proposing is to map exiting implemented browser
functionality to API and get rid of SDP. Ideally we should get rid of
offer/answer as well but if it is too much work we can leave with it for
version 1.0. This way we get predictable behavior based on predictable
input. If we miss some of the future capabilities and they would require
API modification to implement, we can deal with it in 2.0. In API, unlike
network protocols, predictability and testability counts more then ability
for future extension.
_____________
Roman Shpount

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<br><div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 12:37 PM, Adam Roach=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank=
">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I think this is a hopelessly na=EFve interpretation of the facts on the gro=
und. Simply discarding SDP doesn&#39;t magically make the underlying issues=
 go away. We would still need to settle a vast number of issues around thin=
gs like simulcast, FEC, codec parameters, indication of supported codecs, c=
orrelation of RTP streams with MediaStreamTracks, attempts by both parties =
to operate on the same stream simultaneously [1], etc. Basically, with very=
 rare exception, the same set of problems that we need to solve if we *don&=
#39;t* throw SDP out the window.<br>
</blockquote><div><br></div><div>I would argue that we are not as na=EFve a=
s you think.</div><div><br></div><div>First of all, a lot of the issues you=
 are talking about are not yet solved. At the same time we have something a=
lready working and useful with current browser implementation. So, what we =
want is to have these features released as version 1.0. The problem that we=
 face, that when you implement all the above mentioned features, even thoug=
h the API will stay the same, SDP that it will produce will change, and tha=
t will require all, except the simplest application, and all external compo=
nents interfacing WebRTC clients to change. And then when more features are=
 implemented and more different versions are in use more and more interop f=
or applications and external components will be required. What we want is a=
n API that is not based on blob with unpredictable data.</div>
<div><br></div><div>Second, offer/answer and SDP introduce a number of nego=
tiation constraints that really should not be there. They cause unrelated t=
hings to be negotiated together. It should be possible to negotiate a data =
connection and then add and remove media streams without renegotiating the =
underlying data connection. To force selection of single codec you need to =
do two offer/answer exchanges, which caused by no other reason then framewo=
rk limitations. Resolution of glare is restricted to offer back off. Furthe=
rmore, even from the point of view of offer/answer the current API is broke=
n. It does not allow forking. It introduces a new PRANSWER which is not a p=
art of normal offer/answer specification. It does not allow change send med=
ia parameters (ie change send codec or send codec parameters) without doing=
 an offer/answer exchange, which is not necessary with normal offer/answer =
framework.=A0</div>
<div><br></div><div>So, what we are proposing is to map exiting implemented=
 browser functionality to API and get rid of SDP. Ideally we should get rid=
 of offer/answer as well but if it is too much work we can leave with it fo=
r version 1.0. This way we get predictable behavior based on predictable in=
put. If we miss some of the future capabilities and they would require API =
modification to implement, we can deal with it in 2.0. In API, unlike netwo=
rk protocols, predictability and testability counts more then ability for f=
uture extension.=A0<br>
_____________=A0</div><div>Roman Shpount=A0</div></div>

--001a11c3669e5c4faf04e1e0ab5d--

From adam@nostrum.com  Fri Jul 19 10:31:19 2013
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 12:27, Peter Thatcher wrote:
>
>     But, honestly, any course of action that relegates this to the
>     applications seems to have the dual properties of forcing it to be
>     implemented hundreds of thousands of times while making the actual
>     user experience worse.
>
>
> Hundreds of thousands of times?
>

How many web apps do you anticipate will use real time communications 
between now and the time that this work becomes obsolete?

It's not important to come to consensus on this number -- my key point 
here is that it's significantly more than the number of web browsers 
that will be developed in that time, and by several orders of magnitude.

/a

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<html>
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    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 7/19/13 12:27, Peter Thatcher wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrXDUGLwOEWCbZU8vS53pW9fZt_RdeKgmmw9My-dbgDn_PkqQ@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra">
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div bgcolor="#FFFFFF" text="#000000"> But, honestly, any
                course of action that relegates this to the applications
                seems to have the dual properties of forcing it to be
                implemented hundreds of thousands of times while making
                the actual user experience worse.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Hundreds of thousands of times?</div>
            <div>Â <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    How many web apps do you anticipate will use real time
    communications between now and the time that this work becomes
    obsolete?<br>
    <br>
    It's not important to come to consensus on this number -- my key
    point here is that it's significantly more than the number of web
    browsers that will be developed in that time, and by several orders
    of magnitude.<br>
    <br>
    /a<br>
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--------------040208060104080403030705--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 12:29, Roman Shpount wrote:
> First of all, a lot of the issues you are talking about are not yet 
> solved.

That was my *point*.

/a

From martin.thomson@gmail.com  Fri Jul 19 10:37:22 2013
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 19 July 2013 10:34, Adam Roach <adam@nostrum.com> wrote:
> On 7/19/13 12:29, Roman Shpount wrote:
>>
>> First of all, a lot of the issues you are talking about are not yet
>> solved.
>
> That was my *point*.

So what, the logical conclusion is to tell people that are working on
(or have proposed) solutions to go away because they don't fit with
your preconceptions about what a solution should look like?

From roman@telurix.com  Fri Jul 19 10:37:47 2013
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Date: Fri, 19 Jul 2013 13:37:33 -0400
Message-ID: <CAD5OKxsvUzXttQmdZrpR8-dW8UGY0srdRHSmcRsE2Jibqs+E0Q@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Adam Roach <adam@nostrum.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 1:25 PM, Adam Roach <adam@nostrum.com> wrote:

>  I can't +1 this hard enough. I certainly don't want every javascript
> application that makes use of the WebRTC API to independently discover that
> mobile terminal Foocom A1 runnning MobileOS 3.1.7(a) bogs down to unusable
> if you try to send it more than 320x200 video, and then try to solve that
> problem.
>
> Again and again, for every permutation of phone and operating system
> version.
>
> Yeah, someone has to do this kind of characterization, and some of it can
> be done real-time if you're interacting directly with the operating system.
> So... maybe we could add yet another API to WebRTC to allow applications to
> build this functionality themselves rather than counting on them
> characterizing the systems they care about and blowing up on the ones they
> don't.
>
> But, honestly, any course of action that relegates this to the
> applications seems to have the dual properties of forcing it to be
> implemented hundreds of thousands of times while making the actual user
> experience worse.
>

You cannot be more wrong. Each application deals with predictable set of
scenarios (ones it produces) and interfaces with predictable set of non
webrtc devices. These external devices are under control of the application
developer. Application developer is responsible for interop with these
devices. What is needed is predictable media and signaling generated by
webrtc devices running the same application. With negotiation and SDP
backed into the webrtc device it is very hard to achieve. So, for me as
webrtc developer let me develop a negotiation scheme that works for me and
make sure it stay working regardless of the browser type or version. Right
now controlling webrtc application feels like driving the boat down the
river by tickling the captain.
_____________
Roman Shpount

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<div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:25 PM, Adam Roach <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">ada=
m@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div class=3D"h5">
    <div>I can&#39;t +1 this hard enough. I certainly don&#39;t want every =
javascript
    application that makes use of the WebRTC API to independently
    discover that mobile terminal Foocom A1 runnning MobileOS 3.1.7(a)
    bogs down to unusable if you try to send it more than 320x200 video,
    and then try to solve that problem.</div></div></div>
    <br>
    Again and again, for every permutation of phone and operating system
    version.<br>
    <br>
    Yeah, someone has to do this kind of characterization, and some of
    it can be done real-time if you&#39;re interacting directly with the
    operating system. So... maybe we could add yet another API to WebRTC
    to allow applications to build this functionality themselves rather
    than counting on them characterizing the systems they care about and
    blowing up on the ones they don&#39;t.<br>
    <br>
    But, honestly, any course of action that relegates this to the
    applications seems to have the dual properties of forcing it to be
    implemented hundreds of thousands of times while making the actual
    user experience worse.</div></blockquote><div><br></div><div>You cannot=
 be more wrong. Each application deals with predictable set of scenarios (o=
nes it produces) and interfaces with predictable set of non webrtc devices.=
 These external devices are under control of the application developer. App=
lication developer is responsible for interop with these devices. What is n=
eeded is predictable media and signaling generated by webrtc devices runnin=
g the same application. With negotiation and SDP backed into the webrtc dev=
ice it is very hard to achieve. So, for me as webrtc developer let me devel=
op a negotiation scheme that works for me and make sure it stay working reg=
ardless of the browser type or version. Right now controlling webrtc applic=
ation feels like driving the boat down the river by tickling the captain.</=
div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--089e0112cf9a2ad3f304e1e0c8ce--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 10:37:31 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 10:31 AM, Adam Roach <adam@nostrum.com> wrote:

>  On 7/19/13 12:27, Peter Thatcher wrote:
>
>
>   But, honestly, any course of action that relegates this to the
>> applications seems to have the dual properties of forcing it to be
>> implemented hundreds of thousands of times while making the actual user
>> experience worse.
>>
>
>  Hundreds of thousands of times?
>
>
>
> How many web apps do you anticipate will use real time communications
> between now and the time that this work becomes obsolete?
>

That's an excellent point.  Along with that, how many will only be using
the data channel?  And how many will be using something other than SDP for
signalling?  And how many will be doing lots of SDP munging?  And how many
libraries will be written to hide the ugliness of SDP?  Hundreds of
thousands?


>
> It's not important to come to consensus on this number -- my key point
> here is that it's significantly more than the number of web browsers that
> will be developed in that time, and by several orders of magnitude.
>

That's an excellent point.  If we spent a little bit of extra work making a
really good API, think of how many people will benefit.  It's several
orders of magnitude more than the number of web browser that will need to
do a little bit of extra work to implement a good API.


>
>
> /a
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:31 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im">
    <div>On 7/19/13 12:27, Peter Thatcher wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><br>
        <div class=3D"gmail_extra">
          <div class=3D"gmail_quote">
            <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">
              <div bgcolor=3D"#FFFFFF" text=3D"#000000"> But, honestly, any
                course of action that relegates this to the applications
                seems to have the dual properties of forcing it to be
                implemented hundreds of thousands of times while making
                the actual user experience worse.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Hundreds of thousands of times?</div>
            <div>=C2=A0<br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div>
    How many web apps do you anticipate will use real time
    communications between now and the time that this work becomes
    obsolete?<br></div></blockquote><div><br class=3D"">That&#39;s an excel=
lent point. =C2=A0Along with that, how many will only be using the data cha=
nnel? =C2=A0And how many will be using something other than SDP for signall=
ing? =C2=A0And how many will be doing lots of SDP munging? =C2=A0And how ma=
ny libraries will be written to hide the ugliness of SDP? =C2=A0Hundreds of=
 thousands?<br>

</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bo=
rder-left-style:solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#0=
00000">
    <br>
    It&#39;s not important to come to consensus on this number -- my key
    point here is that it&#39;s significantly more than the number of web
    browsers that will be developed in that time, and by several orders
    of magnitude.</div></blockquote><div><br></div><div>That&#39;s an excel=
lent point. =C2=A0If we spent a little bit of extra work making a really go=
od API, think of how many people will benefit. =C2=A0It&#39;s several order=
s of magnitude more than the number of web browser that will need to do a l=
ittle bit of extra work to implement a good API. =C2=A0</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000"=
><span class=3D""><font color=3D"#888888"><br>


    <br>
    /a<br>
  </font></span></div>

</blockquote></div><br></div></div>

--bcaec520e9f772be3404e1e0ca00--

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Date: Fri, 19 Jul 2013 13:38:50 -0400
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From: Roman Shpount <roman@telurix.com>
To: Adam Roach <adam@nostrum.com>
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 1:34 PM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 12:29, Roman Shpount wrote:
>
>> First of all, a lot of the issues you are talking about are not yet
>> solved.
>>
>
> That was my *point*.
>
>
And my point is that each time you solve any of those issue you will break
all the applications working so far. Not a good API design.
_____________
Roman Shpount

--001a11c355feb738e304e1e0cc64
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<div><br></div><div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:34 PM, =
Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=
=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
<div class=3D"im">On 7/19/13 12:29, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
First of all, a lot of the issues you are talking about are not yet solved.=
<br>
</blockquote>
<br></div>
That was my *point*.<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br></font></span></blockquote><div><br></div><div>And my point is that eac=
h time you solve any of those issue you will break all the applications wor=
king so far. Not a good API design.</div><div>_____________<br>Roman Shpoun=
t</div>
<div>=A0</div></div>

--001a11c355feb738e304e1e0cc64--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 10:39:43 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 10:34 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 12:29, Roman Shpount wrote:
>
>> First of all, a lot of the issues you are talking about are not yet
>> solved.
>>
>
> That was my *point*.
>
>
So, are you trying to say "Look how hard this is to do with SDP.  We're not
even done yet.  It will be even harder without SDP"?


>  /a
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--bcaec520f3b9c820a904e1e0d36e
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:34 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/19/13 12:29, Roman Sh=
pount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
First of all, a lot of the issues you are talking about are not yet solved.=
<br>
</blockquote>
<br></div>
That was my *point*.<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br></font></span></blockquote><div><br></div><div>So, are you trying to sa=
y &quot;Look how hard this is to do with SDP. =C2=A0We&#39;re not even done=
 yet. =C2=A0It will be even harder without SDP&quot;?</div><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">

<span class=3D"HOEnZb"><font color=3D"#888888">
/a</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--bcaec520f3b9c820a904e1e0d36e--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 12:18, Peter Thatcher wrote:
> Please try to consider what you sound like to a web developer who is 
> trying to use WebRTC and finding SDP to be difficult API surface.

I think any time someone says they've used the SDP directly as a control 
surface in a JavaScript application, it indicates a hole in the current 
WebRTC specification that we need to fix.

SDP is not supposed to be the WebRTC API surface.

I think we all pretty much agree that SDP is not supposed to be the 
WebRTC API surface.

I think we're pretty much all on the same page that we need to add stuff 
to the current API to handle these use cases.



> Maybe I'm wrong, but I'm guessing what you're saying will sound like. 
>  "my use case (legacy interop) is the most important and my solution 
> (SDP) is the best solution for everyone and I know better because I've 
> been working on it for longer".  Hubris?

No, you've misunderstood pretty much everything I said. I'll summarize 
in fewer words.

My point is that we need to solve these problems one way or another, and 
SDP isn't really the reason it's difficult.

But by incorrectly deciding that SDP (or offer/answer, depending on who 
you listen to) is the reason it's hard, we're trying to throw legacy 
interop out the window for very little gain. And, while not the only (or 
even main) consideration, legacy interop has significant value. It would 
be a shame to destroy that value based on a misconception.

These issues aren't hard because of SDP. These issues are hard because 
they're hard.

/a

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 12:39, Peter Thatcher wrote:
>
> So, are you trying to say "Look how hard this is to do with SDP. 
>  We're not even done yet.  It will be even harder without SDP"?
>

No. I'm not saying it would be harder. I'm saying it wouldn't be easier, 
and you would lose tangible benefits.

/a

From roman@telurix.com  Fri Jul 19 10:55:28 2013
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 1:49 PM, Adam Roach <adam@nostrum.com> wrote:

>
> These issues aren't hard because of SDP. These issues are hard because
> they're hard.
>
>
I agree that those issues are hard but with SDP they are harder. SDP
requires to create solutions that are backwards compatible within a very
restrictive framework. API allows solutions that are backwards compatible
if they are coded specific way. This is a huge difference.
_____________
Roman Shpount

--001a11c25aa0a7f79404e1e1067c
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<div>On Fri, Jul 19, 2013 at 1:49 PM, Adam Roach <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;<=
/span> wrote:</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">
<div class=3D"im"><br></div>
These issues aren&#39;t hard because of SDP. These issues are hard because =
they&#39;re hard.<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br></font></span></blockquote><div><br></div><div>I agree that those issue=
s are hard but with SDP they are harder. SDP requires to create solutions t=
hat are backwards compatible within a very restrictive framework. API allow=
s solutions that are backwards compatible if they are coded specific way. T=
his is a huge difference.</div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div>

--001a11c25aa0a7f79404e1e1067c--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 19 July 2013 10:39, Peter Thatcher <pthatcher@google.com> wrote:
> So, are you trying to say "Look how hard this is to do with SDP.  We're not
> even done yet.  It will be even harder without SDP"?

In the Atlanta meeting (Nov 2012) I remember waiting for those damned
elevators with Justin.  He said something like: "So, if we'd chosen
comment 22 [CU-RTC-Web] do you think we'd be done by now?"  I was
quick to answer, "Of course not."  After all, we'd only made that
proposal 3-4 months earlier.  I know that nothing gets done in less
than a year, and this is not a small undertaking.

Since then, I've gained a more nuanced view.  We've set an impossibly
high bar for this specification, including all sorts of crazy
features: FEC, simulcast, layered codecs, congestion control,
undeployed codecs, multiplexing, and not to mention a new data
channel.

In reality, SDP never negotiated all that crap before.  Worse, despite
the existence of RFCs for most of these features, it turns out that
most implementations were proprietary.  We couldn't even agree on what
an m-line represents.

So, if asked the same question today, I'd probably have to say: "We'd
at least have had basic scenarios shipped.  We might not have sorted
out the hard cases like layered codecs, but we do have basic
functionality."

What we have this the complete antithesis of any project I've been
involved in over the past 10 years.  It's gold-plating the pink
squirrel.

If we want to ship this thing, then we should be managing scope, not
protecting it.

From matthew.kaufman@skype.net  Fri Jul 19 11:25:32 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Adam Roach <adam@nostrum.com>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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--_2707f48b-d572-4350-8676-6f4f109abc5a_
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Peter Thatcher said:=20

"It's interesting that most of your list of things that needed to be solved=
 without SDP (simulcast=2C FEC=2C correlation of RTP streams with MediaStre=
amTracks=2C glare) still haven't been solved for WebRTC even with SDP=2C de=
spite many months (years?) of effort.  =0A=
=0A=

I don't think anyone is saying "without SDP=2C there would be issues".  I t=
hink they're saying "without SDP=2C the issues are much easier/faster to so=
lve=2C and the API would be much more usable and implementable".   And I do=
n't think that's a naive opinion."

[BA] I would even go further - I'd claim that several of these issues are b=
est solved without SDP=2C either in the API or on the wire.  RTP stacks use=
d in WebRTC need to be able to adapt quickly=2C which implies that the adap=
tations cannot be signaled.  Also=2C if you want to scale=2C then it's impo=
rtant to strip out unnecessary operations that create performance bottlenec=
ks in media-aware network elements (e.g. decrypt/encrypt operations =2C par=
sing of codec-specific parameter sets=2C transcoding=2C etc.). All this poi=
nts to moving functionality out of signaling (and codecs) and into RTP.  If=
 you take this perspective=2C then removing SDP dependencies isn't just a b=
etter plan for getting to "done" - it's an essential part of delivering the=
 next generation realtime architecture.=20


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Peter Thatcher said: <br><div><d=
iv dir=3D"ltr"><div class=3D"ecxgmail_extra"><div class=3D"ecxgmail_quote">=
<div><br></div><div>"It's interesting that most of your list of things that=
 needed to be solved without SDP (simulcast=2C FEC=2C correlation of RTP st=
reams with MediaStreamTracks=2C glare) still haven't been solved for WebRTC=
 even with SDP=2C despite many months (years?) of effort. &nbsp=3B</div>=0A=
=0A=
<div><br></div><div>I don't think anyone is saying "without SDP=2C there wo=
uld be issues". &nbsp=3BI think they're saying "without SDP=2C the issues a=
re much easier/faster to solve=2C and the API would be much more usable and=
 implementable". &nbsp=3B And I don't think that's a naive opinion."<br><br=
>[BA] I would even go further - I'd claim that several of these issues are =
best solved without SDP=2C either in the API or on the wire.&nbsp=3B RTP st=
acks used in WebRTC need to be able to adapt quickly=2C which implies that =
the adaptations cannot be signaled.&nbsp=3B Also=2C if you want to scale=2C=
 then it's important to strip out unnecessary operations that create perfor=
mance bottlenecks in media-aware network elements (e.g. decrypt/encrypt ope=
rations =2C parsing of codec-specific parameter sets=2C transcoding=2C etc.=
). All this points to moving functionality out of signaling (and codecs) an=
d into RTP.&nbsp=3B If you take this perspective=2C then removing SDP depen=
dencies isn't just a better plan for getting to "done" - it's an essential =
part of delivering the next generation realtime architecture. <br><br><br><=
/div></div></div></div></div> 		 	   		  </div></body>
</html>=

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Roman Shpount <roman@telurix.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Date: Fri, 19 Jul 2013 18:27:48 +0000
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--_000_AE1A6B5FD507DC4FB3C5166F3A05A48423716F6BTK5EX14MBXC266r_
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+1 to Roman's comment below. No matter what additional API is produced, as =
long as the offer/answer state machine is there and there is SDP going betw=
een the browsers the JavaScript is either A) not actually in control of wha=
t's happening or B) manipulating a bunch of SDP and running the other side =
of that state machine so as to achieve that control.

There's another way, we wrote it down and published it almost a year ago, a=
nd I still believe that starting with that would get us to a full spec soon=
er than the current path... even more so, now that the last 9 months has ac=
hieved approximately nothing.

Matthew Kaufman

From: Roman Shpount [mailto:roman@telurix.com]


You cannot be more wrong. Each application deals with predictable set of sc=
enarios (ones it produces) and interfaces with predictable set of non webrt=
c devices. These external devices are under control of the application deve=
loper. Application developer is responsible for interop with these devices.=
 What is needed is predictable media and signaling generated by webrtc devi=
ces running the same application. With negotiation and SDP backed into the =
webrtc device it is very hard to achieve. So, for me as webrtc developer le=
t me develop a negotiation scheme that works for me and make sure it stay w=
orking regardless of the browser type or version. Right now controlling web=
rtc application feels like driving the boat down the river by tickling the =
captain.
_____________
Roman Shpount


--_000_AE1A6B5FD507DC4FB3C5166F3A05A48423716F6BTK5EX14MBXC266r_
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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D">&#43;1 to Roman&#8217;s comment below. No matter what additional API is =
produced, as long as the offer/answer state machine is there and there
 is SDP going between the browsers the JavaScript is either A) not actually=
 in control of what&#8217;s happening or B) manipulating a bunch of SDP and=
 running the other side of that state machine so as to achieve that control=
.
<o:p></o:p></span></a></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">There&#8217;s another way=
, we wrote it down and published it almost a year ago, and I still believe =
that starting with that would get us to a full spec sooner than
 the current path&#8230; even more so, now that the last 9 months has achie=
ved approximately nothing.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Matthew Kaufman<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
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size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</=
span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;"> Roman Shpount [mailto:roman@telurix.com]
<br>
<br>
</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">You cannot be more wrong. Each application deals wit=
h predictable set of scenarios (ones it produces) and interfaces with predi=
ctable set of non webrtc devices. These external devices are under control =
of the application developer. Application
 developer is responsible for interop with these devices. What is needed is=
 predictable media and signaling generated by webrtc devices running the sa=
me application. With negotiation and SDP backed into the webrtc device it i=
s very hard to achieve. So, for
 me as webrtc developer let me develop a negotiation scheme that works for =
me and make sure it stay working regardless of the browser type or version.=
 Right now controlling webrtc application feels like driving the boat down =
the river by tickling the captain.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_AE1A6B5FD507DC4FB3C5166F3A05A48423716F6BTK5EX14MBXC266r_--

From matthew.kaufman@skype.net  Fri Jul 19 11:38:38 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Adam Roach <adam@nostrum.com>, =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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References: <CAJrXDUGMohpBdi-ft-o_uE7ewFkw7wRY9x7gYEncjov7qi-Bew@mail.gmail.com> <CABcZeBPa4wBS8pYq=0wesMOfL6TkeC7QGAZ8pWwOcnkhkJqWfA@mail.gmail.com> <CAJrXDUFxo8P8wxh8jX3019yPQOuwQ0eVdsFmRXsbWdWinnc5oA@mail.gmail.com> <CABcZeBOTKpmFC34waqZ4kA-P8t+E6yY9gX1JFCHhsBH0+CF-Qw@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30BC0F@ESESSMB209.ericsson.se> <CAD5OKxtKLMf_d=8GSMrqfNhDHPe9MFP2ZTKzZHFn9CyMr-gSVQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30C833@ESESSMB209.ericsson.se> <CAD5OKxvGfkgRp6tXwbOu_kVteHiBBqsyR5ixH18FMKjCNGO8VQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30CD1E@ESESSMB209.ericsson.se> <BLU401-EAS386F88B3FE140492B39B59693610@phx.gbl> <AE1A6B5FD507DC4FB3C5166F3A05A484213E41E7@TK5EX14MBXC265.redmond.corp.microsoft.com> <C50FDAD5-492C-4A83-AD6D-464242FB4A05@iii.ca> <CALiegfneUj=kzDjR_E1=S-bqAajaPUE3f_A2g8oGriFyPhamPA@mail.gmail.com> <51E96B5B.2050302@nostrum.com>
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Martin Thomson <martin.thomson@gmail.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Ted Hardie <ted.ietf@gmail.com>, Martin Thomson <martin.thomson@gmail.com>
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References: <CAJrXDUGMohpBdi-ft-o_uE7ewFkw7wRY9x7gYEncjov7qi-Bew@mail.gmail.com> <CABcZeBPa4wBS8pYq=0wesMOfL6TkeC7QGAZ8pWwOcnkhkJqWfA@mail.gmail.com> <CAJrXDUFxo8P8wxh8jX3019yPQOuwQ0eVdsFmRXsbWdWinnc5oA@mail.gmail.com> <CABcZeBOTKpmFC34waqZ4kA-P8t+E6yY9gX1JFCHhsBH0+CF-Qw@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30BC0F@ESESSMB209.ericsson.se> <CAD5OKxtKLMf_d=8GSMrqfNhDHPe9MFP2ZTKzZHFn9CyMr-gSVQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30C833@ESESSMB209.ericsson.se> <CAD5OKxvGfkgRp6tXwbOu_kVteHiBBqsyR5ixH18FMKjCNGO8VQ@mail.gmail.com> <1447FA0C20ED5147A1AA0EF02890A64B1C30CD1E@ESESSMB209.ericsson.se> <BLU401-EAS386F88B3FE140492B39B59693610@phx.gbl> <AE1A6B5FD507DC4FB3C5166F3A05A484213E41E7@TK5EX14MBXC265.redmond.corp.microsoft.com> <C50FDAD5-492C-4A83-AD6D-464242FB4A05@iii.ca> <CALiegfneUj=kzDjR_E1=S-bqAajaPUE3f_A2g8oGriFyPhamPA@mail.gmail.com> <51E96B5B.2050302@nostrum.com> <CABkgnnXa-eTzRHcLMnHam4c+1D9kkvRwi9=V-9P43+p+pKE_sw@mail.gmail.com> <CA+9kkMCjt2UHFynLqwns0J0f5ZtnxtMX3ppzR66e5q_rJ9D5Ug@mail.gmail.com>
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Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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So we should have APIs that let JavaScript applications adapt to these diff=
erent environments (like we do for everything else in "HTML5"), *not* try t=
o solve this by making the API be "tickling the ship captain" and attemptin=
g to do everything else in a pairwise negotiation between two elements, one=
 of which might not even be a browser.

Matthew Kaufman

From: Ted Hardie [mailto:ted.ietf@gmail.com]
Sent: Friday, July 19, 2013 10:07 AM
To: Martin Thomson
Cc: Adam Roach; I=F1aki Baz Castillo; Cullen Jennings; Matthew Kaufman (SKY=
PE); <rtcweb@ietf.org>; public-webrtc@w3.org
Subject: Re: On babies and bathwater (was Re: [rtcweb] Summary of Applicati=
on Developers' opinions of the current WebRTC API and SDP as a control surf=
ace)

On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson <martin.thomson@gmail.com<m=
ailto:martin.thomson@gmail.com>> wrote:

Negotiation is a hole.  A vast, soul-sucking, waste of time.

Even if you have the same javascript application downloaded, you will have =
disparate capabilities in the environments into which it is downloaded (bro=
wser/os/codecs/media sources/available network capacity).  Getting set inte=
rsection and preference order for those capabilities is something that appl=
ications actually want.  You may be able to move the pain of that around, b=
ut it isn't a waste of time.

My personal opinion, with no IETF hats or company swag,

Ted

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<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D">So we should have APIs that let JavaScript applications adapt to these d=
ifferent environments (like we do for everything else in &#8220;HTML5&#8221=
;),
 *<b>not</b>* try to solve this by making the API be &#8220;tickling the sh=
ip captain&#8221; and attempting to do everything else in a pairwise negoti=
ation between two elements, one of which might not even be a browser.<o:p><=
/o:p></span></a></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Matthew Kaufman<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
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0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"> Ted Ha=
rdie [mailto:ted.ietf@gmail.com]
<br>
<b>Sent:</b> Friday, July 19, 2013 10:07 AM<br>
<b>To:</b> Martin Thomson<br>
<b>Cc:</b> Adam Roach; I=F1aki Baz Castillo; Cullen Jennings; Matthew Kaufm=
an (SKYPE); &lt;rtcweb@ietf.org&gt;; public-webrtc@w3.org<br>
<b>Subject:</b> Re: On babies and bathwater (was Re: [rtcweb] Summary of Ap=
plication Developers' opinions of the current WebRTC API and SDP as a contr=
ol surface)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">On Fri, Jul 19, 2013 at 9:54 AM, Martin Thomson &lt;=
<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomso=
n@gmail.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Negotiation is a hole. &nbsp;A vast, soul-sucking, waste of time.<o:p></o:p=
></p>
</blockquote>
</div>
<p class=3D"MsoNormal"><br>
Even if you have the same javascript application downloaded, you will have =
disparate capabilities in the environments into which it is downloaded (bro=
wser/os/codecs/media sources/available network capacity).&nbsp; Getting set=
 intersection and preference order for
 those capabilities is something that applications actually want.&nbsp; You=
 may be able to move the pain of that around, but it isn't a waste of time.=
<br>
<br>
My personal opinion, with no IETF hats or company swag,<br>
<br>
Ted<o:p></o:p></p>
</div>
</div>
</body>
</html>

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 11:47:10 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 10:49 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 12:18, Peter Thatcher wrote:
>
>> Please try to consider what you sound like to a web developer who is
>> trying to use WebRTC and finding SDP to be difficult API surface.
>>
>
> I think any time someone says they've used the SDP directly as a control
> surface in a JavaScript application, it indicates a hole in the current
> WebRTC specification that we need to fix.
>
> SDP is not supposed to be the WebRTC API surface.
>
> I think we all pretty much agree that SDP is not supposed to be the WebRTC
> API surface.
>
> I think we're pretty much all on the same page that we need to add stuff
> to the current API to handle these use cases.


If SDP isn't the control surface, than what is the API surface?  The only
way to tell the browser to setup an ICE connection is to give it SDP.  The
only way to tell it to send and receive RTP packets is to give it SDP.  The
browser might produce the SDP for you, and you might be told not to change
it, but at the end of the day, the commands that make the browser setup ICE
connections and send and receive RTP is the SDP.   Therefore, SDP is the
API control surface.  Until JS can setup an ICE connection and send and
receive RTP packets without giving the browser SDP, then SDP is the control
surface.

This might not be so bad for applications that use SDP for signalling, but
any app that uses something other than SDP for signalling necessarily views
SDP as the API surface, since there is no other way to tell the browser
what to do.





>
>
>
>  Maybe I'm wrong, but I'm guessing what you're saying will sound like.
>>  "my use case (legacy interop) is the most important and my solution (SDP)
>> is the best solution for everyone and I know better because I've been
>> working on it for longer".  Hubris?
>>
>
> No, you've misunderstood pretty much everything I said. I'll summarize in
> fewer words.
>
>
You called everyone who doubts the SDP API naive and full of hubris, and
now you're telling me I'm misunderstanding pretty much everything.


> My point is that we need to solve these problems one way or another, and
> SDP isn't really the reason it's difficult.
>

One problem is that this is a very difficult to use API for web developers.
 And SDP is at the heart of that problem.  Lots of people have said so, and
why, in great detail.  Do you really believe all of them are naive and full
of hubris?


>
> But by incorrectly deciding that SDP (or offer/answer, depending on who
> you listen to) is the reason it's hard, we're trying to throw legacy
> interop out the window for very little gain.


I think this is the real issue at hand:  You value legacy interop more than
a usable API.  Other value a usable API more than legacy interop.  You're
willing to sacrifice API usability for legacy interop, and they're willing
to sacrifice legacy interop for a usable API.  There are different groups
with different needs and different perspectives.

I think we can have both legacy interop and a usable API, and there are
many ways we can achieve that.  But we've been asked to delay talking about
it until "1.0" is done.  So, we'll have to wait for further discussion.




> And, while not the only (or even main) consideration, legacy interop has
> significant value. It would be a shame to destroy that value based on a
> misconception.
>
> These issues aren't hard because of SDP. These issues are hard because
> they're hard.


SDP as an API surface makes the API worse.  And API surface that didn't
require SDP would be a better API surface.  It's a fallacy to assume that
because we need legacy interop, we have to support it to the detriment of
of non-legacy interop use cases.  We could have both, if we wanted to
design an API for both.

Please try to realize that there are worlds of use cases beyond yours, and
there are lots of people with different needs than yours.  And please try
to listen to them rather than just calling them naive and full of hubris.


>
>
> /a
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:49 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/19/13 12:18, Peter Thatcher wrote:<=
br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Please try to consider what you sound like to a web developer who is trying=
 to use WebRTC and finding SDP to be difficult API surface.<br>
</blockquote>
<br></div>
I think any time someone says they&#39;ve used the SDP directly as a contro=
l surface in a JavaScript application, it indicates a hole in the current W=
ebRTC specification that we need to fix.<br>
<br>
SDP is not supposed to be the WebRTC API surface.<br>
<br>
I think we all pretty much agree that SDP is not supposed to be the WebRTC =
API surface.<br>
<br>
I think we&#39;re pretty much all on the same page that we need to add stuf=
f to the current API to handle these use cases.</blockquote><div><br></div>=
<div>If SDP isn&#39;t the control surface, than what is the API surface? =
=C2=A0The only way to tell the browser to setup an ICE connection is to giv=
e it SDP. =C2=A0The only way to tell it to send and receive RTP packets is =
to give it SDP. =C2=A0The browser might produce the SDP for you, and you mi=
ght be told not to change it, but at the end of the day, the commands that =
make the browser setup ICE connections and send and receive RTP is the SDP.=
 =C2=A0 Therefore, SDP is the API control surface. =C2=A0Until JS can setup=
 an ICE connection and send and receive RTP packets without giving the brow=
ser SDP, then SDP is the control surface. =C2=A0</div>

<div><br></div><div>This might not be so bad for applications that use SDP =
for signalling, but any app that uses something other than SDP for signalli=
ng necessarily views SDP as the API surface, since there is no other way to=
 tell the browser what to do.</div>

<div><br></div><div><br></div><div><br></div><div><br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
">

<div class=3D"im"><br>
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Maybe I&#39;m wrong, but I&#39;m guessing what you&#39;re saying will sound=
 like. =C2=A0&quot;my use case (legacy interop) is the most important and m=
y solution (SDP) is the best solution for everyone and I know better becaus=
e I&#39;ve been working on it for longer&quot;. =C2=A0Hubris?<br>


</blockquote>
<br></div>
No, you&#39;ve misunderstood pretty much everything I said. I&#39;ll summar=
ize in fewer words.<br>
<br></blockquote><div><br></div><div>You called everyone who doubts the SDP=
 API naive and full of hubris, and now you&#39;re telling me I&#39;m misund=
erstanding pretty much everything.</div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


My point is that we need to solve these problems one way or another, and SD=
P isn&#39;t really the reason it&#39;s difficult.<br></blockquote><div><br>=
</div><div>One problem is that this is a very difficult to use API for web =
developers. =C2=A0And SDP is at the heart of that problem. =C2=A0Lots of pe=
ople have said so, and why, in great detail. =C2=A0Do you really believe al=
l of them are naive and full of hubris?</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex"><br>
But by incorrectly deciding that SDP (or offer/answer, depending on who you=
 listen to) is the reason it&#39;s hard, we&#39;re trying to throw legacy i=
nterop out the window for very little gain.</blockquote><div><br></div>

<div>I think this is the real issue at hand: =C2=A0You value legacy interop=
 more than a usable API. =C2=A0Other value a usable API more than legacy in=
terop. =C2=A0You&#39;re willing to sacrifice API usability for legacy inter=
op, and they&#39;re willing to sacrifice legacy interop for a usable API. =
=C2=A0There are different groups with different needs and different perspec=
tives. =C2=A0</div>

<div><br></div><div>I think we can have both legacy interop and a usable AP=
I, and there are many ways we can achieve that. =C2=A0But we&#39;ve been as=
ked to delay talking about it until &quot;1.0&quot; is done. =C2=A0So, we&#=
39;ll have to wait for further discussion. =C2=A0</div>

<div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex"> And, while=
 not the only (or even main) consideration, legacy interop has significant =
value. It would be a shame to destroy that value based on a misconception.<=
br>


<br></blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">These issues aren&#39;t hard because of S=
DP. These issues are hard because they&#39;re hard.</blockquote>

<div><br></div><div>SDP as an API surface makes the API worse. =C2=A0And AP=
I surface that didn&#39;t require SDP would be a better API surface. =C2=A0=
It&#39;s a fallacy to assume that because we need legacy interop, we have t=
o support it to the detriment of of non-legacy interop use cases. =C2=A0We =
could have both, if we wanted to design an API for both.</div>

<div><br></div><div>Please try to realize that there are worlds of use case=
s beyond yours, and there are lots of people with different needs than your=
s. =C2=A0And please try to listen to them rather than just calling them nai=
ve and full of hubris.</div>

<div><div>=C2=A0</div></div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,20=
4);border-left-style:solid;padding-left:1ex"><span class=3D""><font color=
=3D"#888888"><br>


<br>
/a<br>
</font></span></blockquote></div><br></div></div>

--047d7b6705b9953e7d04e1e1c354--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 13:47, Peter Thatcher wrote:
>  Do you really believe all of them are naive and full of hubris?

No, I think they're using an API that is nowhere near finished yet, 
which is necessarily going to be painful. The complaints are legitimate.

The scapegoating of SDP is what's misguided.

/a

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--e89a8ff24881eceb3804e1e1e532
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On Fri, Jul 19, 2013 at 10:51 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 12:39, Peter Thatcher wrote:
>
>>
>> So, are you trying to say "Look how hard this is to do with SDP.  We're
>> not even done yet.  It will be even harder without SDP"?
>>
>>
> No. I'm not saying it would be harder. I'm saying it wouldn't be easier,



It wouldn't be harder, and it wouldn't be easier.  So, it would be about
the same amount of work for either?



> and you would lose tangible benefits.


I still don't understand how this could be true.  If we define an API that
allows JS to use either SDP or not, at its choice, how is that a loss?  It
seems like only a gain.






> /a
>

--e89a8ff24881eceb3804e1e1e532
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:51 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div>On 7/19/13 12:39, Peter Thatcher wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
So, are you trying to say &quot;Look how hard this is to do with SDP. =C2=
=A0We&#39;re not even done yet. =C2=A0It will be even harder without SDP&qu=
ot;?<br>
<br>
</blockquote>
<br></div>
No. I&#39;m not saying it would be harder. I&#39;m saying it wouldn&#39;t b=
e easier,</blockquote><div><br></div><div><div><br>It wouldn&#39;t be harde=
r, and it wouldn&#39;t be easier. =C2=A0So, it would be about the same amou=
nt of work for either?=C2=A0</div>



</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(=
204,204,204);border-left-style:solid;padding-left:1ex"> and you would lose =
tangible benefits.</blockquote>



<div><br></div><div>I still don&#39;t understand how this could be true. =
=C2=A0If we define an API that allows JS to use either SDP or not, at its c=
hoice, how is that a loss? =C2=A0It seems like only a gain.</div><div><br><=
/div><div>


<br></div><div>=C2=A0</div><div><br></div><div>=C2=A0</div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
><span><font color=3D"#888888">
/a<br>
</font></span></blockquote></div><br></div></div>

--e89a8ff24881eceb3804e1e1e532--

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Adam Roach <adam@nostrum.com>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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--001a11c39d18ca604604e1e20115
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 19, 2013 at 10:57 AM, Martin Thomson
<martin.thomson@gmail.com>wrote:

> On 19 July 2013 10:39, Peter Thatcher <pthatcher@google.com> wrote:
> > So, are you trying to say "Look how hard this is to do with SDP.  We're
> not
> > even done yet.  It will be even harder without SDP"?
>
> In the Atlanta meeting (Nov 2012) I remember waiting for those damned
> elevators with Justin.  He said something like: "So, if we'd chosen
> comment 22 [CU-RTC-Web] do you think we'd be done by now?"  I was
> quick to answer, "Of course not."  After all, we'd only made that
> proposal 3-4 months earlier.  I know that nothing gets done in less
> than a year, and this is not a small undertaking.
>
> Since then, I've gained a more nuanced view.  We've set an impossibly
> high bar for this specification, including all sorts of crazy
> features: FEC, simulcast, layered codecs, congestion control,
> undeployed codecs, multiplexing, and not to mention a new data
> channel.
>
> In reality, SDP never negotiated all that crap before.  Worse, despite
> the existence of RFCs for most of these features, it turns out that
> most implementations were proprietary.  We couldn't even agree on what
> an m-line represents.
>
> So, if asked the same question today, I'd probably have to say: "We'd
> at least have had basic scenarios shipped.  We might not have sorted
> out the hard cases like layered codecs, but we do have basic
> functionality."
>
> What we have this the complete antithesis of any project I've been
> involved in over the past 10 years.  It's gold-plating the pink
> squirrel.
>

> If we want to ship this thing, then we should be managing scope, not
> protecting it.
>

Slight aside, but is this a correct model of what has happened with regards
to scope?

The W3C is waiting for the IETF RTCWEB WG.  The RTCWEB WG is waiting for
the MMUSIC WG.   The MMUSIC WG has years worth of things that aren't well
defined in SDP that it's now taking the opportunity to define (a sort of
"RAI 2.0").  And everything back to the W3C is blocked by these things that
aren't well defined in SDP, meaning WebRTC isn't done until MMUSIC defines
all these things in SDP that have been piling up for years.

Is that accurate?  If so, I agree that "scope management" would be a good
thing to consider.

--001a11c39d18ca604604e1e20115
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 10:57 AM, Martin Thomson <span dir=3D"ltr">=
&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.th=
omson@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 19 July 2013 10:39, Pet=
er Thatcher &lt;<a href=3D"mailto:pthatcher@google.com">pthatcher@google.co=
m</a>&gt; wrote:<br>


&gt; So, are you trying to say &quot;Look how hard this is to do with SDP. =
=C2=A0We&#39;re not<br>
&gt; even done yet. =C2=A0It will be even harder without SDP&quot;?<br>
<br>
</div>In the Atlanta meeting (Nov 2012) I remember waiting for those damned=
<br>
elevators with Justin. =C2=A0He said something like: &quot;So, if we&#39;d =
chosen<br>
comment 22 [CU-RTC-Web] do you think we&#39;d be done by now?&quot; =C2=A0I=
 was<br>
quick to answer, &quot;Of course not.&quot; =C2=A0After all, we&#39;d only =
made that<br>
proposal 3-4 months earlier. =C2=A0I know that nothing gets done in less<br=
>
than a year, and this is not a small undertaking.<br>
<br>
Since then, I&#39;ve gained a more nuanced view. =C2=A0We&#39;ve set an imp=
ossibly<br>
high bar for this specification, including all sorts of crazy<br>
features: FEC, simulcast, layered codecs, congestion control,<br>
undeployed codecs, multiplexing, and not to mention a new data<br>
channel.<br>
<br>
In reality, SDP never negotiated all that crap before. =C2=A0Worse, despite=
<br>
the existence of RFCs for most of these features, it turns out that<br>
most implementations were proprietary. =C2=A0We couldn&#39;t even agree on =
what<br>
an m-line represents.<br>
<br>
So, if asked the same question today, I&#39;d probably have to say: &quot;W=
e&#39;d<br>
at least have had basic scenarios shipped. =C2=A0We might not have sorted<b=
r>
out the hard cases like layered codecs, but we do have basic<br>
functionality.&quot;<br>
<br>
What we have this the complete antithesis of any project I&#39;ve been<br>
involved in over the past 10 years. =C2=A0It&#39;s gold-plating the pink<br=
>
squirrel.<br></blockquote><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
If we want to ship this thing, then we should be managing scope, not<br>
protecting it.<br>
</blockquote><div><br></div><div>Slight aside, but is this a correct model =
of what has happened with regards to scope?</div><div><br></div><div>The W3=
C is waiting for the IETF RTCWEB WG. =C2=A0The RTCWEB WG is waiting for the=
 MMUSIC WG. =C2=A0 The MMUSIC WG has years worth of things that aren&#39;t =
well defined in SDP that it&#39;s now taking the opportunity to define (a s=
ort of &quot;RAI 2.0&quot;). =C2=A0And everything back to the W3C is blocke=
d by these things that aren&#39;t well defined in SDP, meaning WebRTC isn&#=
39;t done until MMUSIC defines all these things in SDP that have been pilin=
g up for years.=C2=A0</div>

<div><br></div><div>Is that accurate? =C2=A0If so, I agree that &quot;scope=
 management&quot; would be a good thing to consider.</div></div></div></div=
>

--001a11c39d18ca604604e1e20115--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On 7/19/13 13:47, Peter Thatcher wrote:
> I think this is the real issue at hand:  You value legacy interop more 
> than a usable API.

This. *This* is why I've told you that you're misunderstanding 
everything. Don't take offense, just go back and read more carefully. I 
never said anything that implied that legacy interop is more valuable 
than a usable API. It's a nice strawman for you to build up and tear 
down, but you are arguing with a fictional character who is not me when 
you do so.

What I'm trying to point out is that these goals are not at odds with 
each other. Your statement above implies that you have taken it as given 
that we can't do both -- that there is a tradeoff here to be made. If 
you take that as a fundamental principle, then I can see how nothing I 
say makes any sense.

But they're not mutually exclusive goals. Keep that in mind, and go back 
to re-read what I've written.

/a

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 12:13:29 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 11:51 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 13:47, Peter Thatcher wrote:
>
>>  Do you really believe all of them are naive and full of hubris?
>>
>
> No, I think they're using an API that is nowhere near finished yet, which
> is necessarily going to be painful. The complaints are legitimate.
>
>
"nowhere near finished yet"?  Yikes.  We just spent almost a year arguing
about what an m-line is.  What's your timeline for finishing?

The scapegoating of SDP is what's misguided.


Well, I guess we're going to have to agree to disagree, because from my
perspective as a web developer and a browser implementer, the API being
based on SDP Offer/Answer is the heart of the problem, and not just a
scapegoat.  I hope that we can do much better in a 2.0 API while
maintaining legacy interop (make everyone happy), but of course I'll
refrain from talking about that further until the right time has come.



>
>
> /a
>

--e89a8ff2431996c5e104e1e221ce
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 11:51 AM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/19/13 13:47, Peter Thatcher wrote:<=
br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
=C2=A0Do you really believe all of them are naive and full of hubris?<br>
</blockquote>
<br></div>
No, I think they&#39;re using an API that is nowhere near finished yet, whi=
ch is necessarily going to be painful. The complaints are legitimate.<br>
<br></blockquote><div><br></div><div>&quot;nowhere near finished yet&quot;?=
 =C2=A0Yikes. =C2=A0We just spent almost a year arguing about what an m-lin=
e is. =C2=A0What&#39;s your timeline for finishing? =C2=A0=C2=A0</div><div>=
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex">


The scapegoating of SDP is what&#39;s misguided.</blockquote><div><br></div=
><div>Well, I guess we&#39;re going to have to agree to disagree, because f=
rom my perspective as a web developer and a browser implementer, the API be=
ing based on SDP Offer/Answer is the heart of the problem, and not just a s=
capegoat. =C2=A0I hope that we can do much better in a 2.0 API while mainta=
ining legacy interop (make everyone happy), but of course I&#39;ll refrain =
from talking about that further until the right time has come.=C2=A0</div>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex"><span class=3D""><font col=
or=3D"#888888"><br>


<br>
/a<br>
</font></span></blockquote></div><br></div></div>

--e89a8ff2431996c5e104e1e221ce--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 12:42:38 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 12:06 PM, Adam Roach <adam@nostrum.com> wrote:

> On 7/19/13 13:47, Peter Thatcher wrote:
>
>> I think this is the real issue at hand:  You value legacy interop more
>> than a usable API.
>>
>
> This. *This* is why I've told you that you're misunderstanding everything.
> Don't take offense, just go back and read more carefully. I never said
> anything that implied that legacy interop is more valuable than a usable
> API. It's a nice strawman for you to build up and tear down, but you are
> arguing with a fictional character who is not me when you do so.
>
> What I'm trying to point out is that these goals are not at odds with each
> other. Your statement above implies that you have taken it as given that we
> can't do both -- that there is a tradeoff here to be made.

If you take that as a fundamental principle, then I can see how nothing I
> say makes any sense.
>
> But they're not mutually exclusive goals. Keep that in mind, and go back
> to re-read what I've written.
>
>
You are claiming that there is no tradeoff between API usability and legacy
interop.  If that is the case, then why have we designed such a terribly
unusable API?  If we can have an API with both legacy interop and
usability, then why didn't we create that API?  If we can make an API that
serves both sides (roughly, "web developers" and "legacy interop") well,
why have we picked an API that only serves one well?

I do believe that we can have an API that provides both good usability and
legacy interop, and that can server both sides (and even more sides) well.
 But is most assuredly not the API we've designed so far, and it's not the
API we're headed toward (at least not with "1.0").  I have hope for the 2.0
API, but let's not kid ourselves.  The 1.0 API picked a side and ran with
it.



> /a
>

--001a11330f2aded74204e1e28938
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 12:06 PM, Adam Roach <span dir=3D"ltr">&lt;=
<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&=
gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 7/19/13 13:47, Peter Thatcher wrote:<=
br>


</div><div class=3D"im"><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);b=
order-left-style:solid;padding-left:1ex">
I think this is the real issue at hand: =C2=A0You value legacy interop more=
 than a usable API.<br>
</blockquote>
<br></div>
This. *This* is why I&#39;ve told you that you&#39;re misunderstanding ever=
ything. Don&#39;t take offense, just go back and read more carefully. I nev=
er said anything that implied that legacy interop is more valuable than a u=
sable API. It&#39;s a nice strawman for you to build up and tear down, but =
you are arguing with a fictional character who is not me when you do so.<br=
>


<br></blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">What I&#39;m trying to point out is that =
these goals are not at odds with each other. Your statement above implies t=
hat you have taken it as given that we can&#39;t do both -- that there is a=
 tradeoff here to be made. </blockquote>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">If you take that as a fundamental principle, then I can se=
e how nothing I say makes any sense.<br>


<br>
But they&#39;re not mutually exclusive goals. Keep that in mind, and go bac=
k to re-read what I&#39;ve written.<span class=3D""><font color=3D"#888888"=
><br>
<br></font></span></blockquote><div><br></div><div>You are claiming that th=
ere is no tradeoff between API usability and legacy interop. =C2=A0If that =
is the case, then why have we designed such a terribly unusable API? =C2=A0=
If we can have an API with both legacy interop and usability, then why didn=
&#39;t we create that API? =C2=A0If we can make an API that serves both sid=
es (roughly, &quot;web developers&quot; and &quot;legacy interop&quot;) wel=
l, why have we picked an API that only serves one well?</div>

<div><br></div><div>I do believe that we can have an API that provides both=
 good usability and legacy interop, and that can server both sides (and eve=
n more sides) well. =C2=A0But is most assuredly not the API we&#39;ve desig=
ned so far, and it&#39;s not the API we&#39;re headed toward (at least not =
with &quot;1.0&quot;). =C2=A0I have hope for the 2.0 API, but let&#39;s not=
 kid ourselves. =C2=A0The 1.0 API picked a side and ran with it.</div>

<div><div>=C2=A0</div></div><div>=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span class=
=3D""><font color=3D"#888888">
/a<br>
</font></span></blockquote></div><br></div></div>

--001a11330f2aded74204e1e28938--

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Hadriel Kaplan'" <hadriel.kaplan@oracle.com>, <rtcweb@ietf.org>
References: <CA+9kkMBuCTdFsUMtmuBz6BnrSJMpHywEZU+x+m8ARnGprvzDzA@mail.gmail.com>	<CABkgnnXOa44ZkZj-g6r7Qdk8dwm6m81yT4U=Q23-hE1Q7Hn22w@mail.gmail.com>	<F9556428-B6B8-407D-9D62-9A1CC04D4253@oracle.com> <B2DF729D-B313-4D3E-9C06-DA00AF7A14FF@oracle.com>
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Date: Sat, 20 Jul 2013 01:22:52 +0530
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Subject: Re: [rtcweb] A compromise for SDES
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Hi Hadriel & all,

I agree with Hadriel that DTLS-SRTP as MTI has market benefit but still I'm
not seeing much difference in terms of end-user security. I was supporting
DTLS-SRTP earlier as MTI as WebRTC signaling may takes place in HTTP. In
reality, HTTPS is expected by webrtc operator as they are worried about Web
signaling Security (on-path attack). In case of HTTPS based WebRTC signaling
traffic is used, I'm not seeing any difference between DTLS-SRTP & SDES
because it is possible to intercept the traffic by web provider (signaling
server) in case the web provider is intended to do so as mentioned in Sec
5.7.1. of draft-ietf-rtcweb-security-arch-07. Dr.Evil website shall design
as follows:

     -----Web Server & IdP-----
    |            |             |
Browser------B2BMGW------Browser

Back to back media gateway (B2BMGW) has the capability to terminate and
originate DTLS-SRTP webrtc media traffic without informing browser. Please
look into http://www.webrtc2sip.org/ for how easy to design such B2BMGW. 

The identity is not mandatory in WebRTC. Webrtc session request using HTTPS
URL like https://apprtc.appspot.com/?r=80112689 works well and it has good
web application like conference link. So, there is no much difference
between DTLS-SRTP & SDES-SRTP in case of no identity scenario. Even if the
identity is mandated using Idp, Dr.Evil provider shall mandate authoritative
Idp as (@drevil.com) which helps to intercept the media in B2BMGW. I could
not understand why Dr.Evil web provider will provide the mechanism for
establishing webrtc session using third party identity provider. As of now,
I'm not very clear how webrtc security trust model with DTLS-SRTP really
benefit browser user in terms of security whereas SDES-SRTP compromise the
security.

I see the difference between SDES-SRTP & DTLS-SRTP in case HTTP origin. So
apart of Hadriel compromise plan, some of the SDES compromise plan could be 

1) forbid HTTP origin when SDES is requested by API
2) forbid HTTP origin entirely and allow SDES-SRTP as one of the security
key mechanism. Let application selects its preferred security key mechanism.

Thanks
Partha

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Saturday, July 13, 2013 11:37 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] A compromise for SDES
> 
> Howdy,
> Someone's asked me to explain the compromise I plan on presenting in
> Berlin now, so we have more time to argue over it or chew on it.  The
> reason I was going to wait is because I wanted to explain how I got to
> the point of thinking such a compromise would make sense.  So I'll try
> to do that in this email, which will make this email long.  Nothin'
> much I can do about that.
> 
> Obviously this is all my personal opinion, not that of my employer, and
> not a statement claiming to be objective fact, etc.
> 
> Background:
> I think many people would agree that SDES has some benefits compared to
> DTLS for those of us wishing to interface to the SIP world.  Some of
> those benefits are commercial/cost factors, some are more tangible like
> reducing early media clipping.  Whether you agree with such benefits or
> not, so long as WebRTC could support SDES in a sufficiently-secure
> manner, there'd be less concern over having it.  Most of the arguments
> against SDES have been directed at whether it could in fact be
> "sufficiently secure".  Personally I find most of the arguments against
> it being secure enough to be essentially FUD, and/or also apply to
> DTLS-SRTP.
> 
> On the flip-side, I think the commercial/cost factor for SDES is not a
> strong cause, because I believe the market will bear whatever the extra
> cost may be. (I wasn't sure of this 2 years ago when this all started,
> but I'm fairly confident of it now)  And I think having only DTLS-SRTP
> as MTI would have a marketing benefit for WebRTC as a technology, which
> I value highly.
> 
> Given those personal feelings, I didn't much care either way, so long
> as a decision was made - although I still preferred having SDES to
> avoid the early media clipping problem.  So when I was going to present
> on this topic back in the Vancouver meeting, my last slide basically
> said: "If we support SDES and it turns out later we were wrong, we can
> always rescind it".  This was based on the notion that browsers get
> updated all the time, especially for security issues.
> 
> But later it occurred to me that's actually the Achilles' heel of
> supporting SDES in WebRTC, for those of us wanting to gateway this
> stuff to SIP.  Imagine if it turns out we were wrong, and some
> expected-or-unexpected vulnerability is exploited against some large
> WebRTC service provider, and makes the news/slashdot.  Browser vendors
> would instantly have new code removing SDES support, and browsers in
> devices would be updated virtually overnight - some users may take a
> long time to upgrade their specific browsers on their specific devices
> - but many of them would do so within days.  That would be untenable
> for a WebRTC service provider.  Even the smaller Enterprises take a
> long time to upgrade code on their servers, so even for them changing
> their gateways to suddenly support DTLS-SRTP would be unrealistic.
> Imagine what it would be for larger providers, who might even have to
> deploy more servers to handle the sudden additional overhead of DTLS-
> SRTP.  Meanwhile a growing perc
>  entage of their users can no longer use the service.  That's bad mojo.
> 
> The things that a WebRTC-to-SIP service provider can control, and
> arguably the much larger use-case for this stuff to begin with, are not
> really true "browsers" anyway - they're web-based-framework device
> Apps, provided by the service provider to begin with.  I.e., the apps
> they build using web application framework things like
> PhoneGap/Cordova, Appcelerator Titanium, etc.  I mean using a real
> Browser is nice for ad-hoc stuff, like being able to click on a "talk
> to a rep" button on a website, or a webex/meetecho type thing; but for
> real communication "service", whether it be as a subscriber of a
> traditional carrier, or Skype, or an employee of an Enterprise, or a
> call-center attendant, or whatever - for those most people would never
> want to have to open a browser just to receive/make calls.  They'd give
> you an app to use instead.
> 
> So it's the app use-case that would benefit the most from SDES,
> especially for issues like media clipping.  And the app use-case
> doesn't have the security concerns nor concerns for unforeseen
> overnight updates.
> 
> The Compromise:
> So given that background, I was planning to propose that the security
> doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a
> statement that web-based application frameworks SHOULD also support
> SDES. (with text about why and how, etc.)
> 
> I don't think this is too controversial, because web-based frameworks
> are never beholden to following browser behavior anyway - they're used
> to build a native application, and native applications have a very
> different security/threat model in practice.  They're written for a
> specific purpose, and installed by users from known sources for that
> known purpose.  Technically, afaik, nothing we do in RTCWEB WG or W3C's
> WEBRTC group have any requirements/mandates for native applications
> anyway - an app maker would just ignore something they don't think
> applies to them - but I think web-based frameworks do generally try to
> follow W3C models for Javascript APIs, and will likely read the IETF
> RFCs for the media-layer stuff too.  So I think having this SHOULD
> statement would be beneficial.
> 
> Or if the WG prefers, we could even write a separate doc on what a web-
> based framework maker should consider supporting/not-supporting. (I can
> imagine a few other things they might want to offer that a browser
> can't)
> 
> -hadriel
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Eric Rescorla'" <ekr@rtfm.com>
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Hi Eric,

Sec 4.2.2 of draft-ietf-rtcweb-security-05 has the following statement
twice:

"Once consent is verified, there still is some concern about
   misinterpretation attacks as described by Huang et al.[huang-w2sp]."

Regards
Partha


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Date: Fri, 19 Jul 2013 15:05:39 -0700
From: Adam Fineberg <fineberg@vline.me>
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To: Bernard Aboba <bernard_aboba@hotmail.com>
References: <51E7324A.3090405@vline.me> <BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl> <51E80DA2.4030500@vline.me> <BLU169-W107EA5AE4A7E9D3DB6AFD0893630@phx.gbl> <51E8B3B8.3090502@vline.me> <BLU401-EAS75C2417EF2B049AF14CDF893630@phx.gbl>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Bernard,

I'll take a look at the other codecs to see what fields are needed and 
how to encode them in the header.  Any help is appreciated as I'm not 
that familiar with H.264 and not at all familiar with H.265.

Regards,
Adam

On 7/18/13 10:56 PM, Bernard Aboba wrote:
> If we can support VP8/9 as well as H.264/5 SVC
> that would be a start. It seems doable to me.
>
> On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me 
> <mailto:fineberg@vline.me>> wrote:
>
>> Bernard,
>>
>> Are there other codecs you are thinking should be supported?  If it's 
>> generalized I would think we want to be able to cover all known 
>> scalable codecs. I'll look into the H264/SVC fields to see how to 
>> encode them in a generalized header.
>>
>> Regards,
>> Adam
>>
>> On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>> I think it may be possible to generalize this.  For example, for 
>>> H.264/SVC which can support temporal, spatial and quality 
>>> scalability, you would need the quality_id and dependency_id in 
>>> addition to the temporal_id (what you call the temporal layer index).
>>>
>>> ------------------------------------------------------------------------
>>> Date: Thu, 18 Jul 2013 08:45:38 -0700
>>> From: fineberg@vline.me
>>> To: bernard_aboba@hotmail.com
>>> CC: rtcweb@ietf.org; avtext@ietf.org
>>> Subject: Re: [rtcweb] Fwd: New Version Notification for 
>>> draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>
>>> Bernard,
>>>
>>> Good question.  I'm not familiar enough with the parameter 
>>> requirements of all other scalable codecs to be able to generalize.  
>>> If you'd like to help specify them, I'd be fine revising the draft 
>>> to generalize.
>>>
>>> Regards,
>>> Adam
>>>
>>> On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>>
>>>     Since the need is not codec specific (e.g. it arises with any
>>>     codec supporting temporal, spatial and quality scalability), why
>>>      a VP8-specific RTP extension?
>>>
>>>     ------------------------------------------------------------------------
>>>     Date: Wed, 17 Jul 2013 17:09:46 -0700
>>>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>     To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>     Subject: [rtcweb] Fwd: New Version Notification for
>>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>
>>>     Hi,
>>>
>>>     I'm working on WebRTC services and have found that while
>>>     developing services that forward VP8 video streams if we want to
>>>     take advantage of the VP8 temporal scaling we must get the
>>>     temporal layer information from the RTP header which requires us
>>>     to decrypt the SRTP packets. This is undesirable both because
>>>     the middle-box needs to have access to the keys as well as the
>>>     because of the added overhead of the decrypt/encrypt cycle. This
>>>     draft proposes an RTP header extension that will allow us to use
>>>     the VP8 temporal layer information included in the header
>>>     extension and therefore do forwarding without SRTP decryption.
>>>     Comments welcome.
>>>
>>>     Regards,
>>>     Adam Fineberg
>>>     fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>>>
>>>     -------- Original Message --------
>>>     Subject: 	New Version Notification for
>>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>     Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>>>     From: 	internet-drafts at ietf.org
>>>     <mailto:internet-drafts%20at%20ietf.org>
>>>     To: 	Adam Fineberg<fineberg at vline.com>
>>>     <mailto:fineberg%20at%20vline.com>
>>>
>>>
>>>
>>>     A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>     has been successfully submitted by Adam Fineberg and posted to the
>>>     IETF repository.
>>>
>>>     Filename:	 draft-fineberg-avtext-temporal-layer-ext
>>>     Revision:	 00
>>>     Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>>>     Creation date:	 2013-07-08
>>>     Group:		 Individual Submission
>>>     Number of pages: 6
>>>     URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>     Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>>>     Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>>
>>>
>>>     Abstract:
>>>         This document defines a mechanism by which packets of Real-Time
>>>         Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>>         indicate, in an RTP header extension, the temporal layer information
>>>         about the frame encoded in the RTP packet.  This information can be
>>>         used in a middlebox performing bandwidth management of streams
>>>         without requiring it to decrypt the streams.
>>>
>>>
>>>     _______________________________________________ rtcweb mailing
>>>     list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>> -- 
>>> Regards,
>>> Adam
>>

-- 
Regards,
Adam


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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Bernard,<br>
    <br>
    I'll take a look at the other codecs to see what fields are needed
    and how to encode them in the header.Â  Any help is appreciated as
    I'm not that familiar with H.264 and not at all familiar with H.265.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/18/13 10:56 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU401-EAS75C2417EF2B049AF14CDF893630@phx.gbl"
      type="cite">
      <meta http-equiv="content-type" content="text/html; charset=UTF-8">
      <div>If we can support VP8/9 as well as H.264/5 SVC</div>
      <div>that would be a start. It seems doable to me.</div>
      <div><br>
        On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" &lt;<a
          moz-do-not-send="true" href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;
        wrote:<br>
        <br>
      </div>
      <blockquote type="cite">
        <div>
          <meta content="text/html; charset=UTF-8"
            http-equiv="Content-Type">
          <div class="moz-cite-prefix">Bernard,<br>
            <br>
            Are there other codecs you are thinking should be
            supported?Â  If it's generalized I would think we want to be
            able to cover all known scalable codecs. I'll look into the
            H264/SVC fields to see how to encode them in a generalized
            header.<br>
            <br>
            Regards,<br>
            Adam<br>
            <br>
            On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
          </div>
          <blockquote
            cite="mid:BLU169-W107EA5AE4A7E9D3DB6AFD0893630@phx.gbl"
            type="cite">
            <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style>
            <div dir="ltr">I think it may be possible to generalize
              this.Â  For example, for H.264/SVC which can support
              temporal, spatial and quality scalability, you would need
              the quality_id and dependency_id in addition to the
              temporal_id (what you call the temporal layer index). Â Â  <br>
              <br>
              <div>
                <hr id="stopSpelling">Date: Thu, 18 Jul 2013 08:45:38
                -0700<br>
                From: <a moz-do-not-send="true"
                  class="moz-txt-link-abbreviated"
                  href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                To: <a moz-do-not-send="true"
                  class="moz-txt-link-abbreviated"
                  href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a><br>
                CC: <a moz-do-not-send="true"
                  class="moz-txt-link-abbreviated"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a
                  moz-do-not-send="true"
                  class="moz-txt-link-abbreviated"
                  href="mailto:avtext@ietf.org">avtext@ietf.org</a><br>
                Subject: Re: [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                <br>
                Bernard,<br>
                <br>
                Good question.Â  I'm not familiar enough with the
                parameter requirements of all other scalable codecs to
                be able to generalize.Â  If you'd like to help specify
                them, I'd be fine revising the draft to generalize.<br>
                <br>
                Regards,<br>
                Adam<br>
                <br>
                <div class="ecxmoz-cite-prefix">On 7/17/13 8:26 PM,
                  Bernard Aboba wrote:<br>
                </div>
                <blockquote
                  cite="mid:BLU169-W1238F58A9A258620DA2FE0F93620@phx.gbl">
                  <style><!--
.ExternalClass .ecxhmmessage P {
padding:0px;
}

.ExternalClass body.ecxhmmessage {
font-size:12pt;
font-family:Calibri;
}

--></style>
                  <div dir="ltr">Since the need is not codec specific
                    (e.g. it arises with any codec supporting temporal,
                    spatial and quality scalability), why <br>
                    Â a VP8-specific RTP extension? <br>
                    Â <br>
                    <div>
                      <hr id="ecxstopSpelling">Date: Wed, 17 Jul 2013
                      17:09:46 -0700<br>
                      From: <a moz-do-not-send="true"
                        class="ecxmoz-txt-link-abbreviated"
                        href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                      To: <a moz-do-not-send="true"
                        class="ecxmoz-txt-link-abbreviated"
                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                      Subject: [rtcweb] Fwd: New Version Notification
                      for
                      draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                      <br>
                      <span style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline

                        !important;white-space:normal;background-color:rgb(255,

                        255, 255);">Hi,</span><br style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <br style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <span style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline

                        !important;white-space:normal;background-color:rgb(255,

                        255, 255);">I'm working on WebRTC services and
                        have found that while developing services that
                        forward VP8 video streams if we want to take
                        advantage of the VP8 temporal scaling we must
                        get the temporal layer information from the RTP
                        header which requires us to decrypt the SRTP
                        packets. This is undesirable both because the
                        middle-box needs to have access to the keys as
                        well as the because of the added overhead of the
                        decrypt/encrypt cycle. This draft proposes an
                        RTP header extension that will allow us to use
                        the VP8 temporal layer information included in
                        the header extension and therefore do forwarding
                        without SRTP decryption. Comments welcome.</span><br
                        style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <br style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <span style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline

                        !important;white-space:normal;background-color:rgb(255,

                        255, 255);">Regards,</span><br
                        style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <span style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;display:inline

                        !important;white-space:normal;background-color:rgb(255,

                        255, 255);">Adam Fineberg</span><br
                        style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);">
                      <div class="ecxmoz-forward-container"
                        style="color:rgb(0, 0,
                        0);text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;background-color:rgb(255,

                        255, 255);"><a moz-do-not-send="true"
                          class="ecxmoz-txt-link-abbreviated"
                          href="mailto:fineberg%20at%20vline.com"
                          rel="nofollow">fineberg at vline.com</a><br>
                        <br>
                        -------- Original Message --------
                        <table class="ecxmoz-email-headers-table"
                          border="0" cellpadding="0" cellspacing="0">
                          <tbody>
                            <tr>
                              <th align="RIGHT" nowrap="nowrap"
                                valign="BASELINE">Subject:</th>
                              <td>New Version Notification for
                                draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                            </tr>
                            <tr>
                              <th align="RIGHT" nowrap="nowrap"
                                valign="BASELINE">Date:</th>
                              <td>Tue, 09 Jul 2013 10:02:05 -0700</td>
                            </tr>
                            <tr>
                              <th align="RIGHT" nowrap="nowrap"
                                valign="BASELINE">From:</th>
                              <td><a moz-do-not-send="true"
                                  class="ecxmoz-txt-link-abbreviated"
                                  href="mailto:internet-drafts%20at%20ietf.org"
                                  rel="nofollow">internet-drafts at
                                  ietf.org</a></td>
                            </tr>
                            <tr>
                              <th align="RIGHT" nowrap="nowrap"
                                valign="BASELINE">To:</th>
                              <td>Adam Fineberg<span
                                  class="ecxApple-converted-space">Â </span><a
                                  moz-do-not-send="true"
                                  class="ecxmoz-txt-link-rfc2396E"
                                  href="mailto:fineberg%20at%20vline.com"
                                  rel="nofollow">&lt;fineberg at
                                  vline.com&gt;</a></td>
                            </tr>
                          </tbody>
                        </table>
                        <br>
                        <br>
                        <pre style="width:939.59px;white-space:pre-wrap;-ms-word-wrap:break-word;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank" rel="nofollow">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank" rel="nofollow">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank" rel="nofollow">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                      </div>
                      <br>
                      _______________________________________________
                      rtcweb mailing list <a moz-do-not-send="true"
                        class="ecxmoz-txt-link-abbreviated"
                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
                      <a moz-do-not-send="true"
                        class="ecxmoz-txt-link-freetext"
                        href="https://www.ietf.org/mailman/listinfo/rtcweb"
                        target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
                  </div>
                </blockquote>
                <br>
                <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
              </div>
            </div>
          </blockquote>
          <br>
        </div>
      </blockquote>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
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Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not 
at all familiar with H.265.  I'll review the reference and see what I 
can figure.  It seems though to me that you are suggesting that except 
in the simple case, that the data for H.265 would not be well suited to 
a header extension, am I understanding you correctly?  There is no 
reason the middlebox couldn't get out of band signaling of the VPS as 
you mention but that would not be within the scope of this header extension.

Any insights are appreciated.

Regards,
Adam

On 7/19/13 8:33 AM, Stephan Wenger wrote:
> Hi,
> I also believe that 16 bits should be enough.  For H.264 and VP8 that 
> has already been demonstrated.  For H.265, some initial thoughts 
> below.  Apologies for the word-count.
>
> The scalable version of H265 (called SHVC) is currently under 
> development.  The current working draft can be found here: 
> http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip. 
>  Therein, the options for defining layering structures are 
> considerably more complex.  To start, we have 3 bits for the temporal 
> ID in the NAL unit header of the H.265 version 1 (HEVC) base 
> specification (temporal scalability is already nicely supported in 
> version 1).  Just like in SVC.  In the scalable extension, the NAL 
> unit header contains a six bit field that points into a data structure 
> known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits 
> are mapped to to a position in a directed graph (specified through 
> "dimension_id[][]"), which tells you about the reference relationship 
> of the layer in question and its parent layer.  One can recursively 
> follow the graph to determine what used to be called dependency_id, 
> quality_id, view_id, and whatnot.  The six bit pointer field can (or: 
> is to be when possible) organized by the encoder such that it is 
> prudent for a middle box to throw away NAL units (belonging to layers) 
> with higher values of the six bit field first, before throwing away 
> NAL units with lower values.  Relying on this feature, 3+6 bits == 9 
> bits should be fine for the header extension.
>
> That said, the ordering by the encoder is just a recommendation, and 
> there may well be cases where different pruning strategies may be 
> advisable.  For example, a layering structure could be constructed 
> that expands into two branches, one using 2D scalable tools only, the 
> other including view_id for multi view coding.  By looking at the six 
> bit field alone, a middle box will not be able to meaningfully remove 
> NAL units belonging to one of the branches completely while pruning 
> the other branch.  In order to meaningfully deal with that scenario, 
> there would be two options: one to represent the dimension_id[][] (and 
> associated control info) in the header extension, or require the 
> middle box to have access to the VPS and be able to interpret its 
> content.  The further could take considerably more than 16 bits and we 
> would be talking about a variable length data structure.  The latter 
> requires the middle box to have state and a mechanism to intercept the 
> VPS (through signaling---as the encrypted in-band VPS would not be 
> useful under the assumption that the middle box does not have the key 
> to the media---which is the motivation of the draft in the first 
> place).  I personally don't mind at all the second mechanism, as I'm a 
> big fan of out-of-band parameter set transmission and any middle box 
> must be in the signaling path anyway to meaningfully manipulate RTP. 
>  I do not like the first option due to its variable, and possibly 
> substantial, overhead.
>
> Stephan
>
>
> From: Justin Uberti <juberti@google.com <mailto:juberti@google.com>>
> Date: Friday, 19 July, 2013 06:32
> To: Bernard Aboba <bernard_aboba@hotmail.com 
> <mailto:bernard_aboba@hotmail.com>>
> Cc: "avtext@ietf.org <mailto:avtext@ietf.org>" <avtext@ietf.org 
> <mailto:avtext@ietf.org>>, "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" 
> <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
> Subject: Re: [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Agree those are the right codecs to design for. Since in each case 
> there are fairly low limits on the number of supported layers (i.e. 3 
> spatial layers for SVC), I think it should be possible to pack the 
> temporal, spatial, quality layer ids into 16 bits.
>
>
> On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba 
> <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>> wrote:
>
>     If we can support VP8/9 as well as H.264/5 SVC
>     that would be a start. It seems doable to me.
>
>     On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me
>     <mailto:fineberg@vline.me>> wrote:
>
>>     Bernard,
>>
>>     Are there other codecs you are thinking should be supported?  If
>>     it's generalized I would think we want to be able to cover all
>>     known scalable codecs. I'll look into the H264/SVC fields to see
>>     how to encode them in a generalized header.
>>
>>     Regards,
>>     Adam
>>
>>     On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>>     I think it may be possible to generalize this.  For example, for
>>>     H.264/SVC which can support temporal, spatial and quality
>>>     scalability, you would need the quality_id and dependency_id in
>>>     addition to the temporal_id (what you call the temporal layer
>>>     index).
>>>
>>>     ------------------------------------------------------------------------
>>>     Date: Thu, 18 Jul 2013 08:45:38 -0700
>>>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>     To: bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>
>>>     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>; avtext@ietf.org
>>>     <mailto:avtext@ietf.org>
>>>     Subject: Re: [rtcweb] Fwd: New Version Notification for
>>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>
>>>     Bernard,
>>>
>>>     Good question.  I'm not familiar enough with the parameter
>>>     requirements of all other scalable codecs to be able to
>>>     generalize.  If you'd like to help specify them, I'd be fine
>>>     revising the draft to generalize.
>>>
>>>     Regards,
>>>     Adam
>>>
>>>     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>>
>>>         Since the need is not codec specific (e.g. it arises with
>>>         any codec supporting temporal, spatial and quality
>>>         scalability), why
>>>          a VP8-specific RTP extension?
>>>
>>>         ------------------------------------------------------------------------
>>>         Date: Wed, 17 Jul 2013 17:09:46 -0700
>>>         From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>         Subject: [rtcweb] Fwd: New Version Notification for
>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>
>>>         Hi,
>>>
>>>         I'm working on WebRTC services and have found that while
>>>         developing services that forward VP8 video streams if we
>>>         want to take advantage of the VP8 temporal scaling we must
>>>         get the temporal layer information from the RTP header which
>>>         requires us to decrypt the SRTP packets. This is undesirable
>>>         both because the middle-box needs to have access to the keys
>>>         as well as the because of the added overhead of the
>>>         decrypt/encrypt cycle. This draft proposes an RTP header
>>>         extension that will allow us to use the VP8 temporal layer
>>>         information included in the header extension and therefore
>>>         do forwarding without SRTP decryption. Comments welcome.
>>>
>>>         Regards,
>>>         Adam Fineberg
>>>         fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>>>
>>>         -------- Original Message --------
>>>         Subject: 	New Version Notification for
>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>         Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>>>         From: 	internet-drafts at ietf.org
>>>         <mailto:internet-drafts%20at%20ietf.org>
>>>         To: 	Adam Fineberg<fineberg at vline.com>
>>>         <mailto:fineberg%20at%20vline.com>
>>>
>>>
>>>
>>>         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>         has been successfully submitted by Adam Fineberg and posted to the
>>>         IETF repository.
>>>
>>>         Filename:	 draft-fineberg-avtext-temporal-layer-ext
>>>         Revision:	 00
>>>         Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>>>         Creation date:	 2013-07-08
>>>         Group:		 Individual Submission
>>>         Number of pages: 6
>>>         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>>>         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>>
>>>
>>>         Abstract:
>>>             This document defines a mechanism by which packets of Real-Time
>>>             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>>             indicate, in an RTP header extension, the temporal layer information
>>>             about the frame encoded in the RTP packet.  This information can be
>>>             used in a middlebox performing bandwidth management of streams
>>>             without requiring it to decrypt the streams.
>>>
>>>
>>>         _______________________________________________ rtcweb
>>>         mailing list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>         https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>>     -- 
>>>     Regards,
>>>     Adam
>>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

-- 
Regards,
Adam


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    Stephan,<br>
    <br>
    Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm
    not at all familiar with H.265.&nbsp; I'll review the reference and see
    what I can figure.&nbsp; It seems though to me that you are suggesting
    that except in the simple case, that the data for H.265 would not be
    well suited to a header extension, am I understanding you
    correctly?&nbsp; There is no reason the middlebox couldn't get out of
    band signaling of the VPS as you mention but that would not be
    within the scope of this header extension.<br>
    <br>
    Any insights are appreciated.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/19/13 8:33 AM, Stephan Wenger
      wrote:<br>
    </div>
    <blockquote cite="mid:CE0E9F56.9F89B%25stewe@stewe.org" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div>Hi,</div>
      <div>I also believe that 16 bits should be enough. &nbsp;For H.264 and
        VP8 that has already been demonstrated. &nbsp;For H.265, some initial
        thoughts below. &nbsp;Apologies for the word-count.</div>
      <div><br>
      </div>
      <div>The scalable version of H265 (called SHVC) is currently under
        development. &nbsp;The current working draft can be found here:&nbsp;<a
          moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
        &nbsp;Therein, the options for defining layering structures are
        considerably more complex. &nbsp;To start, we have 3 bits for the
        temporal ID in the NAL unit header of the H.265 version 1 (HEVC)
        base specification (temporal scalability is already nicely
        supported in version 1). &nbsp;Just like in SVC. &nbsp;In the scalable
        extension, the NAL unit header contains a six bit field that
        points into a data structure known as "Video Parameter Set"
        (VPS). &nbsp;Inside the VPS, those six bits are mapped to to a
        position in a directed graph (specified through
        "dimension_id[][]"), which tells you about the reference
        relationship of the layer in question and its parent layer. &nbsp;One
        can recursively follow the graph to determine what used to be
        called dependency_id, quality_id, view_id, and whatnot. &nbsp;The six
        bit pointer field can (or: is to be when possible) organized by
        the encoder such that it is prudent for a middle box to throw
        away NAL units (belonging to layers) with higher values of the
        six bit field first, before throwing away NAL units with lower
        values. &nbsp;Relying on this feature, 3+6 bits == 9 bits should be
        fine for the header extension.</div>
      <div><br>
      </div>
      <div>That said, the ordering by the encoder is just a
        recommendation, and there may well be cases where different
        pruning strategies may be advisable. &nbsp;For example, a layering
        structure could be constructed that expands into two branches,
        one using 2D scalable tools only, the other including view_id
        for multi view coding. &nbsp;By looking at the six bit field alone, a
        middle box will not be able to meaningfully remove NAL units
        belonging to one of the branches completely while pruning the
        other branch. &nbsp;In order to meaningfully deal with that scenario,
        there would be two options: one to represent the
        dimension_id[][] (and associated control info) in the header
        extension, or require the middle box to have access to the VPS
        and be able to interpret its content. &nbsp;The further could take
        considerably more than 16 bits and we would be talking about a
        variable length data structure. &nbsp;The latter requires the middle
        box to have state and a mechanism to intercept the VPS (through
        signaling&#8212;as the encrypted in-band VPS would not be useful under
        the assumption that the middle box does not have the key to the
        media&#8212;which is the motivation of the draft in the first place).
        &nbsp;I personally don't mind at all the second mechanism, as I'm a
        big fan of out-of-band parameter set transmission and any middle
        box must be in the signaling path anyway to meaningfully
        manipulate RTP. &nbsp;I do not like the first option due to its
        variable, and possibly substantial, overhead.</div>
      <div><br>
      </div>
      <div>Stephan &nbsp;&nbsp;</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <span id="OLK_SRC_BODY_SECTION">
        <div style="font-family:Calibri; font-size:11pt;
          text-align:left; color:black; BORDER-BOTTOM: medium none;
          BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT:
          0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;
          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
          <span style="font-weight:bold">From: </span>Justin Uberti
          &lt;<a moz-do-not-send="true" href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
          <span style="font-weight:bold">Date: </span>Friday, 19 July,
          2013 06:32 <br>
          <span style="font-weight:bold">To: </span>Bernard Aboba &lt;<a
            moz-do-not-send="true"
            href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
          <span style="font-weight:bold">Cc: </span>"<a
            moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
          <span style="font-weight:bold">Subject: </span>Re: [rtcweb]
          Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
        </div>
        <div><br>
        </div>
        <div>
          <div>
            <div dir="ltr">Agree those are the right codecs to design
              for. Since in each case there are fairly low limits on the
              number of supported layers (i.e. 3 spatial layers for
              SVC), I think it should be possible to pack the temporal,
              spatial, quality layer ids into 16 bits.
              <div class="gmail_extra"><br>
                <br>
                <div class="gmail_quote">On Fri, Jul 19, 2013 at 1:56
                  AM, Bernard Aboba <span dir="ltr">
                    &lt;<a moz-do-not-send="true"
                      href="mailto:bernard_aboba@hotmail.com"
                      target="_blank">bernard_aboba@hotmail.com</a>&gt;</span>
                  wrote:<br>
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    <div dir="auto">
                      <div>If we can support VP8/9 as well as H.264/5
                        SVC</div>
                      <div>that would be a start. It seems doable to me.</div>
                      <div>
                        <div>
                          <div><br>
                            On Jul 18, 2013, at 8:34 PM, "Adam Fineberg"
                            &lt;<a moz-do-not-send="true"
                              href="mailto:fineberg@vline.me"
                              target="_blank">fineberg@vline.me</a>&gt;
                            wrote:<br>
                            <br>
                          </div>
                          <blockquote type="cite">
                            <div>
                              <div>Bernard,<br>
                                <br>
                                Are there other codecs you are thinking
                                should be supported?&nbsp; If it's
                                generalized I would think we want to be
                                able to cover all known scalable codecs.
                                I'll look into the H264/SVC fields to
                                see how to encode them in a generalized
                                header.<br>
                                <br>
                                Regards,<br>
                                Adam<br>
                                <br>
                                On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
                              </div>
                              <blockquote type="cite">
                                <div dir="ltr">I think it may be
                                  possible to generalize this.&nbsp; For
                                  example, for H.264/SVC which can
                                  support temporal, spatial and quality
                                  scalability, you would need the
                                  quality_id and dependency_id in
                                  addition to the temporal_id (what you
                                  call the temporal layer index). &nbsp;&nbsp; <br>
                                  <br>
                                  <div>
                                    <hr>
                                    Date: Thu, 18 Jul 2013 08:45:38
                                    -0700<br>
                                    From: <a moz-do-not-send="true"
                                      href="mailto:fineberg@vline.me"
                                      target="_blank">fineberg@vline.me</a><br>
                                    To: <a moz-do-not-send="true"
                                      href="mailto:bernard_aboba@hotmail.com"
                                      target="_blank">bernard_aboba@hotmail.com</a><br>
                                    CC: <a moz-do-not-send="true"
                                      href="mailto:rtcweb@ietf.org"
                                      target="_blank">rtcweb@ietf.org</a>;
                                    <a moz-do-not-send="true"
                                      href="mailto:avtext@ietf.org"
                                      target="_blank">
                                      avtext@ietf.org</a><br>
                                    Subject: Re: [rtcweb] Fwd: New
                                    Version Notification for
                                    draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                    <br>
                                    Bernard,<br>
                                    <br>
                                    Good question.&nbsp; I'm not familiar
                                    enough with the parameter
                                    requirements of all other scalable
                                    codecs to be able to generalize.&nbsp; If
                                    you'd like to help specify them, I'd
                                    be fine revising the draft to
                                    generalize.<br>
                                    <br>
                                    Regards,<br>
                                    Adam<br>
                                    <br>
                                    <div>On 7/17/13 8:26 PM, Bernard
                                      Aboba wrote:<br>
                                    </div>
                                    <blockquote>
                                      <div dir="ltr">Since the need is
                                        not codec specific (e.g. it
                                        arises with any codec supporting
                                        temporal, spatial and quality
                                        scalability), why
                                        <br>
                                        &nbsp;a VP8-specific RTP extension? <br>
                                        &nbsp;<br>
                                        <div>
                                          <hr>
                                          Date: Wed, 17 Jul 2013
                                          17:09:46 -0700<br>
                                          From: <a
                                            moz-do-not-send="true"
                                            href="mailto:fineberg@vline.me"
                                            target="_blank">fineberg@vline.me</a><br>
                                          To: <a moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                          Subject: [rtcweb] Fwd: New
                                          Version Notification for
                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                          <br>
                                          <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">I'm
                                            working on WebRTC services
                                            and have found that while
                                            developing services that
                                            forward VP8 video streams if
                                            we want to take advantage of
                                            the VP8 temporal scaling we
                                            must get the temporal layer
                                            information from the RTP
                                            header which requires us to
                                            decrypt the SRTP packets.
                                            This is undesirable both
                                            because the middle-box needs
                                            to have access to the keys
                                            as well as the because of
                                            the added overhead of the
                                            decrypt/encrypt cycle. This
                                            draft proposes an RTP header
                                            extension that will allow us
                                            to use the VP8 temporal
                                            layer information included
                                            in the header extension and
                                            therefore do forwarding
                                            without SRTP decryption.
                                            Comments welcome.</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Regards,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Adam
                                            Fineberg</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                          <div
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px"><a
                                              moz-do-not-send="true"
                                              href="mailto:fineberg%20at%20vline.com"
                                              rel="nofollow"
                                              target="_blank">fineberg
                                              at vline.com</a><br>
                                            <br>
                                            -------- Original Message
                                            --------
                                            <table border="0"
                                              cellpadding="0"
                                              cellspacing="0">
                                              <tbody>
                                                <tr>
                                                  <th align="RIGHT"
                                                    nowrap="nowrap"
                                                    valign="BASELINE">Subject:</th>
                                                  <td>New Version
                                                    Notification for
                                                    draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                                                </tr>
                                                <tr>
                                                  <th align="RIGHT"
                                                    nowrap="nowrap"
                                                    valign="BASELINE">Date:</th>
                                                  <td>Tue, 09 Jul 2013
                                                    10:02:05 -0700</td>
                                                </tr>
                                                <tr>
                                                  <th align="RIGHT"
                                                    nowrap="nowrap"
                                                    valign="BASELINE">From:</th>
                                                  <td><a
                                                      moz-do-not-send="true"
href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow"
                                                      target="_blank">internet-drafts
                                                      at ietf.org</a></td>
                                                </tr>
                                                <tr>
                                                  <th align="RIGHT"
                                                    nowrap="nowrap"
                                                    valign="BASELINE">To:</th>
                                                  <td>Adam Fineberg<span>&nbsp;</span><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                      rel="nofollow"
                                                      target="_blank">&lt;fineberg
                                                      at vline.com&gt;</a></td>
                                                </tr>
                                              </tbody>
                                            </table>
                                            <br>
                                            <br>
                                            <pre style="width:939.59px;white-space:pre-wrap">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel="nofollow" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel="nofollow" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" rel="nofollow" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                                          </div>
                                          <br>
                                          _______________________________________________
                                          rtcweb mailing list <a
                                            moz-do-not-send="true"
                                            href="mailto:rtcweb@ietf.org"
                                            target="_blank">
                                            rtcweb@ietf.org</a> <a
                                            moz-do-not-send="true"
                                            href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                            target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
                                      </div>
                                    </blockquote>
                                    <br>
                                    <pre>-- 
Regards,
Adam</pre>
                                  </div>
                                </div>
                              </blockquote>
                              <br>
                            </div>
                          </blockquote>
                        </div>
                      </div>
                    </div>
                    <br>
                    _______________________________________________<br>
                    rtcweb mailing list<br>
                    <a moz-do-not-send="true"
                      href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                    <a moz-do-not-send="true"
                      href="https://www.ietf.org/mailman/listinfo/rtcweb"
                      target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                    <br>
                  </blockquote>
                </div>
                <br>
              </div>
            </div>
          </div>
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      </span>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
avtext mailing list
<a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/avtext">https://www.ietf.org/mailman/listinfo/avtext</a>
</pre>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
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From: Stephan Wenger <stewe@stewe.org>
To: Adam Fineberg <fineberg@vline.me>
Thread-Topic: [avtext] [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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--_000_CE0F0BEB9F95Dstewesteweorg_
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Hi,

From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Friday, 19 July, 2013 15:12
To: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Cc: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, Bernard =
Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>>, "avtex=
t@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtext@ietf.org=
>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not at=
 all familiar with H.265.  I'll review the reference and see what I can fig=
ure.

StW: Good luck :-)

It seems though to me that you are suggesting that except in the simple cas=
e, that the data for H.265 would not be well suited to a header extension, =
am I understanding you correctly?  There is no reason the middlebox couldn'=
t get out of band signaling of the VPS as you mention but that would not be=
 within the scope of this header extension.

StW: well, if you would copy the layer_id into your header extension (just =
as you need to do for the simple case), a really smart middle box could use=
 this information just as a decoder uses it, assuming that it intercepted t=
he VPS in the first place.  Insofar, I wouldn't rule out the second option =
on technical grounds.  Whether any of the actual products would bother to d=
o that, ever, is another question.  I think the case ought to be documented=
, though.  I can help drafting text.
While we are at it: doing this right could mean that you need multiple spec=
s.  First, a generic header extension mechanism dedicated to side informati=
on required for pruning of RTP packet streams=97ideally not only for scalab=
le video, although that is the main customer today.  And second, for each "=
payload" (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v=
2 (including scalable and 3D extensions, which are not yet finalized), VP8,=
 and VP9 (I know little about the latter), plus Daala, and whatnot) a mappi=
ng of the bits available in the generic header extension to the bits in the=
 payload itself (NAL header and VPS in case of H.265, NALU header in SVC, a=
nd the fields you mention for VP8).

Stephan


Any insights are appreciated.

Regards,
Adam

On 7/19/13 8:33 AM, Stephan Wenger wrote:
Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling=97as the encrypted in-band VPS would not be useful under the assu=
mption that the middle box does not have the key to the media=97which is th=
e motivation of the draft in the first place).  I personally don't mind at =
all the second mechanism, as I'm a big fan of out-of-band parameter set tra=
nsmission and any middle box must be in the signaling path anyway to meanin=
gfully manipulate RTP.  I do not like the first option due to its variable,=
 and possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.


On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:

Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).

________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam

On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:        New Version Notification for draft-fineberg-avtext-temporal=
-layer-ext-00.txt
Date:   Tue, 09 Jul 2013 10:02:05 -0700
From:   internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>
To:     Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.=
com>



A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:        draft-fineberg-avtext-temporal-layer-ext
Revision:        00
Title:           A Real-Time Transport Protocol (RTP) Header Extension for =
VP8 Temporal Layer Information
Creation date:   2013-07-08
Group:           Individual Submission
Number of pages: 6
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.


_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--
Regards,
Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb





_______________________________________________
avtext mailing list
avtext@ietf.org<mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinf=
o/avtext


--
Regards,
Adam

--_000_CE0F0BEB9F95Dstewesteweorg_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">Hi,</font></div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; color: rgb=
(0, 0, 0); ">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri, sans-serif=
; font-size: 14px; color: rgb(0, 0, 0); ">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Adam Fineberg &lt;<a href=3D"=
mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 19 July, 2013 15:12 <=
br>
<span style=3D"font-weight:bold">To: </span>Stephan Wenger &lt;<a href=3D"m=
ailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Justin Uberti &lt;<a href=3D"ma=
ilto:juberti@google.com">juberti@google.com</a>&gt;, Bernard Aboba &lt;<a h=
ref=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,=
 &quot;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, &quot;<a h=
ref=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mai=
lto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [avtext] [rtcweb] Fwd:=
 New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.t=
xt<br>
</div>
<div><br>
</div>
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">Stephan,<br>
<br>
Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm n=
ot at all familiar with H.265.&nbsp; I'll review the reference and see what=
 I can figure.&nbsp;</div>
</div>
</span>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; color: rgb=
(0, 0, 0); ">
<br>
</div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">StW: Good luck :-) &nbsp;</font></div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; color: rgb=
(0, 0, 0); ">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri, sans-serif=
; font-size: 14px; color: rgb(0, 0, 0); ">
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">It seems though to me that you ar=
e suggesting that except in the simple case, that the data for H.265 would =
not be well suited to a header extension, am I understanding you correctly?=
&nbsp; There is no reason the middlebox couldn't
 get out of band signaling of the VPS as you mention but that would not be =
within the scope of this header extension.</div>
</div>
</span>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; color: rgb=
(0, 0, 0); ">
<br>
</div>
<div><font color=3D"#0000ff"><font face=3D"Calibri,sans-serif">StW: well, i=
f you would copy the layer_id into your header extension (just as you need =
to do for the simple case), a really smart middle box could use this inform=
ation just as a decoder uses it,&nbsp;assuming&nbsp;that
 it intercepted the VPS in the first place. &nbsp;Insofar, I wouldn't rule =
out the second option on technical grounds. &nbsp;Whether any of the actual=
 products would bother to do that, ever, is another question. &nbsp;I think=
 the case ought to be documented, though. &nbsp;I can
 help drafting text.</font></font></div>
<div><font color=3D"#0000ff"><font face=3D"Calibri,sans-serif">While we are=
 at it: doing this right could mean that you need multiple specs. &nbsp;Fir=
st, a generic header extension mechanism dedicated to side information requ=
ired for pruning of RTP packet streams=97ideally
 not only for scalable video, although that is the main customer today. &nb=
sp;And second, for each &quot;payload&quot; (at present we are talking abou=
t H.264/SVC, H.265v1 (HEVC), H.265v2 (including scalable and 3D extensions,=
 which are not yet finalized), VP8, and VP9 (I
 know little about the latter), plus Daala, and whatnot) a mapping of the b=
its available in the generic header extension to the bits in the payload it=
self (NAL header and VPS in case of H.265, NALU header in SVC, and the fiel=
ds you mention for VP8).</font></font></div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><br>
</div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">Stephan</font></div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri, sans-serif=
; font-size: 14px; color: rgb(0, 0, 0); ">
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000"><br>
<br>
Any insights are appreciated.<br>
<br>
Regards,<br>
Adam<br>
<br>
<div class=3D"moz-cite-prefix">On 7/19/13 8:33 AM, Stephan Wenger wrote:<br=
>
</div>
<blockquote cite=3D"mid:CE0E9F56.9F89B%25stewe@stewe.org" type=3D"cite">
<div>Hi,</div>
<div>I also believe that 16 bits should be enough. &nbsp;For H.264 and VP8 =
that has already been demonstrated. &nbsp;For H.265, some initial thoughts =
below. &nbsp;Apologies for the word-count.</div>
<div><br>
</div>
<div>The scalable version of H265 (called SHVC) is currently under developm=
ent. &nbsp;The current working draft can be found here:&nbsp;<a moz-do-not-=
send=3D"true" href=3D"http://phenix.int-evry.fr/jct/doc_end_user/documents/=
13_Incheon/wg11/JCTVC-M1008-v3.zip">http://phenix.int-evry.fr/jct/doc_end_u=
ser/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.</div>
<div><br>
</div>
<div>That said, the ordering by the encoder is just a recommendation, and t=
here may well be cases where different pruning strategies may be advisable.=
 &nbsp;For example, a layering structure could be constructed that expands =
into two branches, one using 2D scalable
 tools only, the other including view_id for multi view coding. &nbsp;By lo=
oking at the six bit field alone, a middle box will not be able to meaningf=
ully remove NAL units belonging to one of the branches completely while pru=
ning the other branch. &nbsp;In order to meaningfully
 deal with that scenario, there would be two options: one to represent the =
dimension_id[][] (and associated control info) in the header extension, or =
require the middle box to have access to the VPS and be able to interpret i=
ts content. &nbsp;The further could take
 considerably more than 16 bits and we would be talking about a variable le=
ngth data structure. &nbsp;The latter requires the middle box to have state=
 and a mechanism to intercept the VPS (through signaling=97as the encrypted=
 in-band VPS would not be useful under
 the assumption that the middle box does not have the key to the media=97wh=
ich is the motivation of the draft in the first place). &nbsp;I personally =
don't mind at all the second mechanism, as I'm a big fan of out-of-band par=
ameter set transmission and any middle
 box must be in the signaling path anyway to meaningfully manipulate RTP. &=
nbsp;I do not like the first option due to its variable, and possibly subst=
antial, overhead.</div>
<div><br>
</div>
<div>Stephan &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt;
          text-align:left; color:black; BORDER-BOTTOM: medium none;
          BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT:
          0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;
          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Justin Uberti &lt;<a moz-do-n=
ot-send=3D"true" href=3D"mailto:juberti@google.com">juberti@google.com</a>&=
gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 19 July, 2013 06:32 <=
br>
<span style=3D"font-weight:bold">To: </span>Bernard Aboba &lt;<a moz-do-not=
-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotm=
ail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a moz-do-not-send=3D"tru=
e" href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quot; &lt;<a moz-do-=
not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, =
&quot;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@ie=
tf.org</a>&quot;
 &lt;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Fwd: New Vers=
ion Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">Agree those are the right codecs to design for. Since in e=
ach case there are fairly low limits on the number of supported layers (i.e=
. 3 spatial layers for SVC), I think it should be possible to pack the temp=
oral, spatial, quality layer ids into
 16 bits.
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <=
span dir=3D"ltr">
&lt;<a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>If we can support VP8/9 as well as H.264/5 SVC</div>
<div>that would be a start. It seems doable to me.</div>
<div>
<div>
<div><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a moz-do-not-se=
nd=3D"true" href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vl=
ine.me</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div>Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">I think it may be possible to generalize this.&nbsp; For e=
xample, for H.264/SVC which can support temporal, spatial and quality scala=
bility, you would need the quality_id and dependency_id in addition to the =
temporal_id (what you call the temporal
 layer index). &nbsp;&nbsp; <br>
<br>
<div>
<hr>
Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">
bernard_aboba@hotmail.com</a><br>
CC: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a>;
<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org" target=3D"_blan=
k">avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam<br>
<br>
<div>On 7/17/13 8:26 PM, Bernard Aboba wrote:<br>
</div>
<blockquote>
<div dir=3D"ltr">Since the need is not codec specific (e.g. it arises with =
any codec supporting temporal, spatial and quality scalability), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;<br>
<div>
<hr>
Date: Wed, 17 Jul 2013 17:09:46 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br s=
tyle=3D"text-indent:0px;letter-spacing:normal;text-transform:none;white-spa=
ce:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">I'm working on =
WebRTC services and have found that while developing services that forward =
VP8 video streams if we want to take
 advantage of the VP8 temporal scaling we must get the temporal layer infor=
mation from the RTP header which requires us to decrypt the SRTP packets. T=
his is undesirable both because the middle-box needs to have access to the =
keys as well as the because of the
 added overhead of the decrypt/encrypt cycle. This draft proposes an RTP he=
ader extension that will allow us to use the VP8 temporal layer information=
 included in the header extension and therefore do forwarding without SRTP =
decryption. Comments welcome.</span><br style=3D"text-indent:0px;letter-spa=
cing:normal;text-transform:none;white-space:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Regards,</span>=
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Adam Fineberg</=
span><br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none=
;white-space:normal;word-spacing:0px">
<div style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whi=
te-space:normal;word-spacing:0px">
<a moz-do-not-send=3D"true" href=3D"mailto:fineberg%20at%20vline.com" rel=
=3D"nofollow" target=3D"_blank">fineberg at vline.com</a><br>
<br>
-------- Original Message --------
<table border=3D"0" cellpadding=3D"0" cellspacing=3D"0">
<tbody>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Subject:</th>
<td>New Version Notification for draft-fineberg-avtext-temporal-layer-ext-0=
0.txt</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Date:</th>
<td>Tue, 09 Jul 2013 10:02:05 -0700</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">From:</th>
<td><a moz-do-not-send=3D"true" href=3D"mailto:internet-drafts%20at%20ietf.=
org" rel=3D"nofollow" target=3D"_blank">internet-drafts at ietf.org</a></td=
>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">To:</th>
<td>Adam Fineberg<span>&nbsp;</span><a moz-do-not-send=3D"true" href=3D"mai=
lto:fineberg%20at%20vline.com" rel=3D"nofollow" target=3D"_blank">&lt;fineb=
erg at vline.com&gt;</a></td>
</tr>
</tbody>
</table>
<br>
<br>
<pre style=3D"width:939.59px;white-space:pre-wrap">A new version of I-D, dr=
aft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send=3D"true" href=3D"http://www.ietf.org/in=
ternet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel=3D"nofol=
low" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-fineberg-a=
vtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send=3D"true" href=3D"http://datatracker.iet=
f.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel=3D"nofollow" target=
=3D"_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-=
layer-ext</a>
Htmlized:        <a moz-do-not-send=3D"true" href=3D"http://tools.ietf.org/=
html/draft-fineberg-avtext-temporal-layer-ext-00" rel=3D"nofollow" target=
=3D"_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer=
-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
</div>
<br>
_______________________________________________ rtcweb mailing list <a moz-=
do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a moz-do-not-send=3D"true" href=3D"https://www.ietf.or=
g/mailman/listinfo/rtcweb" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
</div>
</blockquote>
<br>
<pre>--=20
Regards,
Adam</pre>
</div>
</div>
</blockquote>
<br>
</div>
</blockquote>
</div>
</div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blan=
k">rtcweb@ietf.org</a><br>
<a moz-do-not-send=3D"true" href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><b=
r>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</span><br>
<fieldset class=3D"mimeAttachmentHeader"></fieldset> <br>
<pre wrap=3D"">_______________________________________________
avtext mailing list
<a class=3D"moz-txt-link-abbreviated" href=3D"mailto:avtext@ietf.org">avtex=
t@ietf.org</a><a class=3D"moz-txt-link-freetext" href=3D"https://www.ietf.o=
rg/mailman/listinfo/avtext">https://www.ietf.org/mailman/listinfo/avtext</a=
></pre>
</blockquote>
<br>
<pre class=3D"moz-signature" cols=3D"72">--=20
Regards,
Adam</pre>
</div>
</div>
</span>
</body>
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Stephan,

On 7/19/13 3:44 PM, Stephan Wenger wrote:
> Hi,
>
> From: Adam Fineberg <fineberg@vline.me <mailto:fineberg@vline.me>>
> Date: Friday, 19 July, 2013 15:12
> To: Stephan Wenger <stewe@stewe.org <mailto:stewe@stewe.org>>
> Cc: Justin Uberti <juberti@google.com <mailto:juberti@google.com>>, 
> Bernard Aboba <bernard_aboba@hotmail.com 
> <mailto:bernard_aboba@hotmail.com>>, "avtext@ietf.org 
> <mailto:avtext@ietf.org>" <avtext@ietf.org <mailto:avtext@ietf.org>>, 
> "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org 
> <mailto:rtcweb@ietf.org>>
> Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Stephan,
>
> Thanks for the info and the reference.  I'm not sure I follow as I'm 
> not at all familiar with H.265.  I'll review the reference and see 
> what I can figure.
>
> StW: Good luck :-)
>
> It seems though to me that you are suggesting that except in the 
> simple case, that the data for H.265 would not be well suited to a 
> header extension, am I understanding you correctly?  There is no 
> reason the middlebox couldn't get out of band signaling of the VPS as 
> you mention but that would not be within the scope of this header 
> extension.
>
> StW: well, if you would copy the layer_id into your header extension 
> (just as you need to do for the simple case), a really smart middle 
> box could use this information just as a decoder uses 
> it, assuming that it intercepted the VPS in the first place.  Insofar, 
> I wouldn't rule out the second option on technical grounds.  Whether 
> any of the actual products would bother to do that, ever, is another 
> question.  I think the case ought to be documented, though.  I can 
> help drafting text.

Any help is appreciated.

> While we are at it: doing this right could mean that you need multiple 
> specs.  First, a generic header extension mechanism dedicated to side 
> information required for pruning of RTP packet streams—ideally not 
> only for scalable video, although that is the main customer today. 
>  And second, for each "payload" (at present we are talking about 
> H.264/SVC, H.265v1 (HEVC), H.265v2 (including scalable and 3D 
> extensions, which are not yet finalized), VP8, and VP9 (I know little 
> about the latter), plus Daala, and whatnot) a mapping of the bits 
> available in the generic header extension to the bits in the payload 
> itself (NAL header and VPS in case of H.265, NALU header in SVC, and 
> the fields you mention for VP8).

I was just thinking about that since it seems like the extension payload 
maybe completely specific and therefore not suitable for the same spec.  
The problem in my mind with taking the hierarchical a[[roach you mention 
is that I'm not sure what if any common data is suitable accross all 
codecs and since there is already a spec for the general header 
extension, the generic one you mention might be fairly redundant.

I'm certainly open to your thoughts though.

Regards,
Adam

>
> Stephan
>
>
> Any insights are appreciated.
>
> Regards,
> Adam
>
> On 7/19/13 8:33 AM, Stephan Wenger wrote:
>> Hi,
>> I also believe that 16 bits should be enough.  For H.264 and VP8 that 
>> has already been demonstrated.  For H.265, some initial thoughts 
>> below.  Apologies for the word-count.
>>
>> The scalable version of H265 (called SHVC) is currently under 
>> development.  The current working draft can be found here: 
>> http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip. 
>>  Therein, the options for defining layering structures are 
>> considerably more complex.  To start, we have 3 bits for the temporal 
>> ID in the NAL unit header of the H.265 version 1 (HEVC) base 
>> specification (temporal scalability is already nicely supported in 
>> version 1).  Just like in SVC.  In the scalable extension, the NAL 
>> unit header contains a six bit field that points into a data 
>> structure known as "Video Parameter Set" (VPS).  Inside the VPS, 
>> those six bits are mapped to to a position in a directed graph 
>> (specified through "dimension_id[][]"), which tells you about the 
>> reference relationship of the layer in question and its parent layer. 
>>  One can recursively follow the graph to determine what used to be 
>> called dependency_id, quality_id, view_id, and whatnot.  The six bit 
>> pointer field can (or: is to be when possible) organized by the 
>> encoder such that it is prudent for a middle box to throw away NAL 
>> units (belonging to layers) with higher values of the six bit field 
>> first, before throwing away NAL units with lower values.  Relying on 
>> this feature, 3+6 bits == 9 bits should be fine for the header extension.
>>
>> That said, the ordering by the encoder is just a recommendation, and 
>> there may well be cases where different pruning strategies may be 
>> advisable.  For example, a layering structure could be constructed 
>> that expands into two branches, one using 2D scalable tools only, the 
>> other including view_id for multi view coding.  By looking at the six 
>> bit field alone, a middle box will not be able to meaningfully remove 
>> NAL units belonging to one of the branches completely while pruning 
>> the other branch.  In order to meaningfully deal with that scenario, 
>> there would be two options: one to represent the dimension_id[][] 
>> (and associated control info) in the header extension, or require the 
>> middle box to have access to the VPS and be able to interpret its 
>> content.  The further could take considerably more than 16 bits and 
>> we would be talking about a variable length data structure.  The 
>> latter requires the middle box to have state and a mechanism to 
>> intercept the VPS (through signaling—as the encrypted in-band VPS 
>> would not be useful under the assumption that the middle box does not 
>> have the key to the media—which is the motivation of the draft in the 
>> first place).  I personally don't mind at all the second mechanism, 
>> as I'm a big fan of out-of-band parameter set transmission and any 
>> middle box must be in the signaling path anyway to meaningfully 
>> manipulate RTP.  I do not like the first option due to its variable, 
>> and possibly substantial, overhead.
>>
>> Stephan
>>
>>
>> From: Justin Uberti <juberti@google.com <mailto:juberti@google.com>>
>> Date: Friday, 19 July, 2013 06:32
>> To: Bernard Aboba <bernard_aboba@hotmail.com 
>> <mailto:bernard_aboba@hotmail.com>>
>> Cc: "avtext@ietf.org <mailto:avtext@ietf.org>" <avtext@ietf.org 
>> <mailto:avtext@ietf.org>>, "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" 
>> <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>> Subject: Re: [rtcweb] Fwd: New Version Notification for 
>> draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>> Agree those are the right codecs to design for. Since in each case 
>> there are fairly low limits on the number of supported layers (i.e. 3 
>> spatial layers for SVC), I think it should be possible to pack the 
>> temporal, spatial, quality layer ids into 16 bits.
>>
>>
>> On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba 
>> <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>> wrote:
>>
>>     If we can support VP8/9 as well as H.264/5 SVC
>>     that would be a start. It seems doable to me.
>>
>>     On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me
>>     <mailto:fineberg@vline.me>> wrote:
>>
>>>     Bernard,
>>>
>>>     Are there other codecs you are thinking should be supported? If
>>>     it's generalized I would think we want to be able to cover all
>>>     known scalable codecs. I'll look into the H264/SVC fields to see
>>>     how to encode them in a generalized header.
>>>
>>>     Regards,
>>>     Adam
>>>
>>>     On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>>>     I think it may be possible to generalize this. For example, for
>>>>     H.264/SVC which can support temporal, spatial and quality
>>>>     scalability, you would need the quality_id and dependency_id in
>>>>     addition to the temporal_id (what you call the temporal layer
>>>>     index).
>>>>
>>>>     ------------------------------------------------------------------------
>>>>     Date: Thu, 18 Jul 2013 08:45:38 -0700
>>>>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>>     To: bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>
>>>>     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>; avtext@ietf.org
>>>>     <mailto:avtext@ietf.org>
>>>>     Subject: Re: [rtcweb] Fwd: New Version Notification for
>>>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>
>>>>     Bernard,
>>>>
>>>>     Good question.  I'm not familiar enough with the parameter
>>>>     requirements of all other scalable codecs to be able to
>>>>     generalize.  If you'd like to help specify them, I'd be fine
>>>>     revising the draft to generalize.
>>>>
>>>>     Regards,
>>>>     Adam
>>>>
>>>>     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>>>
>>>>         Since the need is not codec specific (e.g. it arises with
>>>>         any codec supporting temporal, spatial and quality
>>>>         scalability), why
>>>>          a VP8-specific RTP extension?
>>>>
>>>>         ------------------------------------------------------------------------
>>>>         Date: Wed, 17 Jul 2013 17:09:46 -0700
>>>>         From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>>         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>>         Subject: [rtcweb] Fwd: New Version Notification for
>>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>
>>>>         Hi,
>>>>
>>>>         I'm working on WebRTC services and have found that while
>>>>         developing services that forward VP8 video streams if we
>>>>         want to take advantage of the VP8 temporal scaling we must
>>>>         get the temporal layer information from the RTP header
>>>>         which requires us to decrypt the SRTP packets. This is
>>>>         undesirable both because the middle-box needs to have
>>>>         access to the keys as well as the because of the added
>>>>         overhead of the decrypt/encrypt cycle. This draft proposes
>>>>         an RTP header extension that will allow us to use the VP8
>>>>         temporal layer information included in the header extension
>>>>         and therefore do forwarding without SRTP decryption.
>>>>         Comments welcome.
>>>>
>>>>         Regards,
>>>>         Adam Fineberg
>>>>         fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>>>>
>>>>         -------- Original Message --------
>>>>         Subject: 	New Version Notification for
>>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>>>>         From: 	internet-drafts at ietf.org
>>>>         <mailto:internet-drafts%20at%20ietf.org>
>>>>         To: 	Adam Fineberg<fineberg at vline.com>
>>>>         <mailto:fineberg%20at%20vline.com>
>>>>
>>>>
>>>>
>>>>         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         has been successfully submitted by Adam Fineberg and posted to the
>>>>         IETF repository.
>>>>
>>>>         Filename:	 draft-fineberg-avtext-temporal-layer-ext
>>>>         Revision:	 00
>>>>         Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>>>>         Creation date:	 2013-07-08
>>>>         Group:		 Individual Submission
>>>>         Number of pages: 6
>>>>         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>>>>         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>>>
>>>>
>>>>         Abstract:
>>>>             This document defines a mechanism by which packets of Real-Time
>>>>             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>>>             indicate, in an RTP header extension, the temporal layer information
>>>>             about the frame encoded in the RTP packet.  This information can be
>>>>             used in a middlebox performing bandwidth management of streams
>>>>             without requiring it to decrypt the streams.
>>>>
>>>>
>>>>         _______________________________________________ rtcweb
>>>>         mailing list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>>         https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>>     -- 
>>>>     Regards,
>>>>     Adam
>>>
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>>
>> _______________________________________________
>> avtext mailing list
>> avtext@ietf.orghttps://www.ietf.org/mailman/listinfo/avtext
>
> -- 
> Regards,
> Adam

-- 
Regards,
Adam


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    Stephan,<br>
    <br>
    <div class="moz-cite-prefix">On 7/19/13 3:44 PM, Stephan Wenger
      wrote:<br>
    </div>
    <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">Hi,</font></div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div style="font-family:Calibri; font-size:11pt;
          text-align:left; color:black; BORDER-BOTTOM: medium none;
          BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT:
          0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;
          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
          <span style="font-weight:bold">From: </span>Adam Fineberg
          &lt;<a moz-do-not-send="true" href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
          <span style="font-weight:bold">Date: </span>Friday, 19 July,
          2013 15:12 <br>
          <span style="font-weight:bold">To: </span>Stephan Wenger &lt;<a
            moz-do-not-send="true" href="mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
          <span style="font-weight:bold">Cc: </span>Justin Uberti &lt;<a
            moz-do-not-send="true" href="mailto:juberti@google.com">juberti@google.com</a>&gt;,
          Bernard Aboba &lt;<a moz-do-not-send="true"
            href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
          <span style="font-weight:bold">Subject: </span>Re: [avtext]
          [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
        </div>
        <div><br>
        </div>
        <div>
          <div bgcolor="#FFFFFF" text="#000000">Stephan,<br>
            <br>
            Thanks for the info and the reference.  I'm not sure I
            follow as I'm not at all familiar with H.265.  I'll review
            the reference and see what I can figure. </div>
        </div>
      </span>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">StW: Good luck :-)  <br>
        </font></div>
    </blockquote>
    <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org" type="cite">
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div>
          <div bgcolor="#FFFFFF" text="#000000">It seems though to me
            that you are suggesting that except in the simple case, that
            the data for H.265 would not be well suited to a header
            extension, am I understanding you correctly?  There is no
            reason the middlebox couldn't get out of band signaling of
            the VPS as you mention but that would not be within the
            scope of this header extension.</div>
        </div>
      </span>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <div><font color="#0000ff"><font face="Calibri,sans-serif">StW:
            well, if you would copy the layer_id into your header
            extension (just as you need to do for the simple case), a
            really smart middle box could use this information just as a
            decoder uses it, assuming that it intercepted the VPS in the
            first place.  Insofar, I wouldn't rule out the second option
            on technical grounds.  Whether any of the actual products
            would bother to do that, ever, is another question.  I think
            the case ought to be documented, though.  I can help
            drafting text.</font></font></div>
    </blockquote>
    <br>
    Any help is appreciated.<br>
    <br>
    <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org" type="cite">
      <div><font color="#0000ff"><font face="Calibri,sans-serif">While
            we are at it: doing this right could mean that you need
            multiple specs.  First, a generic header extension mechanism
            dedicated to side information required for pruning of RTP
            packet streams—ideally not only for scalable video, although
            that is the main customer today.  And second, for each
            "payload" (at present we are talking about H.264/SVC,
            H.265v1 (HEVC), H.265v2 (including scalable and 3D
            extensions, which are not yet finalized), VP8, and VP9 (I
            know little about the latter), plus Daala, and whatnot) a
            mapping of the bits available in the generic header
            extension to the bits in the payload itself (NAL header and
            VPS in case of H.265, NALU header in SVC, and the fields you
            mention for VP8).</font></font></div>
    </blockquote>
    <br>
    I was just thinking about that since it seems like the extension
    payload maybe completely specific and therefore not suitable for the
    same spec.  The problem in my mind with taking the hierarchical
    a[[roach you mention is that I'm not sure what if any common data is
    suitable accross all codecs and since there is already a spec for
    the general header extension, the generic one you mention might be
    fairly redundant.<br>
    <br>
    I'm certainly open to your thoughts though.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org" type="cite">
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><br>
      </div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">Stephan</font></div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div>
          <div bgcolor="#FFFFFF" text="#000000"><br>
            <br>
            Any insights are appreciated.<br>
            <br>
            Regards,<br>
            Adam<br>
            <br>
            <div class="moz-cite-prefix">On 7/19/13 8:33 AM, Stephan
              Wenger wrote:<br>
            </div>
            <blockquote cite="mid:CE0E9F56.9F89B%25stewe@stewe.org"
              type="cite">
              <div>Hi,</div>
              <div>I also believe that 16 bits should be enough.  For
                H.264 and VP8 that has already been demonstrated.  For
                H.265, some initial thoughts below.  Apologies for the
                word-count.</div>
              <div><br>
              </div>
              <div>The scalable version of H265 (called SHVC) is
                currently under development.  The current working draft
                can be found here: <a moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
                 Therein, the options for defining layering structures
                are considerably more complex.  To start, we have 3 bits
                for the temporal ID in the NAL unit header of the H.265
                version 1 (HEVC) base specification (temporal
                scalability is already nicely supported in version 1).
                 Just like in SVC.  In the scalable extension, the NAL
                unit header contains a six bit field that points into a
                data structure known as "Video Parameter Set" (VPS).
                 Inside the VPS, those six bits are mapped to to a
                position in a directed graph (specified through
                "dimension_id[][]"), which tells you about the reference
                relationship of the layer in question and its parent
                layer.  One can recursively follow the graph to
                determine what used to be called dependency_id,
                quality_id, view_id, and whatnot.  The six bit pointer
                field can (or: is to be when possible) organized by the
                encoder such that it is prudent for a middle box to
                throw away NAL units (belonging to layers) with higher
                values of the six bit field first, before throwing away
                NAL units with lower values.  Relying on this feature,
                3+6 bits == 9 bits should be fine for the header
                extension.</div>
              <div><br>
              </div>
              <div>That said, the ordering by the encoder is just a
                recommendation, and there may well be cases where
                different pruning strategies may be advisable.  For
                example, a layering structure could be constructed that
                expands into two branches, one using 2D scalable tools
                only, the other including view_id for multi view coding.
                 By looking at the six bit field alone, a middle box
                will not be able to meaningfully remove NAL units
                belonging to one of the branches completely while
                pruning the other branch.  In order to meaningfully deal
                with that scenario, there would be two options: one to
                represent the dimension_id[][] (and associated control
                info) in the header extension, or require the middle box
                to have access to the VPS and be able to interpret its
                content.  The further could take considerably more than
                16 bits and we would be talking about a variable length
                data structure.  The latter requires the middle box to
                have state and a mechanism to intercept the VPS (through
                signaling—as the encrypted in-band VPS would not be
                useful under the assumption that the middle box does not
                have the key to the media—which is the motivation of the
                draft in the first place).  I personally don't mind at
                all the second mechanism, as I'm a big fan of
                out-of-band parameter set transmission and any middle
                box must be in the signaling path anyway to meaningfully
                manipulate RTP.  I do not like the first option due to
                its variable, and possibly substantial, overhead.</div>
              <div><br>
              </div>
              <div>Stephan   </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <span id="OLK_SRC_BODY_SECTION">
                <div style="font-family:Calibri; font-size:11pt;
                  text-align:left; color:black; BORDER-BOTTOM: medium
                  none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in;
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                  #b5c4df 1pt solid; BORDER-RIGHT: medium none;
                  PADDING-TOP: 3pt">
                  <span style="font-weight:bold">From: </span>Justin
                  Uberti &lt;<a moz-do-not-send="true"
                    href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
                  <span style="font-weight:bold">Date: </span>Friday,
                  19 July, 2013 06:32 <br>
                  <span style="font-weight:bold">To: </span>Bernard
                  Aboba &lt;<a moz-do-not-send="true"
                    href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
                  <span style="font-weight:bold">Cc: </span>"<a
                    moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                  &lt;<a moz-do-not-send="true"
                    href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                  "<a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                  &lt;<a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                  <span style="font-weight:bold">Subject: </span>Re:
                  [rtcweb] Fwd: New Version Notification for
                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                </div>
                <div><br>
                </div>
                <div>
                  <div>
                    <div dir="ltr">Agree those are the right codecs to
                      design for. Since in each case there are fairly
                      low limits on the number of supported layers (i.e.
                      3 spatial layers for SVC), I think it should be
                      possible to pack the temporal, spatial, quality
                      layer ids into 16 bits.
                      <div class="gmail_extra"><br>
                        <br>
                        <div class="gmail_quote">On Fri, Jul 19, 2013 at
                          1:56 AM, Bernard Aboba <span dir="ltr">
                            &lt;<a moz-do-not-send="true"
                              href="mailto:bernard_aboba@hotmail.com"
                              target="_blank">bernard_aboba@hotmail.com</a>&gt;</span>
                          wrote:<br>
                          <blockquote class="gmail_quote"
                            style="margin:0 0 0 .8ex;border-left:1px
                            #ccc solid;padding-left:1ex">
                            <div dir="auto">
                              <div>If we can support VP8/9 as well as
                                H.264/5 SVC</div>
                              <div>that would be a start. It seems
                                doable to me.</div>
                              <div>
                                <div>
                                  <div><br>
                                    On Jul 18, 2013, at 8:34 PM, "Adam
                                    Fineberg" &lt;<a
                                      moz-do-not-send="true"
                                      href="mailto:fineberg@vline.me"
                                      target="_blank">fineberg@vline.me</a>&gt;
                                    wrote:<br>
                                    <br>
                                  </div>
                                  <blockquote type="cite">
                                    <div>
                                      <div>Bernard,<br>
                                        <br>
                                        Are there other codecs you are
                                        thinking should be supported? 
                                        If it's generalized I would
                                        think we want to be able to
                                        cover all known scalable codecs.
                                        I'll look into the H264/SVC
                                        fields to see how to encode them
                                        in a generalized header.<br>
                                        <br>
                                        Regards,<br>
                                        Adam<br>
                                        <br>
                                        On 7/18/13 7:40 PM, Bernard
                                        Aboba wrote:<br>
                                      </div>
                                      <blockquote type="cite">
                                        <div dir="ltr">I think it may be
                                          possible to generalize this. 
                                          For example, for H.264/SVC
                                          which can support temporal,
                                          spatial and quality
                                          scalability, you would need
                                          the quality_id and
                                          dependency_id in addition to
                                          the temporal_id (what you call
                                          the temporal layer index).   
                                          <br>
                                          <br>
                                          <div>
                                            <hr>
                                            Date: Thu, 18 Jul 2013
                                            08:45:38 -0700<br>
                                            From: <a
                                              moz-do-not-send="true"
                                              href="mailto:fineberg@vline.me"
                                              target="_blank">fineberg@vline.me</a><br>
                                            To: <a
                                              moz-do-not-send="true"
                                              href="mailto:bernard_aboba@hotmail.com"
                                              target="_blank">
                                              bernard_aboba@hotmail.com</a><br>
                                            CC: <a
                                              moz-do-not-send="true"
                                              href="mailto:rtcweb@ietf.org"
                                              target="_blank">rtcweb@ietf.org</a>;
                                            <a moz-do-not-send="true"
                                              href="mailto:avtext@ietf.org"
                                              target="_blank">avtext@ietf.org</a><br>
                                            Subject: Re: [rtcweb] Fwd:
                                            New Version Notification for
draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                            <br>
                                            Bernard,<br>
                                            <br>
                                            Good question.  I'm not
                                            familiar enough with the
                                            parameter requirements of
                                            all other scalable codecs to
                                            be able to generalize.  If
                                            you'd like to help specify
                                            them, I'd be fine revising
                                            the draft to generalize.<br>
                                            <br>
                                            Regards,<br>
                                            Adam<br>
                                            <br>
                                            <div>On 7/17/13 8:26 PM,
                                              Bernard Aboba wrote:<br>
                                            </div>
                                            <blockquote>
                                              <div dir="ltr">Since the
                                                need is not codec
                                                specific (e.g. it arises
                                                with any codec
                                                supporting temporal,
                                                spatial and quality
                                                scalability), why
                                                <br>
                                                 a VP8-specific RTP
                                                extension? <br>
                                                 <br>
                                                <div>
                                                  <hr>
                                                  Date: Wed, 17 Jul 2013
                                                  17:09:46 -0700<br>
                                                  From: <a
                                                    moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                  To: <a
                                                    moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                  Subject: [rtcweb] Fwd:
                                                  New Version
                                                  Notification for
                                                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                  <br>
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">I'm
                                                    working on WebRTC
                                                    services and have
                                                    found that while
                                                    developing services
                                                    that forward VP8
                                                    video streams if we
                                                    want to take
                                                    advantage of the VP8
                                                    temporal scaling we
                                                    must get the
                                                    temporal layer
                                                    information from the
                                                    RTP header which
                                                    requires us to
                                                    decrypt the SRTP
                                                    packets. This is
                                                    undesirable both
                                                    because the
                                                    middle-box needs to
                                                    have access to the
                                                    keys as well as the
                                                    because of the added
                                                    overhead of the
                                                    decrypt/encrypt
                                                    cycle. This draft
                                                    proposes an RTP
                                                    header extension
                                                    that will allow us
                                                    to use the VP8
                                                    temporal layer
                                                    information included
                                                    in the header
                                                    extension and
                                                    therefore do
                                                    forwarding without
                                                    SRTP decryption.
                                                    Comments welcome.</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Regards,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Adam
                                                    Fineberg</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <div
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px"><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                      rel="nofollow"
                                                      target="_blank">fineberg
                                                      at vline.com</a><br>
                                                    <br>
                                                    -------- Original
                                                    Message --------
                                                    <table border="0"
                                                      cellpadding="0"
                                                      cellspacing="0">
                                                      <tbody>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Subject:</th>
                                                          <td>New
                                                          Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Date:</th>
                                                          <td>Tue, 09
                                                          Jul 2013
                                                          10:02:05 -0700</td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">From:</th>
                                                          <td><a
                                                          moz-do-not-send="true"
href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow"
                                                          target="_blank">internet-drafts
                                                          at ietf.org</a></td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">To:</th>
                                                          <td>Adam
                                                          Fineberg<span> </span><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          rel="nofollow"
target="_blank">&lt;fineberg at vline.com&gt;</a></td>
                                                        </tr>
                                                      </tbody>
                                                    </table>
                                                    <br>
                                                    <br>
                                                    <pre style="width:939.59px;white-space:pre-wrap">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel="nofollow" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel="nofollow" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" rel="nofollow" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                                                  </div>
                                                  <br>
                                                  _______________________________________________
                                                  rtcweb mailing list <a
moz-do-not-send="true" href="mailto:rtcweb@ietf.org" target="_blank">
                                                    rtcweb@ietf.org</a>
                                                  <a
                                                    moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
                                              </div>
                                            </blockquote>
                                            <br>
                                            <pre>-- 
Regards,
Adam</pre>
                                          </div>
                                        </div>
                                      </blockquote>
                                      <br>
                                    </div>
                                  </blockquote>
                                </div>
                              </div>
                            </div>
                            <br>
_______________________________________________<br>
                            rtcweb mailing list<br>
                            <a moz-do-not-send="true"
                              href="mailto:rtcweb@ietf.org"
                              target="_blank">rtcweb@ietf.org</a><br>
                            <a moz-do-not-send="true"
                              href="https://www.ietf.org/mailman/listinfo/rtcweb"
                              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                            <br>
                          </blockquote>
                        </div>
                        <br>
                      </div>
                    </div>
                  </div>
                </div>
              </span><br>
              <fieldset class="mimeAttachmentHeader"></fieldset>
              <br>
              <pre wrap="">_______________________________________________
avtext mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/avtext">https://www.ietf.org/mailman/listinfo/avtext</a></pre>
            </blockquote>
            <br>
            <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
          </div>
        </div>
      </span>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
  </body>
</html>

--------------010603030307070406090203--

From fluffy@iii.ca  Fri Jul 19 17:52:43 2013
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Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Jul 19, 2013, at 9:54 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> Negotiation is a hole.  A vast, soul-sucking, waste of time.

That's might be closer to true when you control both ends, like Skype.

It's not true when a browser from one vendor running an application from =
a scone vendor needs to talk to video conferencing system from a third =
vendor.=20



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On Jul 19, 2013, at 10:18 AM, Peter Thatcher <pthatcher@google.com> =
wrote:

>=20
> It's interesting that most or your list of things that needed to be =
solved without SDP (simulcast, FEC, correlation of RTP streams with =
MediaStreamTracks, glare) still haven't been solved for WebRTC even with =
SDP, despite many months (years?) of effort. =20
>=20

Peter, with the exception of Simulcast, which of these do you think has =
not been solved in SDP when not using bundle?=20





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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 17:56:45 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 5:54 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jul 19, 2013, at 10:18 AM, Peter Thatcher <pthatcher@google.com> wrote:
>
> >
> > It's interesting that most or your list of things that needed to be
> solved without SDP (simulcast, FEC, correlation of RTP streams with
> MediaStreamTracks, glare) still haven't been solved for WebRTC even with
> SDP, despite many months (years?) of effort.
> >
>
> Peter, with the exception of Simulcast, which of these do you think has
> not been solved in SDP when not using bundle?
>
>
>
Has FEC been defined for WebRTC?
Has glare been solved?
Has mapping of RTP streams with MediaStreamTracks been resolved (with the
exception of the "unity plan" which has not yet been approved)?

I think the whole list I gave is still unsolved/unresolved.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 5:54 PM, Cullen Jennings <span dir=3D"ltr">=
&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Jul 19, 2013, at 10:18 AM, Peter Thatcher &lt;<a href=3D"mailto:pthatche=
r@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt; It&#39;s interesting that most or your list of things that needed to b=
e solved without SDP (simulcast, FEC, correlation of RTP streams with Media=
StreamTracks, glare) still haven&#39;t been solved for WebRTC even with SDP=
, despite many months (years?) of effort.<br>


&gt;<br>
<br>
</div>Peter, with the exception of Simulcast, which of these do you think h=
as not been solved in SDP when not using bundle?<br>
<br>
<br></blockquote><div><br></div><div>Has FEC been defined for WebRTC? =C2=
=A0</div><div>Has glare been solved?</div><div>Has mapping of RTP streams w=
ith MediaStreamTracks been resolved (with the exception of the &quot;unity =
plan&quot; which has not yet been approved)?</div>

<div><br></div><div>I think the whole list I gave is still unsolved/unresol=
ved.=C2=A0<br></div><div><br></div><div>=C2=A0</div></div><br></div></div>

--047d7b6705b938996004e1e6ed2f--

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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Jul 19, 2013, at 11:40 AM, Matthew Kaufman (SKYPE) =
<matthew.kaufman@skype.net> wrote:

> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>>=20
>> Negotiation is a hole.  A vast, soul-sucking, waste of time.
>=20
> And negotiation over SDP requires a trip to MMUSIC every time you =
think of something new, too. That's a soul-sucking waste of time =
*itself*.
>=20
> Matthew Kaufman

Better than a trip to W3C, IETF, and XSF


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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 18:00:49 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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The logic of how to talk to the video conferencing system doesn't need to
be backed into the browser.  If the API provides enough control the JS, the
web app can contain the logic to talk to the video conferencing system.

I know I'm going to start sounding like a broken record, but you brought up
video conferencing systems as an example, so I will say it again: there are
are video conferencing systems that don't use SDP for signalling.  For
example, there are video conferencing systems that use Jingle for
signalling.  Would I expect the logic of how to speak to that video
conferencing system to be baked into the browser?  No, I think I'd prefer
it to be built into a web app built on top of a good API.  Why should
SDP-based video conferencing systems be treated special?


On Fri, Jul 19, 2013 at 5:52 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jul 19, 2013, at 9:54 AM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> > Negotiation is a hole.  A vast, soul-sucking, waste of time.
>
> That's might be closer to true when you control both ends, like Skype.
>
> It's not true when a browser from one vendor running an application from a
> scone vendor needs to talk to video conferencing system from a third vendor.
>
>
>
>

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<div dir=3D"ltr">The logic of how to talk to the video conferencing system =
doesn&#39;t need to be backed into the browser. =C2=A0If the API provides e=
nough control the JS, the web app can contain the logic to talk to the vide=
o conferencing system. =C2=A0<div>

<br></div><div>I know I&#39;m going to start sounding like a broken record,=
 but you brought up video conferencing systems as an example, so I will say=
 it again: there are are video conferencing systems that don&#39;t use SDP =
for signalling. =C2=A0For example, there are video conferencing systems tha=
t use Jingle for signalling. =C2=A0Would I expect the logic of how to speak=
 to that video conferencing system to be baked into the browser? =C2=A0No, =
I think I&#39;d prefer it to be built into a web app built on top of a good=
 API. =C2=A0Why should SDP-based video conferencing systems be treated spec=
ial?</div>

<div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, =
Jul 19, 2013 at 5:52 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">

<div class=3D"im"><br>
On Jul 19, 2013, at 9:54 AM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
<br>
&gt; Negotiation is a hole. =C2=A0A vast, soul-sucking, waste of time.<br>
<br>
</div>That&#39;s might be closer to true when you control both ends, like S=
kype.<br>
<br>
It&#39;s not true when a browser from one vendor running an application fro=
m a scone vendor needs to talk to video conferencing system from a third ve=
ndor.<br>
<br>
<br>
<br>
</blockquote></div><br></div></div></div>

--e89a8ff24319db461b04e1e6fbfd--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 18:01:24 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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On Fri, Jul 19, 2013 at 5:57 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jul 19, 2013, at 11:40 AM, Matthew Kaufman (SKYPE) <
> matthew.kaufman@skype.net> wrote:
>
> > From: Martin Thomson [mailto:martin.thomson@gmail.com]
> >>
> >> Negotiation is a hole.  A vast, soul-sucking, waste of time.
> >
> > And negotiation over SDP requires a trip to MMUSIC every time you think
> of something new, too. That's a soul-sucking waste of time *itself*.
> >
> > Matthew Kaufman
>
> Better than a trip to W3C, IETF, and XSF
>

That's definitely a matter of opinion.


>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 5:57 PM, Cullen Jennings <span dir=3D"ltr">=
&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Jul 19, 2013, at 11:40 AM, Matthew Kaufman (SKYPE) &lt;<a href=3D"mailto=
:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a>&gt; wrote:<br>
<br>
&gt; From: Martin Thomson [mailto:<a href=3D"mailto:martin.thomson@gmail.co=
m">martin.thomson@gmail.com</a>]<br>
&gt;&gt;<br>
&gt;&gt; Negotiation is a hole. =C2=A0A vast, soul-sucking, waste of time.<=
br>
&gt;<br>
&gt; And negotiation over SDP requires a trip to MMUSIC every time you thin=
k of something new, too. That&#39;s a soul-sucking waste of time *itself*.<=
br>
&gt;<br>
&gt; Matthew Kaufman<br>
<br>
</div>Better than a trip to W3C, IETF, and XSF<br></blockquote><div><br></d=
iv><div>That&#39;s definitely a matter of opinion.</div><div>=C2=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">


<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--bcaec520f3b9d7cd2004e1e6fd2e--

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On Jul 19, 2013, at 5:56 PM, Peter Thatcher <pthatcher@google.com> =
wrote:

>=20
>=20
>=20
> On Fri, Jul 19, 2013 at 5:54 PM, Cullen Jennings <fluffy@iii.ca> =
wrote:
>=20
> On Jul 19, 2013, at 10:18 AM, Peter Thatcher <pthatcher@google.com> =
wrote:
>=20
> >
> > It's interesting that most or your list of things that needed to be =
solved without SDP (simulcast, FEC, correlation of RTP streams with =
MediaStreamTracks, glare) still haven't been solved for WebRTC even with =
SDP, despite many months (years?) of effort.
> >
>=20
> Peter, with the exception of Simulcast, which of these do you think =
has not been solved in SDP when not using bundle?
>=20
>=20
>=20
I asked about SDP, not WebRTC. The thing that is making this take a long =
time is you don't want to use SDP.=20

> Has FEC been defined for WebRTC? =20
yes, that is defined for SDP O/A

> Has glare been solved?
yes

> Has mapping of RTP streams with MediaStreamTracks been resolved (with =
the exception of the "unity plan" which has not yet been approved)?
This is obviously trivial if not using bundle. Any version of MSID would =
work. It's only complicated by bundle

>=20
> I think the whole list I gave is still unsolved/unresolved.=20
I disagree. They are only unresolved because we want to add bundle and =
that was a major change to RTP resulting in a bunch of work needing to =
be done to see how that impacted SDP.=20



>=20
> =20
>=20


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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Thanks,=20

I will note that over half of that very small sample set is folks that =
have been long termed involved in SIP and VoIP so, uh, might not be a =
representative sample of type of people you are portraying it as.=20


On Jul 19, 2013, at 7:51 AM, Peter Thatcher <pthatcher@google.com> =
wrote:

> Here you go:
>=20
>=20
>=20
>=20
> On Fri, Jul 19, 2013 at 7:00 AM, Cullen Jennings <fluffy@iii.ca> =
wrote:
> Can someone please email a full copy of the spreadsheet to the list so =
it is in the archives - thanks
>=20
> On Jul 9, 2013, at 3:18 AM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:
>=20
> > 2013/7/9 Stefan H=E5kansson LK <stefan.lk.hakansson@ericsson.com>:
> >> I want to be very clear and careful on what I say. So I am =
repeating:
> >>
> >> * My comment that I think Eric is right in that there is consensus =
on
> >> providing APIs that allow most use-cases to be met without SDP =
mangling
> >> is meant only in that context: SDP O/A is kept (and PeerConnection =
more
> >> or less as is and so on).
> >
> > Hi Stefan,
> >
> > You insist that "there is consensus on keeping SDP O/A but providing =
a
> > better API". Given current discussions IMHO it is clear there is not
> > such a consensus (not at all). Or may be you are just talking about
> > two years ago in some IETF meeting (if so I'm sorry).
> >
> > Please review the results in
> > =
https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHl=
ZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1
> >
> >  Bad for advanced stuff: 94%
> >  OK for 1.0: 40%
> >
> > IMHO such a result indicates all but "let's keep SDP O/A and improve =
a
> > bit the API" (I mean now, in July 2013).
> >
> >
> > Best regards.
> >
> >
> >
> > --
> > I=F1aki Baz Castillo
> > <ibc@aliax.net>
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> <Input from JS Developers on their experience with the WebRTC API - =
Inputs.csv>


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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 18:20:40 -0700
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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--bcaec51f99e5c84e1c04e1e74297
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On Fri, Jul 19, 2013 at 6:10 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jul 19, 2013, at 5:56 PM, Peter Thatcher <pthatcher@google.com> wrote:
>
> >
> >
> >
> > On Fri, Jul 19, 2013 at 5:54 PM, Cullen Jennings <fluffy@iii.ca> wrote:
> >
> > On Jul 19, 2013, at 10:18 AM, Peter Thatcher <pthatcher@google.com>
> wrote:
> >
> > >
> > > It's interesting that most or your list of things that needed to be
> solved without SDP (simulcast, FEC, correlation of RTP streams with
> MediaStreamTracks, glare) still haven't been solved for WebRTC even with
> SDP, despite many months (years?) of effort.
> > >
> >
> > Peter, with the exception of Simulcast, which of these do you think has
> not been solved in SDP when not using bundle?
> >
> >
> >
> I asked about SDP, not WebRTC. The thing that is making this take a long
> time is you don't want to use SDP.
>
>
I'm not asking about SDP.  I'm asking about WebRTC.   It's the WebRTC API
I'm concerned with.

 > Has FEC been defined for WebRTC?
> yes, that is defined for SDP O/A
>

FEC still isn't defined for use with the WebRTC API.  I believe it's still
in the long list of things that can't be resolved until Plan A vs. Plan B
is resolved.

> Has glare been solved?
> yes
>
>
In the context of WebRTC, it wasn't at the last IETF.  What has changed
since then?


>  > Has mapping of RTP streams with MediaStreamTracks been resolved (with
> the exception of the "unity plan" which has not yet been approved)?
> This is obviously trivial if not using bundle. Any version of MSID would
> work. It's only complicated by bundle
>
>
Even without BUNDLE, it's still not clear what kind of SDP createOffer
should generate for multiple tracks.  (with the exception of the "unity
plan" which has not yet been approved).


> >
> > I think the whole list I gave is still unsolved/unresolved.
> I disagree. They are only unresolved because we want to add bundle and
> that was a major change to RTP resulting in a bunch of work needing to be
> done to see how that impacted SDP.
>
>
It thinks there's more than just BUNDLE, but you make a good point:
 finishing the WebRTC API currently requires lots of major changes to SDP,
which creates big delays.  In fact, it's the number one things holding back
the API from being done, because major changes to SDP take a really, really
long time.


>
>
> >
> >
> >
>
>

--bcaec51f99e5c84e1c04e1e74297
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jul 19, 2013 at 6:10 PM, Cullen Jennings <span dir=3D"ltr">=
&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im"><br>
On Jul 19, 2013, at 5:56 PM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Jul 19, 2013 at 5:54 PM, Cullen Jennings &lt;<a href=3D"mailto=
:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;<br>
&gt; On Jul 19, 2013, at 10:18 AM, Peter Thatcher &lt;<a href=3D"mailto:pth=
atcher@google.com">pthatcher@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt;<br>
&gt; &gt; It&#39;s interesting that most or your list of things that needed=
 to be solved without SDP (simulcast, FEC, correlation of RTP streams with =
MediaStreamTracks, glare) still haven&#39;t been solved for WebRTC even wit=
h SDP, despite many months (years?) of effort.<br>


&gt; &gt;<br>
&gt;<br>
&gt; Peter, with the exception of Simulcast, which of these do you think ha=
s not been solved in SDP when not using bundle?<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div>I asked about SDP, not WebRTC. The thing that is making this take a l=
ong time is you don&#39;t want to use SDP.<br>
<div class=3D"im"><br></div></blockquote><div><br></div><div>I&#39;m not as=
king about SDP. =C2=A0I&#39;m asking about WebRTC. =C2=A0 It&#39;s the WebR=
TC API I&#39;m concerned with.</div><div><br></div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-le=
ft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div class=3D"im">
&gt; Has FEC been defined for WebRTC?<br>
</div>yes, that is defined for SDP O/A<br></blockquote><div><br></div><div>=
FEC still isn&#39;t defined for use with the WebRTC API. =C2=A0I believe it=
&#39;s still in the long list of things that can&#39;t be resolved until Pl=
an A vs. Plan B is resolved.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-lef=
t-style:solid;padding-left:1ex">
&gt; Has glare been solved?<br>
yes<br>
<div class=3D"im"><br></div></blockquote><div><br></div><div>In the context=
 of WebRTC, it wasn&#39;t at the last IETF. =C2=A0What has changed since th=
en?</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204)=
;border-left-style:solid;padding-left:1ex">

<div class=3D"im">
&gt; Has mapping of RTP streams with MediaStreamTracks been resolved (with =
the exception of the &quot;unity plan&quot; which has not yet been approved=
)?<br>
</div>This is obviously trivial if not using bundle. Any version of MSID wo=
uld work. It&#39;s only complicated by bundle<br>
<div class=3D"im"><br></div></blockquote><div><br></div><div>Even without B=
UNDLE, it&#39;s still not clear what kind of SDP createOffer should generat=
e for multiple tracks. =C2=A0<span style=3D"color:rgb(80,0,80)">(with the e=
xception of the &quot;unity plan&quot; which has not yet been approved).</s=
pan></div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex"><div class=3D"im">
&gt;<br>
&gt; I think the whole list I gave is still unsolved/unresolved.<br>
</div>I disagree. They are only unresolved because we want to add bundle an=
d that was a major change to RTP resulting in a bunch of work needing to be=
 done to see how that impacted SDP.<br>
<br></blockquote><div><br></div><div>It thinks there&#39;s more than just B=
UNDLE, but you make a good point: =C2=A0finishing the WebRTC API currently =
requires lots of major changes to SDP, which creates big delays. =C2=A0In f=
act, it&#39;s the number one things holding back the API from being done, b=
ecause major changes to SDP take a really, really long time.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">
<br>
<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
</blockquote></div><br></div></div>

--bcaec51f99e5c84e1c04e1e74297--

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 19 Jul 2013 18:24:11 -0700
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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You're surprised that the first web developers that try out WebRTC and have
feedback have experience with SIP and VoIP?  We shouldn't find that
surprising, nor should we discount their feedback because of it.

In any case, it's the best data we've got so far.  I made an effort to some
data because no one else had.  If you would like to make an effort to get
better feedback, I'd more than welcome it.  In fact, I'd say you ought not
to have waited for me to do it.


On Fri, Jul 19, 2013 at 6:20 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Thanks,
>
> I will note that over half of that very small sample set is folks that
> have been long termed involved in SIP and VoIP so, uh, might not be a
> representative sample of type of people you are portraying it as.
>
>
> On Jul 19, 2013, at 7:51 AM, Peter Thatcher <pthatcher@google.com> wrote:
>
> > Here you go:
> >
> >
> >
> >
> > On Fri, Jul 19, 2013 at 7:00 AM, Cullen Jennings <fluffy@iii.ca> wrote:
> > Can someone please email a full copy of the spreadsheet to the list so
> it is in the archives - thanks
> >
> > On Jul 9, 2013, at 3:18 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:
> >
> > > 2013/7/9 Stefan H=C3=A5kansson LK <stefan.lk.hakansson@ericsson.com>:
> > >> I want to be very clear and careful on what I say. So I am repeating=
:
> > >>
> > >> * My comment that I think Eric is right in that there is consensus o=
n
> > >> providing APIs that allow most use-cases to be met without SDP
> mangling
> > >> is meant only in that context: SDP O/A is kept (and PeerConnection
> more
> > >> or less as is and so on).
> > >
> > > Hi Stefan,
> > >
> > > You insist that "there is consensus on keeping SDP O/A but providing =
a
> > > better API". Given current discussions IMHO it is clear there is not
> > > such a consensus (not at all). Or may be you are just talking about
> > > two years ago in some IETF meeting (if so I'm sorry).
> > >
> > > Please review the results in
> > >
> https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdH=
lZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1
> > >
> > >  Bad for advanced stuff: 94%
> > >  OK for 1.0: 40%
> > >
> > > IMHO such a result indicates all but "let's keep SDP O/A and improve =
a
> > > bit the API" (I mean now, in July 2013).
> > >
> > >
> > > Best regards.
> > >
> > >
> > >
> > > --
> > > I=C3=B1aki Baz Castillo
> > > <ibc@aliax.net>
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > <Input from JS Developers on their experience with the WebRTC API -
> Inputs.csv>
>
>

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<div dir=3D"ltr"><div>You&#39;re surprised that the first web developers th=
at try out WebRTC and have feedback have experience with SIP and VoIP? =C2=
=A0We shouldn&#39;t find that surprising, nor should we discount their feed=
back because of it.</div>

<div><br></div>In any case, it&#39;s the best data we&#39;ve got so far. =
=C2=A0I made an effort to some data because no one else had. =C2=A0If you w=
ould like to make an effort to get better feedback, I&#39;d more than welco=
me it. =C2=A0In fact, I&#39;d say you ought not to have waited for me to do=
 it.<br>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Jul 1=
9, 2013 at 6:20 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto=
:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">

<br>
Thanks,<br>
<br>
I will note that over half of that very small sample set is folks that have=
 been long termed involved in SIP and VoIP so, uh, might not be a represent=
ative sample of type of people you are portraying it as.<br>
<div><div class=3D"h5"><br>
<br>
On Jul 19, 2013, at 7:51 AM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt; Here you go:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Jul 19, 2013 at 7:00 AM, Cullen Jennings &lt;<a href=3D"mailto=
:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt; Can someone please email a full copy of the spreadsheet to the list so=
 it is in the archives - thanks<br>
&gt;<br>
&gt; On Jul 9, 2013, at 3:18 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mai=
lto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; 2013/7/9 Stefan H=C3=A5kansson LK &lt;<a href=3D"mailto:stefan.lk=
.hakansson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;:<br>
&gt; &gt;&gt; I want to be very clear and careful on what I say. So I am re=
peating:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; * My comment that I think Eric is right in that there is cons=
ensus on<br>
&gt; &gt;&gt; providing APIs that allow most use-cases to be met without SD=
P mangling<br>
&gt; &gt;&gt; is meant only in that context: SDP O/A is kept (and PeerConne=
ction more<br>
&gt; &gt;&gt; or less as is and so on).<br>
&gt; &gt;<br>
&gt; &gt; Hi Stefan,<br>
&gt; &gt;<br>
&gt; &gt; You insist that &quot;there is consensus on keeping SDP O/A but p=
roviding a<br>
&gt; &gt; better API&quot;. Given current discussions IMHO it is clear ther=
e is not<br>
&gt; &gt; such a consensus (not at all). Or may be you are just talking abo=
ut<br>
&gt; &gt; two years ago in some IETF meeting (if so I&#39;m sorry).<br>
&gt; &gt;<br>
&gt; &gt; Please review the results in<br>
&gt; &gt; <a href=3D"https://docs.google.com/a/aliax.net/spreadsheet/ccc?ke=
y=3D0AuaKXw3SkHMSdHlZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1" target=3D"_blank"=
>https://docs.google.com/a/aliax.net/spreadsheet/ccc?key=3D0AuaKXw3SkHMSdHl=
ZdV9RN0xSWFhybVl4anJLRkVPV0E#gid=3D1</a><br>


&gt; &gt;<br>
&gt; &gt; =C2=A0Bad for advanced stuff: 94%<br>
&gt; &gt; =C2=A0OK for 1.0: 40%<br>
&gt; &gt;<br>
&gt; &gt; IMHO such a result indicates all but &quot;let&#39;s keep SDP O/A=
 and improve a<br>
&gt; &gt; bit the API&quot; (I mean now, in July 2013).<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Best regards.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; I=C3=B1aki Baz Castillo<br>
&gt; &gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
</div></div>&gt; &lt;Input from JS Developers on their experience with the =
WebRTC API - Inputs.csv&gt;<br>
<br>
</blockquote></div><br></div></div>

--047d7b111cff5436cd04e1e74fe6--

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On Jul 19, 2013, at 8:03 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/7/19 Cullen Jennings <fluffy@iii.ca>:
>> Folks wanted to sort out how mindless works fist before getting to =
theses smaller details
>=20
> Hi Cullen,
>=20
> All these "smaller details" are in fact issues and problems we don't
> need. Issues not needed at all in WebRTC, issues that WebRTC
> developers and vendors should NOT care about. If those "smaller
> details" do exist is due the mandate of SDP. And all the time the WG
> (or WGs) will spent "fixing/defining" them is just wasted time, since
> nothing useful is being done for WebRTC by "defining" those "minor
> details".
>=20
> Regards.
>=20


Well lets prioritize fixing them. Which ones are causing =
interoperability problems between browsers?=20



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Well that was clearly one or my more mindless emails=20

What I mean to say was something along lines of=20

Folks want to sort out the multiplexing / bundle stuff first.=20

So let me suggest that folks read section 6 of the JSEP draft and see if =
they can answer most of of the questions below. I'm not saying that =
section 6 is right, in fact I think some of it is wrong. But at least =
act like you have read it.


On Jul 19, 2013, at 7:49 AM, Peter Thatcher <pthatcher@google.com> =
wrote:

> "mindless works fist"?  There's a deep meaning there, but I'm still =
trying to figure it out...
>=20
>=20
> On Fri, Jul 19, 2013 at 7:04 AM, Cullen Jennings <fluffy@iii.ca> =
wrote:
> Folks wanted to sort out how mindless works fist before getting to =
theses smaller details but in general I think progress is being made on =
all of these
>=20
> On Jul 17, 2013, at 2:58 PM, "Matthew Kaufman (SKYPE)" =
<matthew.kaufman@skype.net> wrote:
>=20
> > Bernard Aboba:
> >>
> >> The problem was that we never defined what mangling browsers had to
> >> support. Given all the SDP specs that is actually a huge work item.
> >
> >
> > This is exactly what my slides that weren't presented last November =
started to touch upon. It only takes a few minutes with RFC3264 and its =
references to start documenting cases that are clearly "valid SDP =
offer/answer" and yet for which I cannot for the life of me figure out =
what a browser should do if they're presented.
> >
> > Just off the top of my head:
> >
> > Can I...
> > - Change the t=3D line to be something other than t=3D0 0?

Current JSEP draft says the browser does not need to allow that to be =
changed.

> > - Change the rtpmap associations before calling setLocal?

you can remove or reorder

> > - Change a=3Dsendrecv to a=3Drecvonly before calling setLocal?

Current JSEP draft says that is done with constraints

> > - What do you do when you see a=3Dcontent:sl ?

not in current spec but you can find dissection on list about making =
sure that is available to API=20

> > - What if someone adds an r=3D or p=3D or e=3D?

Current JSEP draft says the browser does not need to allow that=20

> - What is the RFC that describes a=3Dgroup:BUNDLE (as seen in some =
browser implementations)?

Uh, I'll leave that as an exercise to the readers. Try hard and see if =
you can find it.=20


> > - Can I remove a=3Dgroup:BUNDLE (or add it) before calling setLocal?

Current JSEP draft says that is done with constraints

> > - How about removing a=3Drtcp-mux?

Current JSEP draft says that is done with constraints


> > - Should I do something special at my end if you set =
a=3Dice-options:google-ice ?

Do whatever the ice spec say=20

> > - If you put a=3Dssrc lines in there, can I change the ssrc before =
passing it back to setLocal?
Current JSEP draft says the browser does not need to allow that to be =
changed.


> > - Can I delete the a=3Dssrc lines?

Current JSEP draft says the browser does not need to allow that to be =
changed.


> > - Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?

Why would a browser want to generate that?

> > - What about 0.0.0.0 in the o=3D line?

Why would a browser want to generate that?

> > - If I get a bunch of a=3Dcandidate lines, can I swap them around to =
change the priority before calling setLocal?

Current JSEP draft says the browser does not need to allow that to be =
changed.

> > - What if someone claims SAVP instead of SAVPF but gives me rtcp =
info?

We need to add text to explicitly cover that. You can find text on what =
I think we should do for this in=20
draft-jennings-rtcweb-plan-01


> >
> > At the *very least* for each and every line that comes from =
createOffer and createAnswer you must be able to answer the following:
> > - Can I delete it?
> > - Duplicate it?
> > - Change it?
> > - If not, how are violation handled? (Both when passed to another =
browser at the far end and when these modifications happen before =
calling setLocal)
> > - And can I add additional valid SDP to what came from createOffer =
or createAnswer and pass it back to setLocal or not?
> >
> > We appear to be nowhere near a document which explains the answers =
to these questions, certainly not in the W3C WG, and not in any IETF RFC =
or draft I can locate.

draft-ietf-rtcweb-jsep-03 has tried to address some of these. I =
certainly don't claim it is perfect but if people have use cases that =
suggest a change to this, love to get text on what it should be.=20

Mathew has been very clear along the way of "we need to define X to have =
an interoperable system" and for most values of X he has proposed, I =
agree with him. However, I think we just need to do that and that it =
actually won't be much work once we get the framework in place to deal =
with thinks like how many ports is the RTP using and how many m-lines =
does the SDP have.=20

> >
> > Matthew Kaufman
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: Cullen Jennings <fluffy@iii.ca>
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To: Peter Thatcher <pthatcher@google.com>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Jul 19, 2013, at 6:24 PM, Peter Thatcher <pthatcher@google.com> =
wrote:

> You're surprised that the first web developers that try out WebRTC and =
have feedback have experience with SIP and VoIP?  We shouldn't find that =
surprising, nor should we discount their feedback because of it.

I was not saying that feedback from folks like Bernard Aboba was not =
useful, of course it can be. I was just noting the sample set.=20=

From pthatcher@google.com  Fri Jul 19 18:56:52 2013
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From: Peter Thatcher <pthatcher@google.com>
To: Cullen Jennings <fluffy@iii.ca>
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On Jul 19, 2013 6:47 PM, "Cullen Jennings" <fluffy@iii.ca> wrote:
>
>
> Well that was clearly one or my more mindless emails
>
> What I mean to say was something along lines of
>
> Folks want to sort out the multiplexing / bundle stuff first.
>
> So let me suggest that folks read section 6 of the JSEP draft and see if
they can answer most of of the questions below. I'm not saying that section
6 is right, in fact I think some of it is wrong. But at least act like you
have read it.
>
>
> On Jul 19, 2013, at 7:49 AM, Peter Thatcher <pthatcher@google.com> wrote:
>
> > "mindless works fist"?  There's a deep meaning there, but I'm still
trying to figure it out...
> >
> >
> > On Fri, Jul 19, 2013 at 7:04 AM, Cullen Jennings <fluffy@iii.ca> wrote:
> > Folks wanted to sort out how mindless works fist before getting to
theses smaller details but in general I think progress is being made on all
of these
> >
> > On Jul 17, 2013, at 2:58 PM, "Matthew Kaufman (SKYPE)" <
matthew.kaufman@skype.net> wrote:
> >
> > > Bernard Aboba:
> > >>
> > >> The problem was that we never defined what mangling browsers had to
> > >> support. Given all the SDP specs that is actually a huge work item.
> > >
> > >
> > > This is exactly what my slides that weren't presented last November
started to touch upon. It only takes a few minutes with RFC3264 and its
references to start documenting cases that are clearly "valid SDP
offer/answer" and yet for which I cannot for the life of me figure out what
a browser should do if they're presented.
> > >
> > > Just off the top of my head:
> > >
> > > Can I...
> > > - Change the t= line to be something other than t=0 0?
>
> Current JSEP draft says the browser does not need to allow that to be
changed.
>
> > > - Change the rtpmap associations before calling setLocal?
>
> you can remove or reorder
>
> > > - Change a=sendrecv to a=recvonly before calling setLocal?
>
> Current JSEP draft says that is done with constraints
>
> > > - What do you do when you see a=content:sl ?
>
> not in current spec but you can find dissection on list about making sure
that is available to API
>
> > > - What if someone adds an r= or p= or e=?
>
> Current JSEP draft says the browser does not need to allow that
>
> > - What is the RFC that describes a=group:BUNDLE (as seen in some
browser implementations)?
>
> Uh, I'll leave that as an exercise to the readers. Try hard and see if
you can find it.
>
>
> > > - Can I remove a=group:BUNDLE (or add it) before calling setLocal?
>
> Current JSEP draft says that is done with constraints
>
> > > - How about removing a=rtcp-mux?
>
> Current JSEP draft says that is done with constraints
>
>
> > > - Should I do something special at my end if you set
a=ice-options:google-ice ?
>
> Do whatever the ice spec say
>
> > > - If you put a=ssrc lines in there, can I change the ssrc before
passing it back to setLocal?
> Current JSEP draft says the browser does not need to allow that to be
changed.
>
>
> > > - Can I delete the a=ssrc lines?
>
> Current JSEP draft says the browser does not need to allow that to be
changed.
>
>
> > > - Is a=rtcp:1 IN IP4 0.0.0.0 valid or not?
>
> Why would a browser want to generate that?
>
> > > - What about 0.0.0.0 in the o= line?
>
> Why would a browser want to generate that?
>
> > > - If I get a bunch of a=candidate lines, can I swap them around to
change the priority before calling setLocal?
>
> Current JSEP draft says the browser does not need to allow that to be
changed.
>
> > > - What if someone claims SAVP instead of SAVPF but gives me rtcp info?
>
> We need to add text to explicitly cover that. You can find text on what I
think we should do for this in
> draft-jennings-rtcweb-plan-01
>
>
> > >
> > > At the *very least* for each and every line that comes from
createOffer and createAnswer you must be able to answer the following:
> > > - Can I delete it?
> > > - Duplicate it?
> > > - Change it?
> > > - If not, how are violation handled? (Both when passed to another
browser at the far end and when these modifications happen before calling
setLocal)
> > > - And can I add additional valid SDP to what came from createOffer or
createAnswer and pass it back to setLocal or not?
> > >
> > > We appear to be nowhere near a document which explains the answers to
these questions, certainly not in the W3C WG, and not in any IETF RFC or
draft I can locate.
>
> draft-ietf-rtcweb-jsep-03 has tried to address some of these. I certainly
don't claim it is perfect but if people have use cases that suggest a
change to this, love to get text on what it should be.
>
> Mathew has been very clear along the way of "we need to define X to have
an interoperable system" and for most values of X he has proposed, I agree
with him. However, I think we just need to do that and that it actually
won't be much work once we get the framework in place to deal with thinks
like how many ports is the RTP using and how many m-lines does the SDP have.

FYI, Adam today said "we're nowhere close" and now you're saying it "won't
be much work".

>
> > >
> > > Matthew Kaufman
> > >
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>

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<p dir=3D"ltr"><br>
On Jul 19, 2013 6:47 PM, &quot;Cullen Jennings&quot; &lt;<a href=3D"mailto:=
fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; Well that was clearly one or my more mindless emails<br>
&gt;<br>
&gt; What I mean to say was something along lines of<br>
&gt;<br>
&gt; Folks want to sort out the multiplexing / bundle stuff first.<br>
&gt;<br>
&gt; So let me suggest that folks read section 6 of the JSEP draft and see =
if they can answer most of of the questions below. I&#39;m not saying that =
section 6 is right, in fact I think some of it is wrong. But at least act l=
ike you have read it.<br>

&gt;<br>
&gt;<br>
&gt; On Jul 19, 2013, at 7:49 AM, Peter Thatcher &lt;<a href=3D"mailto:ptha=
tcher@google.com">pthatcher@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; &quot;mindless works fist&quot;? =C2=A0There&#39;s a deep meaning=
 there, but I&#39;m still trying to figure it out...<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On Fri, Jul 19, 2013 at 7:04 AM, Cullen Jennings &lt;<a href=3D"m=
ailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt; &gt; Folks wanted to sort out how mindless works fist before getting t=
o theses smaller details but in general I think progress is being made on a=
ll of these<br>
&gt; &gt;<br>
&gt; &gt; On Jul 17, 2013, at 2:58 PM, &quot;Matthew Kaufman (SKYPE)&quot; =
&lt;<a href=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net<=
/a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; &gt; Bernard Aboba:<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; The problem was that we never defined what mangling brow=
sers had to<br>
&gt; &gt; &gt;&gt; support. Given all the SDP specs that is actually a huge=
 work item.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; This is exactly what my slides that weren&#39;t presented la=
st November started to touch upon. It only takes a few minutes with RFC3264=
 and its references to start documenting cases that are clearly &quot;valid=
 SDP offer/answer&quot; and yet for which I cannot for the life of me figur=
e out what a browser should do if they&#39;re presented.<br>

&gt; &gt; &gt;<br>
&gt; &gt; &gt; Just off the top of my head:<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Can I...<br>
&gt; &gt; &gt; - Change the t=3D line to be something other than t=3D0 0?<b=
r>
&gt;<br>
&gt; Current JSEP draft says the browser does not need to allow that to be =
changed.<br>
&gt;<br>
&gt; &gt; &gt; - Change the rtpmap associations before calling setLocal?<br=
>
&gt;<br>
&gt; you can remove or reorder<br>
&gt;<br>
&gt; &gt; &gt; - Change a=3Dsendrecv to a=3Drecvonly before calling setLoca=
l?<br>
&gt;<br>
&gt; Current JSEP draft says that is done with constraints<br>
&gt;<br>
&gt; &gt; &gt; - What do you do when you see a=3Dcontent:sl ?<br>
&gt;<br>
&gt; not in current spec but you can find dissection on list about making s=
ure that is available to API<br>
&gt;<br>
&gt; &gt; &gt; - What if someone adds an r=3D or p=3D or e=3D?<br>
&gt;<br>
&gt; Current JSEP draft says the browser does not need to allow that<br>
&gt;<br>
&gt; &gt; - What is the RFC that describes a=3Dgroup:BUNDLE (as seen in som=
e browser implementations)?<br>
&gt;<br>
&gt; Uh, I&#39;ll leave that as an exercise to the readers. Try hard and se=
e if you can find it.<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt; - Can I remove a=3Dgroup:BUNDLE (or add it) before calling s=
etLocal?<br>
&gt;<br>
&gt; Current JSEP draft says that is done with constraints<br>
&gt;<br>
&gt; &gt; &gt; - How about removing a=3Drtcp-mux?<br>
&gt;<br>
&gt; Current JSEP draft says that is done with constraints<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt; - Should I do something special at my end if you set a=3Dice=
-options:google-ice ?<br>
&gt;<br>
&gt; Do whatever the ice spec say<br>
&gt;<br>
&gt; &gt; &gt; - If you put a=3Dssrc lines in there, can I change the ssrc =
before passing it back to setLocal?<br>
&gt; Current JSEP draft says the browser does not need to allow that to be =
changed.<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt; - Can I delete the a=3Dssrc lines?<br>
&gt;<br>
&gt; Current JSEP draft says the browser does not need to allow that to be =
changed.<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt; - Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?<br>
&gt;<br>
&gt; Why would a browser want to generate that?<br>
&gt;<br>
&gt; &gt; &gt; - What about 0.0.0.0 in the o=3D line?<br>
&gt;<br>
&gt; Why would a browser want to generate that?<br>
&gt;<br>
&gt; &gt; &gt; - If I get a bunch of a=3Dcandidate lines, can I swap them a=
round to change the priority before calling setLocal?<br>
&gt;<br>
&gt; Current JSEP draft says the browser does not need to allow that to be =
changed.<br>
&gt;<br>
&gt; &gt; &gt; - What if someone claims SAVP instead of SAVPF but gives me =
rtcp info?<br>
&gt;<br>
&gt; We need to add text to explicitly cover that. You can find text on wha=
t I think we should do for this in<br>
&gt; draft-jennings-rtcweb-plan-01<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; At the *very least* for each and every line that comes from =
createOffer and createAnswer you must be able to answer the following:<br>
&gt; &gt; &gt; - Can I delete it?<br>
&gt; &gt; &gt; - Duplicate it?<br>
&gt; &gt; &gt; - Change it?<br>
&gt; &gt; &gt; - If not, how are violation handled? (Both when passed to an=
other browser at the far end and when these modifications happen before cal=
ling setLocal)<br>
&gt; &gt; &gt; - And can I add additional valid SDP to what came from creat=
eOffer or createAnswer and pass it back to setLocal or not?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; We appear to be nowhere near a document which explains the a=
nswers to these questions, certainly not in the W3C WG, and not in any IETF=
 RFC or draft I can locate.<br>
&gt;<br>
&gt; draft-ietf-rtcweb-jsep-03 has tried to address some of these. I certai=
nly don&#39;t claim it is perfect but if people have use cases that suggest=
 a change to this, love to get text on what it should be.<br>
&gt;<br>
&gt; Mathew has been very clear along the way of &quot;we need to define X =
to have an interoperable system&quot; and for most values of X he has propo=
sed, I agree with him. However, I think we just need to do that and that it=
 actually won&#39;t be much work once we get the framework in place to deal=
 with thinks like how many ports is the RTP using and how many m-lines does=
 the SDP have.</p>

<p dir=3D"ltr">FYI, Adam today said &quot;we&#39;re nowhere close&quot; and=
 now you&#39;re saying it &quot;won&#39;t be much work&quot;.=C2=A0 </p>
<p dir=3D"ltr">&gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Matthew Kaufman<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; rtcweb mailing list<br>
&gt; &gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">htt=
ps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt;<br>
&gt;<br>
</p>

--e89a8fb20834b6328c04e1e7c194--

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From: Peter Thatcher <pthatcher@google.com>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Like I said when I first sent the summary, anyone is welcome to make a
different summary from the raw data, or collect their own data.  I haven't
been around long enough to know how to categorize people and whose input is
considered valuable and whose isn't, so I ignorantly gave everyone equal
weight.  So if you'd like, you can filter out  Bernard and recalculate.

On Jul 19, 2013 6:52 PM, "Cullen Jennings" <fluffy@iii.ca> wrote:

>
> On Jul 19, 2013, at 6:24 PM, Peter Thatcher <pthatcher@google.com> wrote:
>
> > You're surprised that the first web developers that try out WebRTC and
> have feedback have experience with SIP and VoIP?  We shouldn't find that
> surprising, nor should we discount their feedback because of it.
>
> I was not saying that feedback from folks like Bernard Aboba was not
> useful, of course it can be. I was just noting the sample set.

--047d7b6705b95989f604e1e7f209
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<p dir=3D"ltr">Like I said when I first sent the summary, anyone is welcome=
 to make a different summary from the raw data, or collect their own data.=
=C2=A0 I haven&#39;t been around long enough to know how to categorize peop=
le and whose input is considered valuable and whose isn&#39;t, so I ignoran=
tly gave everyone equal weight.=C2=A0 So if you&#39;d like, you can filter =
out=C2=A0 Bernard and recalculate.<br>
<br></p>
<div class=3D"gmail_quote">On Jul 19, 2013 6:52 PM, &quot;Cullen Jennings&q=
uot; &lt;<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br t=
ype=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
On Jul 19, 2013, at 6:24 PM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt; You&#39;re surprised that the first web developers that try out WebRTC=
 and have feedback have experience with SIP and VoIP? =C2=A0We shouldn&#39;=
t find that surprising, nor should we discount their feedback because of it=
.<br>

<br>
I was not saying that feedback from folks like Bernard Aboba was not useful=
, of course it can be. I was just noting the sample set. </blockquote></div=
>

--047d7b6705b95989f604e1e7f209--

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Cullen Jennings <fluffy@iii.ca>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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From: Cullen Jennings [mailto:fluffy@iii.ca]:
> On Jul 19, 2013, at 9:54 AM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>=20
> > Negotiation is a hole.  A vast, soul-sucking, waste of time.
>=20
> That's might be closer to true when you control both ends, like Skype.
>=20
> It's not true when a browser from one vendor running an application from =
a
> scone vendor needs to talk to video conferencing system from a third
> vendor.

If you think that the second vendor's web server + signaling gateway should=
 be doing negotiation between itself and the third vendor's videoconferenci=
ng system, then that's fine.

If you think that the second vendor is going to get the first vendor to shi=
p an updated browser to make its SDP O/A 100% compliant with what the third=
 vendor's system is expecting, then you're crazy. It simply isn't going to =
happen, and what's more, given that the browser from the first vendor conta=
ins a complete JavaScript implementation, there's no reason why it should b=
e. The first vendor should provide the JS APIs that are necessary for the s=
econd vendor to build the system they want to build. Done. And without the =
endless pain that trying to make browser SDP O/A compatible with anything e=
lse is going to be.

Matthew Kaufman


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Date: Fri, 19 Jul 2013 23:17:47 -0400
From: Robin Raymond <robin@hookflash.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Is WebRTC exclusively targeting browser interoperability ? I'm not 
trying to nitpick, but this is a vital point of issue.

If yes, then why do we need to carry so much legacy SDP stuff to achieve 
basic sending of media between browsers? Seems like an incredible 
designoverkill to me. It's much easier to define a minimal feature set 
with a wire protocol that can achieve basic media between browsers. You 
could define "SDP-light" and that would be sufficient.

If no, then why isn't JavaScript the entity generating the SDPs rather 
than the browser binaries?

If the browsers aren't the end-all-be-all for interoperability, then all 
those SDPs have work not just between browsers, but untold numbers of 
SIP devices, servers, gateways, relays, desktop applications and mobile 
devices. Either all of those devices have to become equally capable of 
handling every WebRTC SDP produced from every browser, every version, 
working around every format difference, as every other device on the 
planet -- or -- we'll have to extract the SDP information using JS 
libraries and signal a format strictly compatible to our network 
environment. Personally, I think most will choose the latter in 
practice. Even SIP people using SDP will end up extracting the browser's 
SDP and recreate their own SDP instead.

Given how tough it is for two browser vendors (who even have access to 
each other's source code) to get this SDP negotiation right, I'm 
extremely concerned about an entire planet of devices getting this stuff 
right.

The trouble isn't that SDP isn't expressive enough or it's too ugly a 
format, it's that it's too expressive (okay, and it's ugly too). The 
trouble isn't that it's easy to negotiate O/A only for simple cases, 
it's that when we end up having to signal this stuff in "our own way", 
we'll be fighting with an O/A state negotiation machine instead of 
controlling the RTP engine directly with knobs and dials.

As a developer, I want:
1) to control what the RTP engine does on the wire (to ensure 
compatibility between devices);
2) strict, well defined, easy-to-understand API contracts that do not 
change, and are only extended in well defined ways;
3) knobs and dials to control the exposed capabilities, and _not_ a 
negotiation machine to not fight where I can only indirectly "do what I 
want";
4) predictable outputs, i.e. not having random extensions or odd 
differences in the middle some blob format show up from 
version-to-version, and from browser-to-browser;


I do not believe we can address these by tweaking the SDP format, or 
defining SDP better, or making wrappers for SDP.

The problem is fundamentally SDP O/A as the mandated.


-Robin


> Cullen Jennings <mailto:fluffy@iii.ca>
> 19 July, 2013 9:24 PM
>
>
> Well lets prioritize fixing them. Which ones are causing 
> interoperability problems between browsers?

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 boundary="------------060002030301030201000606"


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<html><head>
<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
</head><body bgcolor="#FFFFFF" text="#000000"><br>
Is WebRTC exclusively targeting browser <span>interoperability </span>? 
I'm not trying to nitpick, but this is a vital point of issue.<br>
<br>
If yes, then why do we need to carry so much legacy SDP stuff to achieve
 basic sending of media between browsers? Seems like an incredible 
design<span> overkill</span> to me. It's much easier to define a minimal
 feature set with a wire protocol that can achieve basic media between 
browsers. You could define "SDP-light" and that would be sufficient.<br>
<br>
If no, then why isn't JavaScript the entity generating the SDPs rather 
than the browser binaries?<br>
<br>
If the browsers aren't the end-all-be-all for interoperability, then all
 those SDPs have work not just between browsers, but untold numbers of 
SIP devices, servers, gateways, relays, desktop applications and mobile 
devices. Either all of those devices have to become equally capable of 
handling every WebRTC SDP produced from every browser, every version, 
working around every format difference, as every other device on the 
planet -- or -- we'll have to extract the SDP information using JS 
libraries and signal a format strictly compatible to our network 
environment. Personally, I think most will choose the latter in 
practice. Even SIP people using SDP will end up extracting the browser's
 SDP and recreate their own SDP instead.<br>
<br>
Given how tough it is for two browser vendors (who even have access to 
each other's source code) to get this SDP negotiation right, I'm 
extremely concerned about an entire planet of devices getting this stuff
 right.<br>
<br>
The trouble isn't that SDP isn't expressive enough or it's too ugly a 
format, it's that it's too expressive (okay, and it's ugly too). The 
trouble isn't that it's easy to negotiate O/A only for simple cases, 
it's that when we end up having to signal this stuff in "our own way", 
we'll be fighting with an O/A state negotiation machine instead of 
controlling the RTP engine directly with knobs and dials.<br>
<br>
As a developer, I want:<br>
1) to control what the RTP engine does on the wire (to ensure 
compatibility between devices);<br>
2) strict, well defined, easy-to-understand API contracts that do not 
change, and are only extended in well defined ways;<br>
3) knobs and dials to control the exposed capabilities, and _not_ a 
negotiation machine to not fight where I can only indirectly "do what I 
want";<br>
4) predictable outputs, i.e. not having random extensions or odd 
differences in the middle some blob format show up from 
version-to-version, and from browser-to-browser;<br>
<br>
<br>
I do not believe we can address these by tweaking the SDP format, or 
defining SDP better, or making wrappers for SDP.<br>
<br>
The problem is fundamentally SDP O/A as the mandated.<br>
<br>
<br>
-Robin<br>
<br>
<br>
<blockquote style="border: 0px none;" 
cite="mid:3C837643-5DB1-4192-8DA3-6D999E867E77@iii.ca" type="cite">
  <div style="margin:30px 25px 10px 25px;" class="__pbConvHr"><div 
style="display:table;width:100%;border-top:1px solid 
#EDEEF0;padding-top:5px"> 	<div 
style="display:table-cell;vertical-align:middle;padding-right:6px;"><img
 photoaddress="fluffy@iii.ca" photoname="Cullen Jennings" 
src="cid:part1.05080703.07090909@hookflash.com" 
name="postbox-contact.jpg" height="25px" width="25px"></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;width:100%">
   	<a moz-do-not-send="true" href="mailto:fluffy@iii.ca" 
style="color:#737F92 
!important;padding-right:6px;font-weight:bold;text-decoration:none 
!important;">Cullen Jennings</a></div>   <div 
style="display:table-cell;white-space:nowrap;vertical-align:middle;">   
  <font color="#9FA2A5"><span style="padding-left:6px">19 July, 2013 
9:24 PM</span></font></div></div></div>
  <div style="color:#888888;margin-left:24px;margin-right:24px;" 
__pbrmquotes="true" class="__pbConvBody"><div><!----><br><br>Well lets 
prioritize fixing them. Which ones are causing interoperability problems
 between browsers? <br></div></div>
</blockquote>
</body></html>

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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You can remove my line in the spreadsheet, but the residue of my bit presenc=
e may remain, polluting nearby cells. I recommend use of bit bleach.=20

On Jul 19, 2013, at 7:10 PM, "Peter Thatcher" <pthatcher@google.com> wrote:

> Like I said when I first sent the summary, anyone is welcome to make a dif=
ferent summary from the raw data, or collect their own data.  I haven't been=
 around long enough to know how to categorize people and whose input is cons=
idered valuable and whose isn't, so I ignorantly gave everyone equal weight.=
  So if you'd like, you can filter out  Bernard and recalculate.
>=20
>=20
> On Jul 19, 2013 6:52 PM, "Cullen Jennings" <fluffy@iii.ca> wrote:
>>=20
>> On Jul 19, 2013, at 6:24 PM, Peter Thatcher <pthatcher@google.com> wrote:=

>>=20
>> > You're surprised that the first web developers that try out WebRTC and h=
ave feedback have experience with SIP and VoIP?  We shouldn't find that surp=
rising, nor should we discount their feedback because of it.
>>=20
>> I was not saying that feedback from folks like Bernard Aboba was not usef=
ul, of course it can be. I was just noting the sample set.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>You can remove my line in the spreadsheet, but the residue of my bit presence may remain, polluting nearby cells. I recommend use of bit bleach.&nbsp;</div><div><br>On Jul 19, 2013, at 7:10 PM, "Peter Thatcher" &lt;<a href="mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div><p dir="ltr">Like I said when I first sent the summary, anyone is welcome to make a different summary from the raw data, or collect their own data.&nbsp; I haven't been around long enough to know how to categorize people and whose input is considered valuable and whose isn't, so I ignorantly gave everyone equal weight.&nbsp; So if you'd like, you can filter out&nbsp; Bernard and recalculate.<br>
<br></p>
<div class="gmail_quote">On Jul 19, 2013 6:52 PM, "Cullen Jennings" &lt;<a href="mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
On Jul 19, 2013, at 6:24 PM, Peter Thatcher &lt;<a href="mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt; You're surprised that the first web developers that try out WebRTC and have feedback have experience with SIP and VoIP? &nbsp;We shouldn't find that surprising, nor should we discount their feedback because of it.<br>

<br>
I was not saying that feedback from folks like Bernard Aboba was not useful, of course it can be. I was just noting the sample set. </blockquote></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>rtcweb mailing list</span><br><span><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>
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_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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From pthatcher@google.com  Fri Jul 19 20:28:32 2013
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Date: Fri, 19 Jul 2013 20:28:31 -0700
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From: Peter Thatcher <pthatcher@google.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Best to nuke the spreadsheet from orbit.  It's the only way to be sure.
On Jul 19, 2013 8:20 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:

> You can remove my line in the spreadsheet, but the residue of my bit
> presence may remain, polluting nearby cells. I recommend use of bit bleach.
>
> On Jul 19, 2013, at 7:10 PM, "Peter Thatcher" <pthatcher@google.com>
> wrote:
>
> Like I said when I first sent the summary, anyone is welcome to make a
> different summary from the raw data, or collect their own data.  I haven't
> been around long enough to know how to categorize people and whose input is
> considered valuable and whose isn't, so I ignorantly gave everyone equal
> weight.  So if you'd like, you can filter out  Bernard and recalculate.
>
> On Jul 19, 2013 6:52 PM, "Cullen Jennings" <fluffy@iii.ca> wrote:
>
>>
>> On Jul 19, 2013, at 6:24 PM, Peter Thatcher <pthatcher@google.com> wrote:
>>
>> > You're surprised that the first web developers that try out WebRTC and
>> have feedback have experience with SIP and VoIP?  We shouldn't find that
>> surprising, nor should we discount their feedback because of it.
>>
>> I was not saying that feedback from folks like Bernard Aboba was not
>> useful, of course it can be. I was just noting the sample set.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<p dir=3D"ltr">Best to nuke the spreadsheet from orbit.=C2=A0 It&#39;s the =
only way to be sure.</p>
<div class=3D"gmail_quote">On Jul 19, 2013 8:20 PM, &quot;Bernard Aboba&quo=
t; &lt;<a href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.c=
om</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"auto"><div>You can remove my line in the spreadsheet, but the r=
esidue of my bit presence may remain, polluting nearby cells. I recommend u=
se of bit bleach.=C2=A0</div><div><br>On Jul 19, 2013, at 7:10 PM, &quot;Pe=
ter Thatcher&quot; &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_b=
lank">pthatcher@google.com</a>&gt; wrote:<br>
<br></div><blockquote type=3D"cite"><div><p dir=3D"ltr">Like I said when I =
first sent the summary, anyone is welcome to make a different summary from =
the raw data, or collect their own data.=C2=A0 I haven&#39;t been around lo=
ng enough to know how to categorize people and whose input is considered va=
luable and whose isn&#39;t, so I ignorantly gave everyone equal weight.=C2=
=A0 So if you&#39;d like, you can filter out=C2=A0 Bernard and recalculate.=
<br>

<br></p>
<div class=3D"gmail_quote">On Jul 19, 2013 6:52 PM, &quot;Cullen Jennings&q=
uot; &lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</=
a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<br>
On Jul 19, 2013, at 6:24 PM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com" target=3D"_blank">pthatcher@google.com</a>&gt; wrote:<br>
<br>
&gt; You&#39;re surprised that the first web developers that try out WebRTC=
 and have feedback have experience with SIP and VoIP? =C2=A0We shouldn&#39;=
t find that surprising, nor should we discount their feedback because of it=
.<br>


<br>
I was not saying that feedback from folks like Bernard Aboba was not useful=
, of course it can be. I was just noting the sample set. </blockquote></div=
>
</div></blockquote><blockquote type=3D"cite"><div><span>___________________=
____________________________</span><br><span>rtcweb mailing list</span><br>=
<span><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org<=
/a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br></div></bl=
ockquote></div></blockquote></div>

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Cullen Jennings <fluffy@iii.ca>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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From: Cullen Jennings [mailto:fluffy@iii.ca]:
>=20
> Well that was clearly one or my more mindless emails
>=20
> What I mean to say was something along lines of
>=20
> Folks want to sort out the multiplexing / bundle stuff first.
>=20
> So let me suggest that folks read section 6 of the JSEP draft and see if =
they
> can answer most of of the questions below. I'm not saying that section 6 =
is
> right, in fact I think some of it is wrong. But at least act like you hav=
e read it.
>=20

Well, first off the "JSEP draft" is an Internet Draft, and I thought we wer=
e writing a W3C API here. But I'll play along this time. I'm assuming of co=
urse that we're all talking about the latest JSEP draft, which appears to b=
e draft-ietf-rtcweb-jsep-03, wherein "section 6" is only a page and a third=
 long.

> > > Can I...
> > > - Change the t=3D line to be something other than t=3D0 0?
>=20
> Current JSEP draft says the browser does not need to allow that to be
> changed.

Where? Don't see that at all.

>=20
> > > - Change the rtpmap associations before calling setLocal?
>=20
> you can remove or reorder

But can I change which numbers are assigned? Like if I want 115 for opus in=
stead of 112?

>=20
> > > - Change a=3Dsendrecv to a=3Drecvonly before calling setLocal?
>=20
> Current JSEP draft says that is done with constraints

And it says "these changes may also be be performed by manipulating the SDP=
 returned from createOffer/createAnswer" (and yes, it really does say "be b=
e")

So I guess the answer to this one is "yes"

>=20
> > > - What do you do when you see a=3Dcontent:sl ?
>=20
> not in current spec but you can find dissection on list about making sure=
 that
> is available to API


Agreed, not specified.

>=20
> > > - What if someone adds an r=3D or p=3D or e=3D?
>=20
> Current JSEP draft says the browser does not need to allow that


No, it says that since should accept valid SDP, it should accept it. At lea=
st that's how I read the spec.

>=20
> > - What is the RFC that describes a=3Dgroup:BUNDLE (as seen in some brow=
ser
> implementations)?
>=20
> Uh, I'll leave that as an exercise to the readers. Try hard and see if yo=
u can
> find it.

Appears to be work in progress in the MMUSIC WG. So no normative spec that =
can be referenced. Guess it must not be allowed, huh?

> > > - Can I remove a=3Dgroup:BUNDLE (or add it) before calling setLocal?
>=20
> Current JSEP draft says that is done with constraints

(Or by modifying the SDP, see above... and there's no API for said constrai=
nt)

>=20
> > > - How about removing a=3Drtcp-mux?
>=20
> Current JSEP draft says that is done with constraints

(Or by modifying the SDP, see above... and no API for this constraint eithe=
r)

>=20
>=20
> > > - Should I do something special at my end if you set a=3Dice-options:=
google-
> ice ?
>=20
> Do whatever the ice spec say

Section 15.5 of RFC5245 says absolutely nothing about what I'm supposed to =
do with this. And that's the only place a=3Dice-options is discussed.

>=20
> > > - If you put a=3Dssrc lines in there, can I change the ssrc before pa=
ssing it
> back to setLocal?
> Current JSEP draft says the browser does not need to allow that to be
> changed.

Not true. The current spec says that the non-changeable lines are the "numb=
er, type and port number of m-lines" "the generated ICE credentials" and "t=
he set of ICE candidates and their parameters".=20

Section 4.1 R-8 suggests that RFC5576 MUST be implemented to signal RTP SSR=
C values, but nowhere does it tell me (and especially not in section 6) if =
the browser will or should let me change them between createOffer and setLo=
cal.


>=20
>=20
> > > - Can I delete the a=3Dssrc lines?
>=20
> Current JSEP draft says the browser does not need to allow that to be
> changed.

Disagree. See above.

>=20
>=20
> > > - Is a=3Drtcp:1 IN IP4 0.0.0.0 valid or not?
>=20
> Why would a browser want to generate that?

I don't know. I have output from browsers that have generated that in the p=
ast. Was that a bug?

>=20
> > > - What about 0.0.0.0 in the o=3D line?
>=20
> Why would a browser want to generate that?

(Same as previous comment)

>=20
> > > - If I get a bunch of a=3Dcandidate lines, can I swap them around to =
change
> the priority before calling setLocal?
>=20
> Current JSEP draft says the browser does not need to allow that to be
> changed.

This appears to be correct. Which is of course unfortunate, as I might very=
 well wish to do this. Or delete candidates, which also appears to be forbi=
dden in section 6, despite several *very* valid use cases for deleting cand=
idates a browser happens to generate.

>=20
> > > - What if someone claims SAVP instead of SAVPF but gives me rtcp info=
?
>=20
> We need to add text to explicitly cover that. You can find text on what I=
 think
> we should do for this in
> draft-jennings-rtcweb-plan-01

Then we should add that text. To a W3C specification that covers what the b=
rowser must do.

>=20
>=20
> > >
> > > At the *very least* for each and every line that comes from createOff=
er
> and createAnswer you must be able to answer the following:
> > > - Can I delete it?
> > > - Duplicate it?
> > > - Change it?
> > > - If not, how are violation handled? (Both when passed to another
> browser at the far end and when these modifications happen before calling
> setLocal)
> > > - And can I add additional valid SDP to what came from createOffer or
> createAnswer and pass it back to setLocal or not?
> > >
> > > We appear to be nowhere near a document which explains the answers
> to these questions, certainly not in the W3C WG, and not in any IETF RFC =
or
> draft I can locate.
>=20
> draft-ietf-rtcweb-jsep-03 has tried to address some of these. I certainly=
 don't
> claim it is perfect but if people have use cases that suggest a change to=
 this,
> love to get text on what it should be.

I don't think it is close at all. See above comments.

>=20
> Mathew has been very clear along the way of "we need to define X to have
> an interoperable system" and for most values of X he has proposed, I agre=
e
> with him. However, I think we just need to do that and that it actually w=
on't
> be much work once we get the framework in place to deal with thinks like
> how many ports is the RTP using and how many m-lines does the SDP have.

Well, at least we do agree that the specification for what the browser must=
 do should be written down somewhere.

What we don't agree on is how trivial that problem is.

Matthew Kaufman



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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Peter Thatcher <pthatcher@google.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Date: Sat, 20 Jul 2013 03:41:16 +0000
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Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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From matthew.kaufman@skype.net  Fri Jul 19 20:45:09 2013
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Cullen Jennings <fluffy@iii.ca>
Thread-Topic: On babies and bathwater (was Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Date: Sat, 20 Jul 2013 03:42:51 +0000
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] On babies and bathwater (was Re: Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface)
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Cullen Jennings [mailto:fluffy@iii.ca]:
>=20
> On Jul 19, 2013, at 11:40 AM, Matthew Kaufman (SKYPE)
> <matthew.kaufman@skype.net> wrote:
>=20
> > From: Martin Thomson [mailto:martin.thomson@gmail.com]
> >>
> >> Negotiation is a hole.  A vast, soul-sucking, waste of time.
> >
> > And negotiation over SDP requires a trip to MMUSIC every time you think
> of something new, too. That's a soul-sucking waste of time *itself*.
> >
> > Matthew Kaufman
>=20
> Better than a trip to W3C, IETF, and XSF
>=20

One successfully early trip to the W3C to get some JS APIs that let me buil=
d anything I want later is a hell of a lot better than one trip to MMUSIC (=
when I'm already on a deadline, no doubt) every time I come up with an idea=
 in the future.

I know you know this, so I don't know why you would say what you said.

Matthew Kaufman


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 20 Jul 2013 16:03:56 +0200
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2013/7/20 Cullen Jennings <fluffy@iii.ca>:
> It's not true when a browser from one vendor running an application from =
a scone vendor needs to talk to video conferencing system from a third vend=
or.

In the WWW, the servide provider provides both the server side (in
this case the Web and the video conference server) and the client side
(the JS). Your concern is good for a SIP phone but not for a browser
nor for a JS custom application controlled and designed by the
provider.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2013/7/20 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> 2013/7/20 Cullen Jennings <fluffy@iii.ca>:
>> It's not true when a browser from one vendor running an application from=
 a scone vendor needs to talk to video conferencing system from a third ven=
dor.
>
> In the WWW, the servide provider provides both the server side (in
> this case the Web and the video conference server) and the client side
> (the JS). Your concern is good for a SIP phone but not for a browser
> nor for a JS custom application controlled and designed by the
> provider.

Cullen, could you please give your opinion about what I've said above?
I also would like to know what happens if the video conference server
uses Jingle or any future RTC protocol (which uses RTP but no SDP for
media sig).

Thanks a lot.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2013/7/20 Cullen Jennings <fluffy@iii.ca>:
> I will note that over half of that very small sample set is folks that ha=
ve been long termed involved in SIP and VoIP so, uh, might not be a represe=
ntative sample of type of people you are portraying it as.

It is even worse if people interested and expert in SIP tells you that
the current SDP-blob-based API is really wrong for using SIP in
WebRTC, don't you think?


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 20 Jul 2013 16:27:42 +0200
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To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/20 Cullen Jennings <fluffy@iii.ca>:
> Well lets prioritize fixing them. Which ones are causing interoperability=
 problems between browsers?

This means wasting time fixing SDP related issues that, once are
fixed, provides *nothing* useful to WebRTC.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [rtcweb] [avtext] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Just to mention that the family of 3D codecs based on H.264, incl. MVC, MVC=
+D, 3D-AVC, and even MFC, developed or being developed, should also be take=
n into consideration.

BR, YK

On Jul 19, 2013, at 15:54, "Adam Fineberg" <fineberg@vline.me<mailto:finebe=
rg@vline.me>> wrote:

Stephan,

On 7/19/13 3:44 PM, Stephan Wenger wrote:
Hi,

From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Friday, 19 July, 2013 15:12
To: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Cc: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, Bernard =
Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>>, "avtex=
t@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtext@ietf.org=
>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not at=
 all familiar with H.265.  I'll review the reference and see what I can fig=
ure.

StW: Good luck :-)

It seems though to me that you are suggesting that except in the simple cas=
e, that the data for H.265 would not be well suited to a header extension, =
am I understanding you correctly?  There is no reason the middlebox couldn'=
t get out of band signaling of the VPS as you mention but that would not be=
 within the scope of this header extension.

StW: well, if you would copy the layer_id into your header extension (just =
as you need to do for the simple case), a really smart middle box could use=
 this information just as a decoder uses it, assuming that it intercepted t=
he VPS in the first place.  Insofar, I wouldn't rule out the second option =
on technical grounds.  Whether any of the actual products would bother to d=
o that, ever, is another question.  I think the case ought to be documented=
, though.  I can help drafting text.

Any help is appreciated.

While we are at it: doing this right could mean that you need multiple spec=
s.  First, a generic header extension mechanism dedicated to side informati=
on required for pruning of RTP packet streams=97ideally not only for scalab=
le video, although that is the main customer today.  And second, for each "=
payload" (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v=
2 (including scalable and 3D extensions, which are not yet finalized), VP8,=
 and VP9 (I know little about the latter), plus Daala, and whatnot) a mappi=
ng of the bits available in the generic header extension to the bits in the=
 payload itself (NAL header and VPS in case of H.265, NALU header in SVC, a=
nd the fields you mention for VP8).

I was just thinking about that since it seems like the extension payload ma=
ybe completely specific and therefore not suitable for the same spec.  The =
problem in my mind with taking the hierarchical a[[roach you mention is tha=
t I'm not sure what if any common data is suitable accross all codecs and s=
ince there is already a spec for the general header extension, the generic =
one you mention might be fairly redundant.

I'm certainly open to your thoughts though.

Regards,
Adam


Stephan


Any insights are appreciated.

Regards,
Adam

On 7/19/13 8:33 AM, Stephan Wenger wrote:
Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling=97as the encrypted in-band VPS would not be useful under the assu=
mption that the middle box does not have the key to the media=97which is th=
e motivation of the draft in the first place).  I personally don't mind at =
all the second mechanism, as I'm a big fan of out-of-band parameter set tra=
nsmission and any middle box must be in the signaling path anyway to meanin=
gfully manipulate RTP.  I do not like the first option due to its variable,=
 and possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.


On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:

Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).

________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam

On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:        New Version Notification for draft-fineberg-avtext-temporal=
-layer-ext-00.txt
Date:   Tue, 09 Jul 2013 10:02:05 -0700
From:   internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>
To:     Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.=
com>



A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:        draft-fineberg-avtext-temporal-layer-ext
Revision:        00
Title:           A Real-Time Transport Protocol (RTP) Header Extension for =
VP8 Temporal Layer Information
Creation date:   2013-07-08
Group:           Individual Submission
Number of pages: 6
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.


_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--
Regards,
Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb





_______________________________________________
avtext mailing list
avtext@ietf.org<mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinf=
o/avtext


--
Regards,
Adam


--
Regards,
Adam

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

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</head>
<body dir=3D"auto">
<div>Just to mention that the family of 3D codecs based on H.264, incl. MVC=
, MVC&#43;D, 3D-AVC, and even MFC, developed or being developed, should als=
o be taken into consideration.<br>
<br>
BR, YK</div>
<div><br>
On Jul 19, 2013, at 15:54, &quot;Adam Fineberg&quot; &lt;<a href=3D"mailto:=
fineberg@vline.me">fineberg@vline.me</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>Stephan,<br>
<br>
<div class=3D"moz-cite-prefix">On 7/19/13 3:44 PM, Stephan Wenger wrote:<br=
>
</div>
<blockquote cite=3D"mid:CE0F0BEB.9F95D%25stewe@stewe.org" type=3D"cite">
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">Hi,</font></div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
<div style=3D"font-family:Calibri; font-size:11pt;
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          0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;
          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Adam Fineberg &lt;<a moz-do-n=
ot-send=3D"true" href=3D"mailto:fineberg@vline.me">fineberg@vline.me</a>&gt=
;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 19 July, 2013 15:12 <=
br>
<span style=3D"font-weight:bold">To: </span>Stephan Wenger &lt;<a moz-do-no=
t-send=3D"true" href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Justin Uberti &lt;<a moz-do-not=
-send=3D"true" href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt=
;, Bernard Aboba &lt;<a moz-do-not-send=3D"true" href=3D"mailto:bernard_abo=
ba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;, &quot;<a moz-do-not-send=
=3D"true" href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quot;
 &lt;<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@iet=
f.org</a>&gt;, &quot;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf=
.org">rtcweb@ietf.org</a>&quot; &lt;<a moz-do-not-send=3D"true" href=3D"mai=
lto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [avtext] [rtcweb] Fwd:=
 New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.t=
xt<br>
</div>
<div><br>
</div>
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">Stephan,<br>
<br>
Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm n=
ot at all familiar with H.265.&nbsp; I'll review the reference and see what=
 I can figure.&nbsp;</div>
</div>
</span>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
<br>
</div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">StW: Good luck :-)&nbsp;
<br>
</font></div>
</blockquote>
<blockquote cite=3D"mid:CE0F0BEB.9F95D%25stewe@stewe.org" type=3D"cite">
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">It seems though to me that you ar=
e suggesting that except in the simple case, that the data for H.265 would =
not be well suited to a header extension, am I understanding you correctly?=
&nbsp; There is no reason the middlebox couldn't
 get out of band signaling of the VPS as you mention but that would not be =
within the scope of this header extension.</div>
</div>
</span>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
<br>
</div>
<div><font color=3D"#0000ff"><font face=3D"Calibri,sans-serif">StW: well, i=
f you would copy the layer_id into your header extension (just as you need =
to do for the simple case), a really smart middle box could use this inform=
ation just as a decoder uses it,&nbsp;assuming&nbsp;that
 it intercepted the VPS in the first place. &nbsp;Insofar, I wouldn't rule =
out the second option on technical grounds. &nbsp;Whether any of the actual=
 products would bother to do that, ever, is another question. &nbsp;I think=
 the case ought to be documented, though. &nbsp;I can
 help drafting text.</font></font></div>
</blockquote>
<br>
Any help is appreciated.<br>
<br>
<blockquote cite=3D"mid:CE0F0BEB.9F95D%25stewe@stewe.org" type=3D"cite">
<div><font color=3D"#0000ff"><font face=3D"Calibri,sans-serif">While we are=
 at it: doing this right could mean that you need multiple specs. &nbsp;Fir=
st, a generic header extension mechanism dedicated to side information requ=
ired for pruning of RTP packet streams=97ideally
 not only for scalable video, although that is the main customer today. &nb=
sp;And second, for each &quot;payload&quot; (at present we are talking abou=
t H.264/SVC, H.265v1 (HEVC), H.265v2 (including scalable and 3D extensions,=
 which are not yet finalized), VP8, and VP9 (I
 know little about the latter), plus Daala, and whatnot) a mapping of the b=
its available in the generic header extension to the bits in the payload it=
self (NAL header and VPS in case of H.265, NALU header in SVC, and the fiel=
ds you mention for VP8).</font></font></div>
</blockquote>
<br>
I was just thinking about that since it seems like the extension payload ma=
ybe completely specific and therefore not suitable for the same spec.&nbsp;=
 The problem in my mind with taking the hierarchical a[[roach you mention i=
s that I'm not sure what if any common
 data is suitable accross all codecs and since there is already a spec for =
the general header extension, the generic one you mention might be fairly r=
edundant.<br>
<br>
I'm certainly open to your thoughts though.<br>
<br>
Regards,<br>
Adam<br>
<br>
<blockquote cite=3D"mid:CE0F0BEB.9F95D%25stewe@stewe.org" type=3D"cite">
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><br>
</div>
<div style=3D"font-family: Calibri, sans-serif; font-size: 14px; "><font co=
lor=3D"#0000ff">Stephan</font></div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000"><br>
<br>
Any insights are appreciated.<br>
<br>
Regards,<br>
Adam<br>
<br>
<div class=3D"moz-cite-prefix">On 7/19/13 8:33 AM, Stephan Wenger wrote:<br=
>
</div>
<blockquote cite=3D"mid:CE0E9F56.9F89B%25stewe@stewe.org" type=3D"cite">
<div>Hi,</div>
<div>I also believe that 16 bits should be enough. &nbsp;For H.264 and VP8 =
that has already been demonstrated. &nbsp;For H.265, some initial thoughts =
below. &nbsp;Apologies for the word-count.</div>
<div><br>
</div>
<div>The scalable version of H265 (called SHVC) is currently under developm=
ent. &nbsp;The current working draft can be found here:&nbsp;<a moz-do-not-=
send=3D"true" href=3D"http://phenix.int-evry.fr/jct/doc_end_user/documents/=
13_Incheon/wg11/JCTVC-M1008-v3.zip">http://phenix.int-evry.fr/jct/doc_end_u=
ser/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.</div>
<div><br>
</div>
<div>That said, the ordering by the encoder is just a recommendation, and t=
here may well be cases where different pruning strategies may be advisable.=
 &nbsp;For example, a layering structure could be constructed that expands =
into two branches, one using 2D scalable
 tools only, the other including view_id for multi view coding. &nbsp;By lo=
oking at the six bit field alone, a middle box will not be able to meaningf=
ully remove NAL units belonging to one of the branches completely while pru=
ning the other branch. &nbsp;In order to meaningfully
 deal with that scenario, there would be two options: one to represent the =
dimension_id[][] (and associated control info) in the header extension, or =
require the middle box to have access to the VPS and be able to interpret i=
ts content. &nbsp;The further could take
 considerably more than 16 bits and we would be talking about a variable le=
ngth data structure. &nbsp;The latter requires the middle box to have state=
 and a mechanism to intercept the VPS (through signaling=97as the encrypted=
 in-band VPS would not be useful under
 the assumption that the middle box does not have the key to the media=97wh=
ich is the motivation of the draft in the first place). &nbsp;I personally =
don't mind at all the second mechanism, as I'm a big fan of out-of-band par=
ameter set transmission and any middle
 box must be in the signaling path anyway to meaningfully manipulate RTP. &=
nbsp;I do not like the first option due to its variable, and possibly subst=
antial, overhead.</div>
<div><br>
</div>
<div>Stephan &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt;
                  text-align:left; color:black; BORDER-BOTTOM: medium
                  none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in;
                  PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP:
                  #b5c4df 1pt solid; BORDER-RIGHT: medium none;
                  PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Justin Uberti &lt;<a moz-do-n=
ot-send=3D"true" href=3D"mailto:juberti@google.com">juberti@google.com</a>&=
gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 19 July, 2013 06:32 <=
br>
<span style=3D"font-weight:bold">To: </span>Bernard Aboba &lt;<a moz-do-not=
-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotm=
ail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a moz-do-not-send=3D"tru=
e" href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quot; &lt;<a moz-do-=
not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, =
&quot;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@ie=
tf.org</a>&quot;
 &lt;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Fwd: New Vers=
ion Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">Agree those are the right codecs to design for. Since in e=
ach case there are fairly low limits on the number of supported layers (i.e=
. 3 spatial layers for SVC), I think it should be possible to pack the temp=
oral, spatial, quality layer ids into
 16 bits.
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <=
span dir=3D"ltr">
&lt;<a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x
                            #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>If we can support VP8/9 as well as H.264/5 SVC</div>
<div>that would be a start. It seems doable to me.</div>
<div>
<div>
<div><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a moz-do-not-se=
nd=3D"true" href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vl=
ine.me</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div>Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">I think it may be possible to generalize this.&nbsp; For e=
xample, for H.264/SVC which can support temporal, spatial and quality scala=
bility, you would need the quality_id and dependency_id in addition to the =
temporal_id (what you call the temporal
 layer index). &nbsp;&nbsp; <br>
<br>
<div>
<hr>
Date: Thu, 18 Jul 2013 08:45:38 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">
bernard_aboba@hotmail.com</a><br>
CC: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a>;
<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org" target=3D"_blan=
k">avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam<br>
<br>
<div>On 7/17/13 8:26 PM, Bernard Aboba wrote:<br>
</div>
<blockquote>
<div dir=3D"ltr">Since the need is not codec specific (e.g. it arises with =
any codec supporting temporal, spatial and quality scalability), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;<br>
<div>
<hr>
Date: Wed, 17 Jul 2013 17:09:46 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br s=
tyle=3D"text-indent:0px;letter-spacing:normal;text-transform:none;white-spa=
ce:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">I'm working on =
WebRTC services and have found that while developing services that forward =
VP8 video streams if we want to take
 advantage of the VP8 temporal scaling we must get the temporal layer infor=
mation from the RTP header which requires us to decrypt the SRTP packets. T=
his is undesirable both because the middle-box needs to have access to the =
keys as well as the because of the
 added overhead of the decrypt/encrypt cycle. This draft proposes an RTP he=
ader extension that will allow us to use the VP8 temporal layer information=
 included in the header extension and therefore do forwarding without SRTP =
decryption. Comments welcome.</span><br style=3D"text-indent:0px;letter-spa=
cing:normal;text-transform:none;white-space:normal;word-spacing:0px">
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Regards,</span>=
<br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whit=
e-space:normal;word-spacing:0px">
<span style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;wh=
ite-space:normal;display:inline!important;word-spacing:0px">Adam Fineberg</=
span><br style=3D"text-indent:0px;letter-spacing:normal;text-transform:none=
;white-space:normal;word-spacing:0px">
<div style=3D"text-indent:0px;letter-spacing:normal;text-transform:none;whi=
te-space:normal;word-spacing:0px">
<a moz-do-not-send=3D"true" href=3D"mailto:fineberg%20at%20vline.com" rel=
=3D"nofollow" target=3D"_blank">fineberg at vline.com</a><br>
<br>
-------- Original Message --------
<table border=3D"0" cellpadding=3D"0" cellspacing=3D"0">
<tbody>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Subject:</th>
<td>New Version Notification for draft-fineberg-avtext-temporal-layer-ext-0=
0.txt</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Date:</th>
<td>Tue, 09 Jul 2013 10:02:05 -0700</td>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">From:</th>
<td><a moz-do-not-send=3D"true" href=3D"mailto:internet-drafts%20at%20ietf.=
org" rel=3D"nofollow" target=3D"_blank">internet-drafts at ietf.org</a></td=
>
</tr>
<tr>
<th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">To:</th>
<td>Adam Fineberg<span>&nbsp;</span><a moz-do-not-send=3D"true" href=3D"mai=
lto:fineberg%20at%20vline.com" rel=3D"nofollow" target=3D"_blank">&lt;fineb=
erg at vline.com&gt;</a></td>
</tr>
</tbody>
</table>
<br>
<br>
<pre style=3D"width:939.59px;white-space:pre-wrap">A new version of I-D, dr=
aft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send=3D"true" href=3D"http://www.ietf.org/in=
ternet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel=3D"nofol=
low" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-fineberg-a=
vtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send=3D"true" href=3D"http://datatracker.iet=
f.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel=3D"nofollow" target=
=3D"_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-=
layer-ext</a>
Htmlized:        <a moz-do-not-send=3D"true" href=3D"http://tools.ietf.org/=
html/draft-fineberg-avtext-temporal-layer-ext-00" rel=3D"nofollow" target=
=3D"_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer=
-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
</div>
<br>
_______________________________________________ rtcweb mailing list <a moz-=
do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a moz-do-not-send=3D"true" href=3D"https://www.ietf.or=
g/mailman/listinfo/rtcweb" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
</div>
</blockquote>
<br>
<pre>--=20
Regards,
Adam</pre>
</div>
</div>
</blockquote>
<br>
</div>
</blockquote>
</div>
</div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blan=
k">rtcweb@ietf.org</a><br>
<a moz-do-not-send=3D"true" href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><b=
r>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</span><br>
<fieldset class=3D"mimeAttachmentHeader"></fieldset> <br>
<pre wrap=3D"">_______________________________________________
avtext mailing list
<a moz-do-not-send=3D"true" class=3D"moz-txt-link-abbreviated" href=3D"mail=
to:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send=3D"true" class=3D=
"moz-txt-link-freetext" href=3D"https://www.ietf.org/mailman/listinfo/avtex=
t">https://www.ietf.org/mailman/listinfo/avtext</a></pre>
</blockquote>
<br>
<pre class=3D"moz-signature" cols=3D"72">--=20
Regards,
Adam</pre>
</div>
</div>
</span></blockquote>
<br>
<pre class=3D"moz-signature" cols=3D"72">--=20
Regards,
Adam</pre>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

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Date: Sun, 21 Jul 2013 18:38:58 -0700
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--001a1132eee6ea076904e20fbf2d
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On Sun, Jul 21, 2013 at 6:34 PM, Lijing (Jessie, Technology Introduction &
Standard and Patent Dept) <lijing80@huawei.com> wrote:

> I fully agree with the idea to " use existing "native" codecs ".
>
> In addition, from a web developer's perspective, in order to make use of
> the "native" codecs, the browser should provide codec APIs to open the
> local codec capabilities of the device to applications and developers. It
> really make sense.
>

I don't understand what this means.

-Ekr

--001a1132eee6ea076904e20fbf2d
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">On Sun, Jul 21, 2013 at 6:34 PM, Lijing (Jessie, Technolog=
y Introduction &amp; Standard and Patent Dept) <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:lijing80@huawei.com" target=3D"_blank">lijing80@huawei.com</a>&=
gt;</span> wrote:<br>

<div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">I fully agree with the idea to &quot; use existing &quot;native&q=
uot; codecs &quot;.<br>


<br>
In addition, from a web developer&#39;s perspective, in order to make use o=
f the &quot;native&quot; codecs, the browser should provide codec APIs to o=
pen the local codec capabilities of the device to applications and develope=
rs. It really make sense.<br>

</blockquote><div><br></div><div>I don&#39;t understand what this means.</d=
iv><div><br></div><div>-Ekr</div><div>=A0</div></div></div></div>

--001a1132eee6ea076904e20fbf2d--

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From: "Lijing (Jessie, Technology Introduction & Standard and Patent Dept)" <lijing80@huawei.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: =?gb2312?B?W3J0Y3dlYl0gtPC4tDogU29tZSB0aG91Z2h0cyBvbiBvcHRpb25hbCBhdWRp?= =?gb2312?Q?o_codecs?=
Thread-Index: Ac6BN11IugiIgzm2TXug3PomYB8N3AFQBe5g//+DcQD//3HGgA==
Date: Mon, 22 Jul 2013 03:42:17 +0000
Message-ID: <A3045C90BB645147BC99159AA47ABAC7419F6161@szxeml558-mbs.china.huawei.com>
References: <BBE9739C2C302046BD34B42713A1E2A22DEE3029@ESESSMB105.ericsson.se> <A3045C90BB645147BC99159AA47ABAC7419F610E@szxeml558-mbs.china.huawei.com> <CABcZeBNrVByn0fnYB_CEgZ9eFMX+nF8Rmmn3yLm-VZET9sBhcA@mail.gmail.com>
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<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Sorry</spa=
n><span style=3D"font-size:10.5pt;color:#1F497D">=A3=AC</span><span lang=3D=
"EN-US" style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;san=
s-serif&quot;;color:#1F497D">I=A1=AFve
 confused two concepts. What I want to express is that,<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Firstly, t=
he browser may be able to make use of =A1=B0native=A1=B1 codec APIs of the =
=A1=B0hard=A1=B1 device(e.g. mobile phone), and include the =A1=B0native=A1=
=B1 codecs in
 the SDP O/A messages.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In additio=
n, for application flexibility, the browser should provide codec capability=
 APIs to developers. Then, &nbsp;the developers can choose proper
 codecs according to the specific scenario. <o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">So there s=
hould be two APIs. The first is provide by the device and used by the brows=
er. The second is provided by the browser and used by the
 developer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Best Regar=
ds,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Lijing<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt">=B7=A2=BC=FE=C8=
=CB<span lang=3D"EN-US">:</span></span></b><span lang=3D"EN-US" style=3D"fo=
nt-size:10.0pt"> Eric Rescorla [mailto:ekr@rtfm.com]
<br>
</span><b><span style=3D"font-size:10.0pt">=B7=A2=CB=CD=CA=B1=BC=E4<span la=
ng=3D"EN-US">:</span></span></b><span lang=3D"EN-US" style=3D"font-size:10.=
0pt"> 2013</span><span style=3D"font-size:10.0pt">=C4=EA<span lang=3D"EN-US=
">7</span>=D4=C2<span lang=3D"EN-US">22</span>=C8=D5<span lang=3D"EN-US"> 9=
:39<br>
</span><b>=CA=D5=BC=FE=C8=CB<span lang=3D"EN-US">:</span></b><span lang=3D"=
EN-US"> Lijing (Jessie, Technology Introduction &amp; Standard and Patent D=
ept)<br>
</span><b>=B3=AD=CB=CD<span lang=3D"EN-US">:</span></b><span lang=3D"EN-US"=
> Bo Burman; rtcweb@ietf.org<br>
</span><b>=D6=F7=CC=E2<span lang=3D"EN-US">:</span></b><span lang=3D"EN-US"=
> Re: [rtcweb] </span>
=B4=F0=B8=B4<span lang=3D"EN-US">: Some thoughts on optional audio codecs<o=
:p></o:p></span></span></p>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">On Sun, Jul 21, 2013 at 6:34 PM=
, Lijing (Jessie, Technology Introduction &amp; Standard and Patent Dept) &=
lt;<a href=3D"mailto:lijing80@huawei.com" target=3D"_blank">lijing80@huawei=
.com</a>&gt; wrote:<o:p></o:p></span></p>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<p class=3D"MsoNormal"><span lang=3D"EN-US">I fully agree with the idea to =
&quot; use existing &quot;native&quot; codecs &quot;.<br>
<br>
In addition, from a web developer's perspective, in order to make use of th=
e &quot;native&quot; codecs, the browser should provide codec APIs to open =
the local codec capabilities of the device to applications and developers. =
It really make sense.<o:p></o:p></span></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I don't understand what this me=
ans.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">-Ekr<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</body>
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sun, 21 Jul 2013 20:46:57 -0700
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To: "Lijing (Jessie, Technology Introduction & Standard and Patent Dept)" <lijing80@huawei.com>
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Subject: Re: [rtcweb] =?gb2312?b?tPC4tDogILTwuLQ6IFNvbWUgdGhvdWdodHMgb24gb3B0?= =?gb2312?b?aW9uYWwgYXVkaW8gY29kZWNz?=
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On Sun, Jul 21, 2013 at 8:42 PM, Lijing (Jessie, Technology Introduction &
Standard and Patent Dept) <lijing80@huawei.com> wrote:

>  Sorry=A3=ACI=A1=AFve confused two concepts. What I want to express is th=
at,****
>
> Firstly, the browser may be able to make use of =A1=B0native=A1=B1 codec =
APIs of the
> =A1=B0hard=A1=B1 device(e.g. mobile phone), and include the =A1=B0native=
=A1=B1 codecs in the
> SDP O/A messages.****
>
> In addition, for application flexibility, the browser should provide code=
c
> capability APIs to developers. Then,  the developers can choose proper
> codecs according to the specific scenario. ****
>
> ** **
>
> So there should be two APIs. The first is provide by the device and used
> by the browser.
>

Well, devices may or may not do this, but that's kind of out of scope for
this WG.


The second is provided by the browser and used by the developer.
>

Yes, this seems not unreasonable...

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sun, Jul 21, 2013 at 8:42 PM, Lijing (Jessie, Technology Introdu=
ction &amp; Standard and Patent Dept) <span dir=3D"ltr">&lt;<a href=3D"mail=
to:lijing80@huawei.com" target=3D"_blank">lijing80@huawei.com</a>&gt;</span=
> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"ZH-CN" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Sorry</spa=
n><span style=3D"font-size:10.5pt;color:#1f497d">=A3=AC</span><span lang=3D=
"EN-US" style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;san=
s-serif&quot;;color:#1f497d">I&rsquo;ve
 confused two concepts. What I want to express is that,<u></u><u></u></span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Firstly, t=
he browser may be able to make use of &ldquo;native&rdquo; codec APIs of th=
e &ldquo;hard&rdquo; device(e.g. mobile phone), and include the &ldquo;nati=
ve&rdquo; codecs in
 the SDP O/A messages.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">In additio=
n, for application flexibility, the browser should provide codec capability=
 APIs to developers. Then, &nbsp;the developers can choose proper
 codecs according to the specific scenario. <u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nb=
sp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">So there s=
hould be two APIs. The first is provide by the device and used by the brows=
er. </span></p>

</div></div></blockquote><div><br></div><div>Well, devices may or may not d=
o this, but that&#39;s kind of out of scope for</div><div>this WG.</div><di=
v><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div lang=3D"ZH-CN" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-family:&quot;Calibr=
i&quot;,&quot;sans-serif&quot;;color:#1f497d">The second is provided by the=
 browser and used by the
 developer.</span></p></div></div></blockquote><div><br></div><div>Yes, thi=
s seems not unreasonable...</div><div><br></div><div>-Ekr</div><div>&nbsp;<=
/div></div></div></div>

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Justin Uberti <juberti@google.com>
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From: <stephane.proust@orange.com>
To: '???/Lingli Deng' <denglingli@chinamobile.com>, 'Paul Coverdale' <coverdale@sympatico.ca>, "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Bogineni, Kalyani'" <Kalyani.Bogineni@VerizonWireless.com>, 'Bo Burman' <bo.burman@ericsson.com>,  "'rtcweb@ietf.org'" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Some thoughts on optional audio codecs
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Date: Mon, 22 Jul 2013 14:57:48 +0000
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Subject: Re: [rtcweb] Some thoughts on optional audio codecs
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From ralphm@ik.nu  Mon Jul 22 08:06:13 2013
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Date: Mon, 22 Jul 2013 17:05:41 +0200
From: Ralph Meijer <ralphm@ik.nu>
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Subject: [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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Hi all,

I would like to inform the group of the recent formation [1] of the 
Jingle Special Interest Group (SIG) at the XMPP Standards Foundation 
(XSF). The recent increase of activity in the WebRTC and rtcweb working 
groups and related high-profile product developments and announcements 
were reasons for the XMPP Council to decide to concentrate efforts 
around Jingle in a SIG.

Jingle [2] is a general framework for managing media sessions between 
XMPP Sessions, including, but not limited to, audio/video streams, file 
transfer and application sharing. There are several documents describing 
applications of Jingle and the used transports, most linked from the 
overall framework specification [3].

The specification of Jingle RTP Sessions [4], most relevant to these 
working groups, defines a Jingle application type for negotiating RTP 
sessions. It has been designed such that interoperability with SIP-based 
systems is possible. This includes mapping negotiation parameters to and 
from SDP, while remaining a signaling protocol in its own right (not 
merely SDP in angle brackets).

The following work items were defined in the kick-off meeting last 
Wednesday, July 17 [5, raw log 6]:

  * Re-examining the state of the various Jingle proposals.
  * Polishing Jingle File Transfer.
  * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
    improvements.
  * Documenting and communicating the value proposition of Jingle/XMPP.

This SIG already includes a number of people participating in 
discussions on the WebRTC and rtcweb mailing lists and is lead by Dave 
Cridland (chair), Philipp Hancke, Lance Stout and myself. It is open to 
anyone, and we are looking forward to cooperate with the WebRTC and 
rtcweb working groups to improve both WebRTC and Jingle.

The discussion venues are the Jingle mailing list [7] and the Jingle 
XMPP multi-user chat room [8]. Our next meeting in the MUC room is 
Wednesday July 24 at 15:30 UTC. This group's experience and input would 
be highly appreciated, and your participation in both the meeting and 
the on-going discussion would be most welcome.

Thanks,

Ralph Meijer

[1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
[2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
[3] <http://xmpp.org/extensions/xep-0166.html>
[4] <http://xmpp.org/extensions/xep-0167.html>
[5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
[6] <http://logs.xmpp.org/jingle/130717/>
[7] <http://mail.jabber.org/mailman/listinfo/jingle>
[8] <xmpp:jingle@muc.xmpp.org?join>

From ibc@aliax.net  Mon Jul 22 08:15:03 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 22 Jul 2013 17:14:36 +0200
Message-ID: <CALiegfk1kUuezLSOqfLRnFC7gNWXgjerv9Q_mPKrR01zp3mdqQ@mail.gmail.com>
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Cc: stox@ietf.org, XMPP Jingle <jingle@xmpp.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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Great. First thing you should complain about is the fact that current
WebRTC specification makes unfeasible for a browser to use SDP-XML as
defined by XEP-0167. So if you have a SIP server you will be able to
directly connect from the browser, but if you have a Jingle server you
will need a gateway.

2013/7/22 Ralph Meijer <ralphm@ik.nu>:
> Hi all,
>
> I would like to inform the group of the recent formation [1] of the Jingl=
e
> Special Interest Group (SIG) at the XMPP Standards Foundation (XSF). The
> recent increase of activity in the WebRTC and rtcweb working groups and
> related high-profile product developments and announcements were reasons =
for
> the XMPP Council to decide to concentrate efforts around Jingle in a SIG.
>
> Jingle [2] is a general framework for managing media sessions between XMP=
P
> Sessions, including, but not limited to, audio/video streams, file transf=
er
> and application sharing. There are several documents describing applicati=
ons
> of Jingle and the used transports, most linked from the overall framework
> specification [3].
>
> The specification of Jingle RTP Sessions [4], most relevant to these work=
ing
> groups, defines a Jingle application type for negotiating RTP sessions. I=
t
> has been designed such that interoperability with SIP-based systems is
> possible. This includes mapping negotiation parameters to and from SDP,
> while remaining a signaling protocol in its own right (not merely SDP in
> angle brackets).
>
> The following work items were defined in the kick-off meeting last
> Wednesday, July 17 [5, raw log 6]:
>
>  * Re-examining the state of the various Jingle proposals.
>  * Polishing Jingle File Transfer.
>  * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
>    improvements.
>  * Documenting and communicating the value proposition of Jingle/XMPP.
>
> This SIG already includes a number of people participating in discussions=
 on
> the WebRTC and rtcweb mailing lists and is lead by Dave Cridland (chair),
> Philipp Hancke, Lance Stout and myself. It is open to anyone, and we are
> looking forward to cooperate with the WebRTC and rtcweb working groups to
> improve both WebRTC and Jingle.
>
> The discussion venues are the Jingle mailing list [7] and the Jingle XMPP
> multi-user chat room [8]. Our next meeting in the MUC room is Wednesday J=
uly
> 24 at 15:30 UTC. This group's experience and input would be highly
> appreciated, and your participation in both the meeting and the on-going
> discussion would be most welcome.
>
> Thanks,
>
> Ralph Meijer
>
> [1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
> [2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
> [3] <http://xmpp.org/extensions/xep-0166.html>
> [4] <http://xmpp.org/extensions/xep-0167.html>
> [5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
> [6] <http://logs.xmpp.org/jingle/130717/>
> [7] <http://mail.jabber.org/mailman/listinfo/jingle>
> [8] <xmpp:jingle@muc.xmpp.org?join>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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On 7/22/13 5:14 PM, IÃ±aki Baz Castillo wrote:
> Great. First thing you should complain about is the fact that current
> WebRTC specification makes unfeasible for a browser to use SDP-XML as
> defined by XEP-0167. So if you have a SIP server you will be able to
> directly connect from the browser, but if you have a Jingle server you
> will need a gateway.
You are obviously misinforming here. SIP is the signaling protocol and a 
SIP server has really little to deal with SDP -- I'm sure you know that. 
And one cannot call directly a SIP endpoint from the browser, as SIP is 
not a mandatory signaling protocol, so there is extensive need of coding 
a javascript SIP stack (or reusing an existing one).

Cheers,
Daniel

>
> 2013/7/22 Ralph Meijer <ralphm@ik.nu>:
>> Hi all,
>>
>> I would like to inform the group of the recent formation [1] of the Jingle
>> Special Interest Group (SIG) at the XMPP Standards Foundation (XSF). The
>> recent increase of activity in the WebRTC and rtcweb working groups and
>> related high-profile product developments and announcements were reasons for
>> the XMPP Council to decide to concentrate efforts around Jingle in a SIG.
>>
>> Jingle [2] is a general framework for managing media sessions between XMPP
>> Sessions, including, but not limited to, audio/video streams, file transfer
>> and application sharing. There are several documents describing applications
>> of Jingle and the used transports, most linked from the overall framework
>> specification [3].
>>
>> The specification of Jingle RTP Sessions [4], most relevant to these working
>> groups, defines a Jingle application type for negotiating RTP sessions. It
>> has been designed such that interoperability with SIP-based systems is
>> possible. This includes mapping negotiation parameters to and from SDP,
>> while remaining a signaling protocol in its own right (not merely SDP in
>> angle brackets).
>>
>> The following work items were defined in the kick-off meeting last
>> Wednesday, July 17 [5, raw log 6]:
>>
>>   * Re-examining the state of the various Jingle proposals.
>>   * Polishing Jingle File Transfer.
>>   * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
>>     improvements.
>>   * Documenting and communicating the value proposition of Jingle/XMPP.
>>
>> This SIG already includes a number of people participating in discussions on
>> the WebRTC and rtcweb mailing lists and is lead by Dave Cridland (chair),
>> Philipp Hancke, Lance Stout and myself. It is open to anyone, and we are
>> looking forward to cooperate with the WebRTC and rtcweb working groups to
>> improve both WebRTC and Jingle.
>>
>> The discussion venues are the Jingle mailing list [7] and the Jingle XMPP
>> multi-user chat room [8]. Our next meeting in the MUC room is Wednesday July
>> 24 at 15:30 UTC. This group's experience and input would be highly
>> appreciated, and your participation in both the meeting and the on-going
>> discussion would be most welcome.
>>
>> Thanks,
>>
>> Ralph Meijer
>>
>> [1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
>> [2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
>> [3] <http://xmpp.org/extensions/xep-0166.html>
>> [4] <http://xmpp.org/extensions/xep-0167.html>
>> [5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
>> [6] <http://logs.xmpp.org/jingle/130717/>
>> [7] <http://mail.jabber.org/mailman/listinfo/jingle>
>> [8] <xmpp:jingle@muc.xmpp.org?join>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From ralphm@ik.nu  Mon Jul 22 08:47:26 2013
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Subject: [rtcweb] XSF Jingle Special Interest Group.
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On 2013-07-22 17:14, I=C3=B1aki Baz Castillo wrote:
> Great. First thing you should complain about is the fact that current
> WebRTC specification makes unfeasible for a browser to use SDP-XML as
> defined by XEP-0167. So if you have a SIP server you will be able to
> directly connect from the browser, but if you have a Jingle server you
> will need a gateway.


Hi I=C3=B1aki,

I have been following the recent discussions for a while now, and think=20
I understand the various stand points, including yours.

As I mentioned in my announcement, mapping SDP to Jingle's negotiation=20
parameters is indeed one of the concerns we are going to look into.=20
Philipp's work on his WebRTC plugin for Strophe.js [1], which uses=20
Jingle signalling, has shown that it indeed takes quite some effort to=20
map and mangle SDP to get things going. The spreadsheet questionnaire=20
that Peter Thatcher asked people to fill in (thanks!) also has several=20
comments to that effect. During our discussions on the Jingle mailing=20
list and the meeting last week, more of such comments were made.

I want to stress, again, that Jingle's negotiation parameters should not=20
be referred to as a flavor of SDP expressed in XML. This is misleading,=20
as it is not managed by MMUSIC, and actually weakens your point of not=20
wanting SDP as an API surface. I can understand that looking at the=20
current specification could give that impression, though. The Jingle=20
examples are currently interleaved by SDP examples to show the mapping,=20
and we are looking into making the distinction clearer, maybe even as a=20
document separate from this one.

[1] https://github.com/ESTOS/strophe.jingle

PS. Yes, I messed up the subject header of the original announcement.

--=20
ralphm

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2013/7/22 Ralph Meijer <ralphm@ik.nu>:
> Hi I=C3=B1aki,
>
> I have been following the recent discussions for a while now, and think I
> understand the various stand points, including yours.
>
> As I mentioned in my announcement, mapping SDP to Jingle's negotiation
> parameters is indeed one of the concerns we are going to look into.
> Philipp's work on his WebRTC plugin for Strophe.js [1], which uses Jingle
> signalling, has shown that it indeed takes quite some effort to map and
> mangle SDP to get things going. The spreadsheet questionnaire that Peter
> Thatcher asked people to fill in (thanks!) also has several comments to t=
hat
> effect. During our discussions on the Jingle mailing list and the meeting
> last week, more of such comments were made.
>
> I want to stress, again, that Jingle's negotiation parameters should not =
be
> referred to as a flavor of SDP expressed in XML. This is misleading, as i=
t
> is not managed by MMUSIC, and actually weakens your point of not wanting =
SDP
> as an API surface. I can understand that looking at the current
> specification could give that impression, though. The Jingle examples are
> currently interleaved by SDP examples to show the mapping, and we are
> looking into making the distinction clearer, maybe even as a document
> separate from this one.

Thanks for the clarification. So Jingle's session description is not
just a plain-to-XML mapping of a common SDP. Good to know.

However, wouldn't you prefer to have a pure JS Object with all the
media and transport information you require to build your Jingle
XML-SDP instead of having to parse a plain SDP string blob?

Thanks a lot.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From fippo@goodadvice.pages.de  Mon Jul 22 10:16:23 2013
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(removed stox)

Am 22.07.2013 17:14, schrieb IÃ±aki Baz Castillo:
> Great. First thing you should complain about is the fact that current
> WebRTC specification makes unfeasible for a browser to use SDP-XML as
> defined by XEP-0167.

I said this before, but since you insist on repeating your argument i'll 
repeat mine: I have running code doing exactly that.

It's hard work and there are some points where this is PITA, I have 
discovered numerous bugs in chrome (and the jingle spec) along the way, 
but it's certainly not unfeasible.


So far, this mapping works using the elements already defined in XEPs 
0167, 0293, 0294 and 0320 even. 
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-03#section-4.1 has a 
quite concrete list of features of what functionality the mapping 
between SDP and the xmlish SD has to define.

I'd note that I consider jingle a better way to talk about the session 
description because it has a very clear separation between codec 
negotation and transport which has led to concepts like trickle-ice. 
Also, it defined actions like content-add which seem to have influenced 
unified plan.


I have yet to see the current API fail completly and have been using it 
in ways that were certainly not imagined, e.g. early transport warmup 
using PRANSWER (and I still owe feedback on that to the JSEP authors; 
ironically, the solution is more SDP in the API).

From roman@telurix.com  Mon Jul 22 10:45:51 2013
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On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke
<fippo@goodadvice.pages.de>wrote:

> I said this before, but since you insist on repeating your argument i'll
> repeat mine: I have running code doing exactly that.
>
> It's hard work and there are some points where this is PITA, I have
> discovered numerous bugs in chrome (and the jingle spec) along the way, but
> it's certainly not unfeasible.
>
>
Out of curiosity, does the same code work with Firefox?
_____________
Roman Shpount

--047d7b874000de284f04e21d3e2b
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<div class=3D"gmail_quote">On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke =
<span dir=3D"ltr">&lt;<a href=3D"mailto:fippo@goodadvice.pages.de" target=
=3D"_blank">fippo@goodadvice.pages.de</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
I said this before, but since you insist on repeating your argument i&#39;l=
l repeat mine: I have running code doing exactly that.<br>
<br>
It&#39;s hard work and there are some points where this is PITA, I have dis=
covered numerous bugs in chrome (and the jingle spec) along the way, but it=
&#39;s certainly not unfeasible.<br>
<br></blockquote></div><br>Out of curiosity, does the same code work with F=
irefox?<br><div>_____________<br>Roman Shpount</div>
<br>

--047d7b874000de284f04e21d3e2b--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 22 Jul 2013 19:46:23 +0200
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2013/7/22 Philipp Hancke <fippo@goodadvice.pages.de>:
> Am 22.07.2013 17:14, schrieb I=C3=B1aki Baz Castillo:
>
>> Great. First thing you should complain about is the fact that current
>> WebRTC specification makes unfeasible for a browser to use SDP-XML as
>> defined by XEP-0167.
>
>
> I said this before, but since you insist on repeating your argument i'll
> repeat mine: I have running code doing exactly that.
>
> It's hard work and there are some points where this is PITA, I have
> discovered numerous bugs in chrome (and the jingle spec) along the way, b=
ut
> it's certainly not unfeasible.
>
>
> So far, this mapping works using the elements already defined in XEPs 016=
7,
> 0293, 0294 and 0320 even.
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-03#section-4.1 has a qu=
ite
> concrete list of features of what functionality the mapping between SDP a=
nd
> the xmlish SD has to define.


Thanks for the information.

Just wondering if you would feel more comfortable with a WebRTC API
that provides you real JS Objects/Classes with all the media
parameters and transport information (instead of a SDP blob string) so
you don't need to parse a plain-SDP (which will be different in each
vendor's browser and browser's release and others WebRTC compatible
devices) but instead take the data from JS Object structures and use
it to construct your own signaling protocol.

Thanks a lot.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2013/7/22 Roman Shpount <roman@telurix.com>:
> On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke <fippo@goodadvice.pages.d=
e>
> wrote:
>>
>> I said this before, but since you insist on repeating your argument i'll
>> repeat mine: I have running code doing exactly that.
>>
>> It's hard work and there are some points where this is PITA, I have
>> discovered numerous bugs in chrome (and the jingle spec) along the way, =
but
>> it's certainly not unfeasible.
>>
>
> Out of curiosity, does the same code work with Firefox?

And will it work with Opera, IE and any other browser without adapting
the code to the SDP those browsers will produce? and will it remain
working in future versions of Chrome and other browsers?

If it was a real JS API and the browsers properly implement it, then
the answer to all these questions would be "yes". But since this is
about dealing with a SDP blob string that can be represented in many
"compatible" ways by different browsers, I don't see it clear
(regardless I don't doubt at all about the quality of the JS code
which seems really good).

Regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: [rtcweb] DataChannel Questions
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--089e0112cf9a90755904e21da862
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I am sorry if I missed this somewhere in the draft, but I got two
DataChannel related questions:

First question: is it possible to bundle DataChannel with DTLS-SRTP? I do
not see how this can be implemented since DTLS in DataChannel can negotiate
any encryption protocol and I do not see a simple way to de-mux this on the
wire. Just wanted to confirm that bundling is not something that is planned
to be supported.

Second question: is there a way to time synchronize DataChannel messages
and real time media? For instance I want to use DataChannel for a subtitle
track, is there a way to synchronize messages with video track?
_____________
Roman Shpount

--089e0112cf9a90755904e21da862
Content-Type: text/html; charset=ISO-8859-1
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I am sorry if I missed this somewhere in the draft, but I got two DataChann=
el related questions:<br><br>First question: is it possible to bundle DataC=
hannel with DTLS-SRTP? I do not see how this can be implemented since DTLS =
in DataChannel can negotiate any encryption protocol and I do not see a sim=
ple way to de-mux this on the wire. Just wanted to confirm that bundling is=
 not something that is planned to be supported.<br>
<br>Second question: is there a way to time synchronize DataChannel message=
s and real time media? For instance I want to use DataChannel for a subtitl=
e track, is there a way to synchronize messages with video track?=A0 <br cl=
ear=3D"all">
<div>_____________<br>Roman Shpount</div>

--089e0112cf9a90755904e21da862--

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On Mon, Jul 22, 2013 at 11:15 AM, Roman Shpount <roman@telurix.com> wrote:

> I am sorry if I missed this somewhere in the draft, but I got two
> DataChannel related questions:
>
> First question: is it possible to bundle DataChannel with DTLS-SRTP? I do
> not see how this can be implemented since DTLS in DataChannel can negotiate
> any encryption protocol and I do not see a simple way to de-mux this on the
> wire.


I don't understand that "negotiate any encryption protocol" means. The SCTP
in the
DataChannel is encrypted under DTLS. The only potential problem here could
be
demuxing the SCTP from non-SCTP traffic once you have demuxed DTLS versus
SRTP, but we don't support anything that's non-SCTP here yet.



> Just wanted to confirm that bundling is not something that is planned to
> be supported.
>

As far as I know, it's fine.


Second question: is there a way to time synchronize DataChannel messages
> and real time media? For instance I want to use DataChannel for a subtitle
> track, is there a way to synchronize messages with video track?


You would presumably need to add the timing data into the DC messages
themselves, since SCTP does not provide any real-time function.

-Ekr

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<div dir=3D"ltr">On Mon, Jul 22, 2013 at 11:15 AM, Roman Shpount <span dir=
=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@t=
elurix.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=
=3D"gmail_quote">

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I am sorry if I missed this somewhere in the=
 draft, but I got two DataChannel related questions:<br><br>First question:=
 is it possible to bundle DataChannel with DTLS-SRTP? I do not see how this=
 can be implemented since DTLS in DataChannel can negotiate any encryption =
protocol and I do not see a simple way to de-mux this on the wire.</blockqu=
ote>

<div><br></div><div>I don&#39;t understand that &quot;negotiate any encrypt=
ion protocol&quot; means. The SCTP in the</div><div>DataChannel is encrypte=
d under DTLS. The only potential problem here could be</div><div>demuxing t=
he SCTP from non-SCTP traffic once you have demuxed DTLS versus</div>

<div>SRTP, but we don&#39;t support anything that&#39;s non-SCTP here yet.<=
/div><div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Just wa=
nted to confirm that bundling is not something that is planned to be suppor=
ted.<br>

</blockquote><div><br></div><div>As far as I know, it&#39;s fine.</div><div=
><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Second question: i=
s there a way to time synchronize DataChannel messages and real time media?=
 For instance I want to use DataChannel for a subtitle track, is there a wa=
y to synchronize messages with video track? =A0</blockquote>

<div><br></div><div>You would presumably need to add the timing data into t=
he DC messages</div><div>themselves, since SCTP does not provide any real-t=
ime function.</div><div><br></div><div>-Ekr</div><div><br></div></div>
</div>
</div>

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From fippo@goodadvice.pages.de  Mon Jul 22 11:27:53 2013
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Am 22.07.2013 19:45, schrieb Roman Shpount:
>> It's hard work and there are some points where this is PITA, I have
>> discovered numerous bugs in chrome (and the jingle spec) along the way, but
>> it's certainly not unfeasible.
>>
>>
> Out of curiosity, does the same code work with Firefox?

Yes, feel free to fork it at https://github.com/ESTOS/strophe.jingle and 
try the example. The core mapping has been relatively stable since I 
showed the first public version at the xmpp summit in Brussels six 
months ago.

Things like chrome suddenly (in canary) requiring a mapping of a=rtcp-fb 
weren't helpful and required extra work, but I do not expect this to 
happen too often once the SDP requirements are agreed upon.

It even sends to and receives rtp from Jitsi, even though no side is 
playing out anything yet; that can hopefully be resolved in Berlin.

From roman@telurix.com  Mon Jul 22 11:33:08 2013
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Thank you for you response. It is all clear now.
_____________
Roman Shpount

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<br>Thank you for you response. It is all clear now.<br><div>_____________<br>Roman Shpount</div>
<br>

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On 22 Jul 2013, at 18:45, Roman Shpount <roman@telurix.com> wrote:

> On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke =
<fippo@goodadvice.pages.de> wrote:
> I said this before, but since you insist on repeating your argument =
i'll repeat mine: I have running code doing exactly that.
>=20
> It's hard work and there are some points where this is PITA, I have =
discovered numerous bugs in chrome (and the jingle spec) along the way, =
but it's certainly not unfeasible.
>=20
>=20
> Out of curiosity, does the same code work with Firefox?

As far as Phono is concerned - yes - but we had to do some work to make =
that happen, more=20
work in fact than adding DTLS on the server side, which is (frankly) =
absurd.

T.



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<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>On 22 Jul 2013, at 18:45, Roman Shpount &lt;<a href="mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div class="gmail_quote">On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke <span dir="ltr">&lt;<a href="mailto:fippo@goodadvice.pages.de" target="_blank">fippo@goodadvice.pages.de</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I said this before, but since you insist on repeating your argument i'll repeat mine: I have running code doing exactly that.<br>
<br>
It's hard work and there are some points where this is PITA, I have discovered numerous bugs in chrome (and the jingle spec) along the way, but it's certainly not unfeasible.<br>
<br></blockquote></div><br>Out of curiosity, does the same code work with Firefox?<br></blockquote><div><br></div><div>As far as Phono is concerned - yes - but we had to do some work to make that happen, more&nbsp;</div><div>work in fact than adding DTLS on the server side, which is (frankly) absurd.</div><div><br></div><div>T.</div><div><br></div><div><br></div></div></body></html>
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Yes. It took some effort on our server side for us to support Firefox with
DTLS (still incomplete) as compared to Chrome with vanilla signaling. I
believe we have long way to build interop in browsers: RTP, JS API and
maybe Signaling (with webserver). ScreenSharing policy is yet to be
resolved.

I see this discussion is stuck at JS API vs JS SDP API, while equally
important standards like screensharing policy are in queue.

regards,
Rishi


On Mon, Jul 22, 2013 at 11:49 PM, tim panton <thp@westhawk.co.uk> wrote:

>
> On 22 Jul 2013, at 18:45, Roman Shpount <roman@telurix.com> wrote:
>
> On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke <fippo@goodadvice.pages.de
> > wrote:
>
>> I said this before, but since you insist on repeating your argument i'll
>> repeat mine: I have running code doing exactly that.
>>
>> It's hard work and there are some points where this is PITA, I have
>> discovered numerous bugs in chrome (and the jingle spec) along the way, but
>> it's certainly not unfeasible.
>>
>>
> Out of curiosity, does the same code work with Firefox?
>
>
> As far as Phono is concerned - yes - but we had to do some work to make
> that happen, more
> work in fact than adding DTLS on the server side, which is (frankly)
> absurd.
>
> T.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Yes. It took some effort on our server side for us to supp=
ort Firefox with DTLS (still incomplete) as compared to Chrome with vanilla=
 signaling. I believe we have long way to build interop in browsers: RTP, J=
S API and maybe Signaling (with webserver). ScreenSharing policy is yet to =
be resolved.=A0<div>
<br></div><div>I see this discussion is stuck at JS API vs JS SDP API, whil=
e equally important standards like screensharing policy are in queue.<br><d=
iv><br></div><div>regards,</div><div>Rishi</div><div><br><div><div class=3D=
"gmail_extra">
<br><div class=3D"gmail_quote">On Mon, Jul 22, 2013 at 11:49 PM, tim panton=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:thp@westhawk.co.uk" target=3D"_bla=
nk">thp@westhawk.co.uk</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
<div style=3D"word-wrap:break-word"><br><div><div><div class=3D"h5"><div>On=
 22 Jul 2013, at 18:45, Roman Shpount &lt;<a href=3D"mailto:roman@telurix.c=
om" target=3D"_blank">roman@telurix.com</a>&gt; wrote:</div><br><blockquote=
 type=3D"cite">
<div class=3D"gmail_quote">On Mon, Jul 22, 2013 at 1:00 PM, Philipp Hancke =
<span dir=3D"ltr">&lt;<a href=3D"mailto:fippo@goodadvice.pages.de" target=
=3D"_blank">fippo@goodadvice.pages.de</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">

I said this before, but since you insist on repeating your argument i&#39;l=
l repeat mine: I have running code doing exactly that.<br>
<br>
It&#39;s hard work and there are some points where this is PITA, I have dis=
covered numerous bugs in chrome (and the jingle spec) along the way, but it=
&#39;s certainly not unfeasible.<br>
<br></blockquote></div><br>Out of curiosity, does the same code work with F=
irefox?<br></blockquote><div><br></div></div></div><div>As far as Phono is =
concerned - yes - but we had to do some work to make that happen, more=A0</=
div>
<div>work in fact than adding DTLS on the server side, which is (frankly) a=
bsurd.</div><span class=3D"HOEnZb"><font color=3D"#888888"><div><br></div><=
div>T.</div><div><br></div><div><br></div></font></span></div></div><br>___=
____________________________________________<br>

rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div></div></div></div>

--047d7bd6be1268331d04e21f19e7--

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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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On 7/18/13 07:28, Rajmohan Banavi wrote:
> Hi  Justin, comments inline with additional comments at the end.
>
>     Correct, but that is a significant advantage, especially when the
>     web and TURN servers are separate entities.
>
>
> IMHO, the standard spec should not be based on any specific 
> implementation method. Now I agree that the TURN server validation 
> becomes easier using this method because it does not have to do some 
> sort of lookup. However, some another TURN server implementer can 
> achieve the same functionality by providing REST API but using 
> randomly generated pasword (rather than use HMAC) and using some sort 
> of lookup for TURN username when TURN ALLOCATE request is received.

I think the value here -- the value in defining a common scheme for 
password generation -- is that I could get an off-the-shelf TURN server 
from arbitrary vendor A, and run an off-the-shelf credential server from 
abitrary vendor B, and it would all work together. What you propose 
requires collusion between the TURN server provider and the credential 
server provider, using a not-yet-defined (and, if I read your proposal, 
never-publicly-defined) protocol to share credential information via a 
back-channel.

What Justin proposes gets us this inter-vendor interop cheaply and 
easily. What you're counterproposing forces people into a single-vendor 
(or, at least, partnered) solution, which kinda defeats the purpose of 
defining this in an SDO in the first place.

/a

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    <div class="moz-cite-prefix">On 7/18/13 07:28, Rajmohan Banavi
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAJWm+fE1G2r0TcUAcZUVCP0WRSC35JFBdZ-oMqJfAykhNExqyA@mail.gmail.com"
      type="cite">
      <div dir="ltr">Hi &nbsp;Justin, comments inline with additional
        comments at the end.
        <div class="gmail_extra"><br>
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div dir="ltr">
                <div class="gmail_extra">
                  <div class="gmail_quote">
                    <div>Correct, but that is a significant advantage,
                      especially when the web and TURN servers are
                      separate entities.&nbsp;</div>
                  </div>
                </div>
              </div>
            </blockquote>
            <div>
              <br>
            </div>
            <div>IMHO, the standard spec should not be based on any
              specific implementation method. Now I agree that the TURN
              server validation becomes easier using this method because
              it does not have to do some sort of lookup. However, some
              another TURN server implementer can achieve the same
              functionality by providing REST API but using randomly
              generated pasword (rather than use HMAC) and using some
              sort of lookup for TURN username when TURN ALLOCATE
              request is received.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I think the value here -- the value in defining a common scheme for
    password generation -- is that I could get an off-the-shelf TURN
    server from arbitrary vendor A, and run an off-the-shelf credential
    server from abitrary vendor B, and it would all work together. What
    you propose requires collusion between the TURN server provider and
    the credential server provider, using a not-yet-defined (and, if I
    read your proposal, never-publicly-defined) protocol to share
    credential information via a back-channel.<br>
    <br>
    What Justin proposes gets us this inter-vendor interop cheaply and
    easily. What you're counterproposing forces people into a
    single-vendor (or, at least, partnered) solution, which kinda
    defeats the purpose of defining this in an SDO in the first place.<br>
    <br>
    /a<br>
  </body>
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--------------070307020801070707040107--

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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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Am 22.07.2013 22:16, schrieb Adam Roach:
> What Justin proposes gets us this inter-vendor interop cheaply and
> easily. What you're counterproposing forces people into a single-vendor
> (or, at least, partnered) solution, which kinda defeats the purpose of
> defining this in an SDO in the first place.

Right. What Justin proposed was originally located at
https://code.google.com/p/webrtc/issues/detail?id=1197
I liked the plan, implemented it in restund and improved it a litle. It 
was discussed at 
https://groups.google.com/forum/#!topic/discuss-webrtc/nn8b6UboqRA/discussion 
and also implemented in rfc-5766-turn-server.

Rough consensus and running code...

From fineberg@vline.me  Tue Jul 23 08:54:06 2013
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I've been thinking about this and given the ease at which RFC5285 allows 
for the specification of a header extension and the complexity 
introduced by trying to generalize the header extension to support all 
forms of scalability for all codecs that the generalization might not be 
the best approach.  I'm not sure what we really gain by trying to 
capture all this in a single header extension rather than one per that 
can succinctly explain the fields without the complexity of multiplexing 
the bits.

Thoughts?

Regards,
Adam

On 7/19/13 3:44 PM, Stephan Wenger wrote:
> Hi,
>
> From: Adam Fineberg <fineberg@vline.me <mailto:fineberg@vline.me>>
> Date: Friday, 19 July, 2013 15:12
> To: Stephan Wenger <stewe@stewe.org <mailto:stewe@stewe.org>>
> Cc: Justin Uberti <juberti@google.com <mailto:juberti@google.com>>, 
> Bernard Aboba <bernard_aboba@hotmail.com 
> <mailto:bernard_aboba@hotmail.com>>, "avtext@ietf.org 
> <mailto:avtext@ietf.org>" <avtext@ietf.org <mailto:avtext@ietf.org>>, 
> "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org 
> <mailto:rtcweb@ietf.org>>
> Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Stephan,
>
> Thanks for the info and the reference.  I'm not sure I follow as I'm 
> not at all familiar with H.265.  I'll review the reference and see 
> what I can figure.
>
> StW: Good luck :-)
>
> It seems though to me that you are suggesting that except in the 
> simple case, that the data for H.265 would not be well suited to a 
> header extension, am I understanding you correctly?  There is no 
> reason the middlebox couldn't get out of band signaling of the VPS as 
> you mention but that would not be within the scope of this header 
> extension.
>
> StW: well, if you would copy the layer_id into your header extension 
> (just as you need to do for the simple case), a really smart middle 
> box could use this information just as a decoder uses 
> it, assuming that it intercepted the VPS in the first place.  Insofar, 
> I wouldn't rule out the second option on technical grounds.  Whether 
> any of the actual products would bother to do that, ever, is another 
> question.  I think the case ought to be documented, though.  I can 
> help drafting text.
> While we are at it: doing this right could mean that you need multiple 
> specs.  First, a generic header extension mechanism dedicated to side 
> information required for pruning of RTP packet streams—ideally not 
> only for scalable video, although that is the main customer today. 
>  And second, for each "payload" (at present we are talking about 
> H.264/SVC, H.265v1 (HEVC), H.265v2 (including scalable and 3D 
> extensions, which are not yet finalized), VP8, and VP9 (I know little 
> about the latter), plus Daala, and whatnot) a mapping of the bits 
> available in the generic header extension to the bits in the payload 
> itself (NAL header and VPS in case of H.265, NALU header in SVC, and 
> the fields you mention for VP8).
>
> Stephan
>
>
> Any insights are appreciated.
>
> Regards,
> Adam
>
> On 7/19/13 8:33 AM, Stephan Wenger wrote:
>> Hi,
>> I also believe that 16 bits should be enough.  For H.264 and VP8 that 
>> has already been demonstrated.  For H.265, some initial thoughts 
>> below.  Apologies for the word-count.
>>
>> The scalable version of H265 (called SHVC) is currently under 
>> development.  The current working draft can be found here: 
>> http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip. 
>>  Therein, the options for defining layering structures are 
>> considerably more complex.  To start, we have 3 bits for the temporal 
>> ID in the NAL unit header of the H.265 version 1 (HEVC) base 
>> specification (temporal scalability is already nicely supported in 
>> version 1).  Just like in SVC.  In the scalable extension, the NAL 
>> unit header contains a six bit field that points into a data 
>> structure known as "Video Parameter Set" (VPS).  Inside the VPS, 
>> those six bits are mapped to to a position in a directed graph 
>> (specified through "dimension_id[][]"), which tells you about the 
>> reference relationship of the layer in question and its parent layer. 
>>  One can recursively follow the graph to determine what used to be 
>> called dependency_id, quality_id, view_id, and whatnot.  The six bit 
>> pointer field can (or: is to be when possible) organized by the 
>> encoder such that it is prudent for a middle box to throw away NAL 
>> units (belonging to layers) with higher values of the six bit field 
>> first, before throwing away NAL units with lower values.  Relying on 
>> this feature, 3+6 bits == 9 bits should be fine for the header extension.
>>
>> That said, the ordering by the encoder is just a recommendation, and 
>> there may well be cases where different pruning strategies may be 
>> advisable.  For example, a layering structure could be constructed 
>> that expands into two branches, one using 2D scalable tools only, the 
>> other including view_id for multi view coding.  By looking at the six 
>> bit field alone, a middle box will not be able to meaningfully remove 
>> NAL units belonging to one of the branches completely while pruning 
>> the other branch.  In order to meaningfully deal with that scenario, 
>> there would be two options: one to represent the dimension_id[][] 
>> (and associated control info) in the header extension, or require the 
>> middle box to have access to the VPS and be able to interpret its 
>> content.  The further could take considerably more than 16 bits and 
>> we would be talking about a variable length data structure.  The 
>> latter requires the middle box to have state and a mechanism to 
>> intercept the VPS (through signaling—as the encrypted in-band VPS 
>> would not be useful under the assumption that the middle box does not 
>> have the key to the media—which is the motivation of the draft in the 
>> first place).  I personally don't mind at all the second mechanism, 
>> as I'm a big fan of out-of-band parameter set transmission and any 
>> middle box must be in the signaling path anyway to meaningfully 
>> manipulate RTP.  I do not like the first option due to its variable, 
>> and possibly substantial, overhead.
>>
>> Stephan
>>
>>
>> From: Justin Uberti <juberti@google.com <mailto:juberti@google.com>>
>> Date: Friday, 19 July, 2013 06:32
>> To: Bernard Aboba <bernard_aboba@hotmail.com 
>> <mailto:bernard_aboba@hotmail.com>>
>> Cc: "avtext@ietf.org <mailto:avtext@ietf.org>" <avtext@ietf.org 
>> <mailto:avtext@ietf.org>>, "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" 
>> <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>> Subject: Re: [rtcweb] Fwd: New Version Notification for 
>> draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>> Agree those are the right codecs to design for. Since in each case 
>> there are fairly low limits on the number of supported layers (i.e. 3 
>> spatial layers for SVC), I think it should be possible to pack the 
>> temporal, spatial, quality layer ids into 16 bits.
>>
>>
>> On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba 
>> <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>> wrote:
>>
>>     If we can support VP8/9 as well as H.264/5 SVC
>>     that would be a start. It seems doable to me.
>>
>>     On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me
>>     <mailto:fineberg@vline.me>> wrote:
>>
>>>     Bernard,
>>>
>>>     Are there other codecs you are thinking should be supported? If
>>>     it's generalized I would think we want to be able to cover all
>>>     known scalable codecs. I'll look into the H264/SVC fields to see
>>>     how to encode them in a generalized header.
>>>
>>>     Regards,
>>>     Adam
>>>
>>>     On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>>>     I think it may be possible to generalize this. For example, for
>>>>     H.264/SVC which can support temporal, spatial and quality
>>>>     scalability, you would need the quality_id and dependency_id in
>>>>     addition to the temporal_id (what you call the temporal layer
>>>>     index).
>>>>
>>>>     ------------------------------------------------------------------------
>>>>     Date: Thu, 18 Jul 2013 08:45:38 -0700
>>>>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>>     To: bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>
>>>>     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>; avtext@ietf.org
>>>>     <mailto:avtext@ietf.org>
>>>>     Subject: Re: [rtcweb] Fwd: New Version Notification for
>>>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>
>>>>     Bernard,
>>>>
>>>>     Good question.  I'm not familiar enough with the parameter
>>>>     requirements of all other scalable codecs to be able to
>>>>     generalize.  If you'd like to help specify them, I'd be fine
>>>>     revising the draft to generalize.
>>>>
>>>>     Regards,
>>>>     Adam
>>>>
>>>>     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>>>
>>>>         Since the need is not codec specific (e.g. it arises with
>>>>         any codec supporting temporal, spatial and quality
>>>>         scalability), why
>>>>          a VP8-specific RTP extension?
>>>>
>>>>         ------------------------------------------------------------------------
>>>>         Date: Wed, 17 Jul 2013 17:09:46 -0700
>>>>         From: fineberg@vline.me <mailto:fineberg@vline.me>
>>>>         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>>         Subject: [rtcweb] Fwd: New Version Notification for
>>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>
>>>>         Hi,
>>>>
>>>>         I'm working on WebRTC services and have found that while
>>>>         developing services that forward VP8 video streams if we
>>>>         want to take advantage of the VP8 temporal scaling we must
>>>>         get the temporal layer information from the RTP header
>>>>         which requires us to decrypt the SRTP packets. This is
>>>>         undesirable both because the middle-box needs to have
>>>>         access to the keys as well as the because of the added
>>>>         overhead of the decrypt/encrypt cycle. This draft proposes
>>>>         an RTP header extension that will allow us to use the VP8
>>>>         temporal layer information included in the header extension
>>>>         and therefore do forwarding without SRTP decryption.
>>>>         Comments welcome.
>>>>
>>>>         Regards,
>>>>         Adam Fineberg
>>>>         fineberg at vline.com <mailto:fineberg%20at%20vline.com>
>>>>
>>>>         -------- Original Message --------
>>>>         Subject: 	New Version Notification for
>>>>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>>>>         From: 	internet-drafts at ietf.org
>>>>         <mailto:internet-drafts%20at%20ietf.org>
>>>>         To: 	Adam Fineberg<fineberg at vline.com>
>>>>         <mailto:fineberg%20at%20vline.com>
>>>>
>>>>
>>>>
>>>>         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         has been successfully submitted by Adam Fineberg and posted to the
>>>>         IETF repository.
>>>>
>>>>         Filename:	 draft-fineberg-avtext-temporal-layer-ext
>>>>         Revision:	 00
>>>>         Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>>>>         Creation date:	 2013-07-08
>>>>         Group:		 Individual Submission
>>>>         Number of pages: 6
>>>>         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>>>>         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>>>>         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>>>
>>>>
>>>>         Abstract:
>>>>             This document defines a mechanism by which packets of Real-Time
>>>>             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>>>             indicate, in an RTP header extension, the temporal layer information
>>>>             about the frame encoded in the RTP packet.  This information can be
>>>>             used in a middlebox performing bandwidth management of streams
>>>>             without requiring it to decrypt the streams.
>>>>
>>>>
>>>>         _______________________________________________ rtcweb
>>>>         mailing list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>>         https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>>     -- 
>>>>     Regards,
>>>>     Adam
>>>
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>>
>> _______________________________________________
>> avtext mailing list
>> avtext@ietf.orghttps://www.ietf.org/mailman/listinfo/avtext
>
> -- 
> Regards,
> Adam

-- 
Regards,
Adam


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<html>
  <head>
    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I've been thinking about this and given the ease at which RFC5285
    allows for the specification of a header extension and the
    complexity introduced by trying to generalize the header extension
    to support all forms of scalability for all codecs that the
    generalization might not be the best approach.  I'm not sure what we
    really gain by trying to capture all this in a single header
    extension rather than one per that can succinctly explain the fields
    without the complexity of multiplexing the bits.<br>
    <br>
    Thoughts?<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/19/13 3:44 PM, Stephan Wenger
      wrote:<br>
    </div>
    <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">Hi,</font></div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div style="font-family:Calibri; font-size:11pt;
          text-align:left; color:black; BORDER-BOTTOM: medium none;
          BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT:
          0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;
          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
          <span style="font-weight:bold">From: </span>Adam Fineberg
          &lt;<a moz-do-not-send="true" href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
          <span style="font-weight:bold">Date: </span>Friday, 19 July,
          2013 15:12 <br>
          <span style="font-weight:bold">To: </span>Stephan Wenger &lt;<a
            moz-do-not-send="true" href="mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
          <span style="font-weight:bold">Cc: </span>Justin Uberti &lt;<a
            moz-do-not-send="true" href="mailto:juberti@google.com">juberti@google.com</a>&gt;,
          Bernard Aboba &lt;<a moz-do-not-send="true"
            href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
          <span style="font-weight:bold">Subject: </span>Re: [avtext]
          [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
        </div>
        <div><br>
        </div>
        <div>
          <div bgcolor="#FFFFFF" text="#000000">Stephan,<br>
            <br>
            Thanks for the info and the reference.  I'm not sure I
            follow as I'm not at all familiar with H.265.  I'll review
            the reference and see what I can figure. </div>
        </div>
      </span>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">StW: Good luck :-)  </font></div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div>
          <div bgcolor="#FFFFFF" text="#000000">It seems though to me
            that you are suggesting that except in the simple case, that
            the data for H.265 would not be well suited to a header
            extension, am I understanding you correctly?  There is no
            reason the middlebox couldn't get out of band signaling of
            the VPS as you mention but that would not be within the
            scope of this header extension.</div>
        </div>
      </span>
      <div style="font-family: Calibri, sans-serif; font-size: 14px;
        color: rgb(0, 0, 0); ">
        <br>
      </div>
      <div><font color="#0000ff"><font face="Calibri,sans-serif">StW:
            well, if you would copy the layer_id into your header
            extension (just as you need to do for the simple case), a
            really smart middle box could use this information just as a
            decoder uses it, assuming that it intercepted the VPS in the
            first place.  Insofar, I wouldn't rule out the second option
            on technical grounds.  Whether any of the actual products
            would bother to do that, ever, is another question.  I think
            the case ought to be documented, though.  I can help
            drafting text.</font></font></div>
      <div><font color="#0000ff"><font face="Calibri,sans-serif">While
            we are at it: doing this right could mean that you need
            multiple specs.  First, a generic header extension mechanism
            dedicated to side information required for pruning of RTP
            packet streams—ideally not only for scalable video, although
            that is the main customer today.  And second, for each
            "payload" (at present we are talking about H.264/SVC,
            H.265v1 (HEVC), H.265v2 (including scalable and 3D
            extensions, which are not yet finalized), VP8, and VP9 (I
            know little about the latter), plus Daala, and whatnot) a
            mapping of the bits available in the generic header
            extension to the bits in the payload itself (NAL header and
            VPS in case of H.265, NALU header in SVC, and the fields you
            mention for VP8).</font></font></div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><br>
      </div>
      <div style="font-family: Calibri, sans-serif; font-size: 14px; "><font
          color="#0000ff">Stephan</font></div>
      <span id="OLK_SRC_BODY_SECTION" style="font-family: Calibri,
        sans-serif; font-size: 14px; color: rgb(0, 0, 0); ">
        <div>
          <div bgcolor="#FFFFFF" text="#000000"><br>
            <br>
            Any insights are appreciated.<br>
            <br>
            Regards,<br>
            Adam<br>
            <br>
            <div class="moz-cite-prefix">On 7/19/13 8:33 AM, Stephan
              Wenger wrote:<br>
            </div>
            <blockquote cite="mid:CE0E9F56.9F89B%25stewe@stewe.org"
              type="cite">
              <div>Hi,</div>
              <div>I also believe that 16 bits should be enough.  For
                H.264 and VP8 that has already been demonstrated.  For
                H.265, some initial thoughts below.  Apologies for the
                word-count.</div>
              <div><br>
              </div>
              <div>The scalable version of H265 (called SHVC) is
                currently under development.  The current working draft
                can be found here: <a moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
                 Therein, the options for defining layering structures
                are considerably more complex.  To start, we have 3 bits
                for the temporal ID in the NAL unit header of the H.265
                version 1 (HEVC) base specification (temporal
                scalability is already nicely supported in version 1).
                 Just like in SVC.  In the scalable extension, the NAL
                unit header contains a six bit field that points into a
                data structure known as "Video Parameter Set" (VPS).
                 Inside the VPS, those six bits are mapped to to a
                position in a directed graph (specified through
                "dimension_id[][]"), which tells you about the reference
                relationship of the layer in question and its parent
                layer.  One can recursively follow the graph to
                determine what used to be called dependency_id,
                quality_id, view_id, and whatnot.  The six bit pointer
                field can (or: is to be when possible) organized by the
                encoder such that it is prudent for a middle box to
                throw away NAL units (belonging to layers) with higher
                values of the six bit field first, before throwing away
                NAL units with lower values.  Relying on this feature,
                3+6 bits == 9 bits should be fine for the header
                extension.</div>
              <div><br>
              </div>
              <div>That said, the ordering by the encoder is just a
                recommendation, and there may well be cases where
                different pruning strategies may be advisable.  For
                example, a layering structure could be constructed that
                expands into two branches, one using 2D scalable tools
                only, the other including view_id for multi view coding.
                 By looking at the six bit field alone, a middle box
                will not be able to meaningfully remove NAL units
                belonging to one of the branches completely while
                pruning the other branch.  In order to meaningfully deal
                with that scenario, there would be two options: one to
                represent the dimension_id[][] (and associated control
                info) in the header extension, or require the middle box
                to have access to the VPS and be able to interpret its
                content.  The further could take considerably more than
                16 bits and we would be talking about a variable length
                data structure.  The latter requires the middle box to
                have state and a mechanism to intercept the VPS (through
                signaling—as the encrypted in-band VPS would not be
                useful under the assumption that the middle box does not
                have the key to the media—which is the motivation of the
                draft in the first place).  I personally don't mind at
                all the second mechanism, as I'm a big fan of
                out-of-band parameter set transmission and any middle
                box must be in the signaling path anyway to meaningfully
                manipulate RTP.  I do not like the first option due to
                its variable, and possibly substantial, overhead.</div>
              <div><br>
              </div>
              <div>Stephan   </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <span id="OLK_SRC_BODY_SECTION">
                <div style="font-family:Calibri; font-size:11pt;
                  text-align:left; color:black; BORDER-BOTTOM: medium
                  none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in;
                  PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP:
                  #b5c4df 1pt solid; BORDER-RIGHT: medium none;
                  PADDING-TOP: 3pt">
                  <span style="font-weight:bold">From: </span>Justin
                  Uberti &lt;<a moz-do-not-send="true"
                    href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
                  <span style="font-weight:bold">Date: </span>Friday,
                  19 July, 2013 06:32 <br>
                  <span style="font-weight:bold">To: </span>Bernard
                  Aboba &lt;<a moz-do-not-send="true"
                    href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
                  <span style="font-weight:bold">Cc: </span>"<a
                    moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                  &lt;<a moz-do-not-send="true"
                    href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                  "<a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                  &lt;<a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                  <span style="font-weight:bold">Subject: </span>Re:
                  [rtcweb] Fwd: New Version Notification for
                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                </div>
                <div><br>
                </div>
                <div>
                  <div>
                    <div dir="ltr">Agree those are the right codecs to
                      design for. Since in each case there are fairly
                      low limits on the number of supported layers (i.e.
                      3 spatial layers for SVC), I think it should be
                      possible to pack the temporal, spatial, quality
                      layer ids into 16 bits.
                      <div class="gmail_extra"><br>
                        <br>
                        <div class="gmail_quote">On Fri, Jul 19, 2013 at
                          1:56 AM, Bernard Aboba <span dir="ltr">
                            &lt;<a moz-do-not-send="true"
                              href="mailto:bernard_aboba@hotmail.com"
                              target="_blank">bernard_aboba@hotmail.com</a>&gt;</span>
                          wrote:<br>
                          <blockquote class="gmail_quote"
                            style="margin:0 0 0 .8ex;border-left:1px
                            #ccc solid;padding-left:1ex">
                            <div dir="auto">
                              <div>If we can support VP8/9 as well as
                                H.264/5 SVC</div>
                              <div>that would be a start. It seems
                                doable to me.</div>
                              <div>
                                <div>
                                  <div><br>
                                    On Jul 18, 2013, at 8:34 PM, "Adam
                                    Fineberg" &lt;<a
                                      moz-do-not-send="true"
                                      href="mailto:fineberg@vline.me"
                                      target="_blank">fineberg@vline.me</a>&gt;
                                    wrote:<br>
                                    <br>
                                  </div>
                                  <blockquote type="cite">
                                    <div>
                                      <div>Bernard,<br>
                                        <br>
                                        Are there other codecs you are
                                        thinking should be supported? 
                                        If it's generalized I would
                                        think we want to be able to
                                        cover all known scalable codecs.
                                        I'll look into the H264/SVC
                                        fields to see how to encode them
                                        in a generalized header.<br>
                                        <br>
                                        Regards,<br>
                                        Adam<br>
                                        <br>
                                        On 7/18/13 7:40 PM, Bernard
                                        Aboba wrote:<br>
                                      </div>
                                      <blockquote type="cite">
                                        <div dir="ltr">I think it may be
                                          possible to generalize this. 
                                          For example, for H.264/SVC
                                          which can support temporal,
                                          spatial and quality
                                          scalability, you would need
                                          the quality_id and
                                          dependency_id in addition to
                                          the temporal_id (what you call
                                          the temporal layer index).   
                                          <br>
                                          <br>
                                          <div>
                                            <hr>
                                            Date: Thu, 18 Jul 2013
                                            08:45:38 -0700<br>
                                            From: <a
                                              moz-do-not-send="true"
                                              href="mailto:fineberg@vline.me"
                                              target="_blank">fineberg@vline.me</a><br>
                                            To: <a
                                              moz-do-not-send="true"
                                              href="mailto:bernard_aboba@hotmail.com"
                                              target="_blank">
                                              bernard_aboba@hotmail.com</a><br>
                                            CC: <a
                                              moz-do-not-send="true"
                                              href="mailto:rtcweb@ietf.org"
                                              target="_blank">rtcweb@ietf.org</a>;
                                            <a moz-do-not-send="true"
                                              href="mailto:avtext@ietf.org"
                                              target="_blank">avtext@ietf.org</a><br>
                                            Subject: Re: [rtcweb] Fwd:
                                            New Version Notification for
draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                            <br>
                                            Bernard,<br>
                                            <br>
                                            Good question.  I'm not
                                            familiar enough with the
                                            parameter requirements of
                                            all other scalable codecs to
                                            be able to generalize.  If
                                            you'd like to help specify
                                            them, I'd be fine revising
                                            the draft to generalize.<br>
                                            <br>
                                            Regards,<br>
                                            Adam<br>
                                            <br>
                                            <div>On 7/17/13 8:26 PM,
                                              Bernard Aboba wrote:<br>
                                            </div>
                                            <blockquote>
                                              <div dir="ltr">Since the
                                                need is not codec
                                                specific (e.g. it arises
                                                with any codec
                                                supporting temporal,
                                                spatial and quality
                                                scalability), why
                                                <br>
                                                 a VP8-specific RTP
                                                extension? <br>
                                                 <br>
                                                <div>
                                                  <hr>
                                                  Date: Wed, 17 Jul 2013
                                                  17:09:46 -0700<br>
                                                  From: <a
                                                    moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                  To: <a
                                                    moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                  Subject: [rtcweb] Fwd:
                                                  New Version
                                                  Notification for
                                                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                  <br>
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Hi,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">I'm
                                                    working on WebRTC
                                                    services and have
                                                    found that while
                                                    developing services
                                                    that forward VP8
                                                    video streams if we
                                                    want to take
                                                    advantage of the VP8
                                                    temporal scaling we
                                                    must get the
                                                    temporal layer
                                                    information from the
                                                    RTP header which
                                                    requires us to
                                                    decrypt the SRTP
                                                    packets. This is
                                                    undesirable both
                                                    because the
                                                    middle-box needs to
                                                    have access to the
                                                    keys as well as the
                                                    because of the added
                                                    overhead of the
                                                    decrypt/encrypt
                                                    cycle. This draft
                                                    proposes an RTP
                                                    header extension
                                                    that will allow us
                                                    to use the VP8
                                                    temporal layer
                                                    information included
                                                    in the header
                                                    extension and
                                                    therefore do
                                                    forwarding without
                                                    SRTP decryption.
                                                    Comments welcome.</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Regards,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <span
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline!important;word-spacing:0px">Adam
                                                    Fineberg</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px">
                                                  <div
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px"><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                      rel="nofollow"
                                                      target="_blank">fineberg
                                                      at vline.com</a><br>
                                                    <br>
                                                    -------- Original
                                                    Message --------
                                                    <table border="0"
                                                      cellpadding="0"
                                                      cellspacing="0">
                                                      <tbody>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Subject:</th>
                                                          <td>New
                                                          Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Date:</th>
                                                          <td>Tue, 09
                                                          Jul 2013
                                                          10:02:05 -0700</td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">From:</th>
                                                          <td><a
                                                          moz-do-not-send="true"
href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow"
                                                          target="_blank">internet-drafts
                                                          at ietf.org</a></td>
                                                        </tr>
                                                        <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">To:</th>
                                                          <td>Adam
                                                          Fineberg<span> </span><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          rel="nofollow"
target="_blank">&lt;fineberg at vline.com&gt;</a></td>
                                                        </tr>
                                                      </tbody>
                                                    </table>
                                                    <br>
                                                    <br>
                                                    <pre style="width:939.59px;white-space:pre-wrap">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel="nofollow" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel="nofollow" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" rel="nofollow" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                                                  </div>
                                                  <br>
                                                  _______________________________________________
                                                  rtcweb mailing list <a
moz-do-not-send="true" href="mailto:rtcweb@ietf.org" target="_blank">
                                                    rtcweb@ietf.org</a>
                                                  <a
                                                    moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
                                              </div>
                                            </blockquote>
                                            <br>
                                            <pre>-- 
Regards,
Adam</pre>
                                          </div>
                                        </div>
                                      </blockquote>
                                      <br>
                                    </div>
                                  </blockquote>
                                </div>
                              </div>
                            </div>
                            <br>
_______________________________________________<br>
                            rtcweb mailing list<br>
                            <a moz-do-not-send="true"
                              href="mailto:rtcweb@ietf.org"
                              target="_blank">rtcweb@ietf.org</a><br>
                            <a moz-do-not-send="true"
                              href="https://www.ietf.org/mailman/listinfo/rtcweb"
                              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                            <br>
                          </blockquote>
                        </div>
                        <br>
                      </div>
                    </div>
                  </div>
                </div>
              </span><br>
              <fieldset class="mimeAttachmentHeader"></fieldset>
              <br>
              <pre wrap="">_______________________________________________
avtext mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/avtext">https://www.ietf.org/mailman/listinfo/avtext</a></pre>
            </blockquote>
            <br>
            <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
          </div>
        </div>
      </span>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
  </body>
</html>

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From roman@telurix.com  Tue Jul 23 09:22:21 2013
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Date: Tue, 23 Jul 2013 12:22:10 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: [rtcweb] Unified plan IPR
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Changing the title and the mailing list to the more appropriate.

The situation would be a bit clearer if patent holders were to provide the
licensing policy regarding this IPR release. Given that Ericsson is
actively involved in this working group, I think it would be reasonable to
ask them for this.

Regards,
_____________
Roman Shpount


On Tue, Jul 23, 2013 at 11:51 AM, Adam Roach <adam@nostrum.com> wrote:

>  On 7/23/13 09:20, tim panton wrote:
>
> On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK <stefan.lk.hakansson@eric=
sson.com> <stefan.lk.hakansson@ericsson.com> wrote:
>
>
>  On 7/23/13 5:03 AM, Justin Uberti wrote:
>
>  With the compromise reached in the Unified Plan document,
>
>  Presumably http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>
>  I'll draw folks attention to this IPR claim
> https://datatracker.ietf.org/ipr/2141/
>
> What does that mean in practice?
>
>
>
> In practice, it has very little effect.
>
> For those of you unfamiliar with IETF IPR policy, it is documented in RFC
> 3979:
>
> http://www.ietf.org/rfc/rfc3979.txt
>
> The disclosure Tim points to is made pursuant to section 6.1.3 of that
> document. During the course of developing the unified plan, the
> applications that are mentioned in that disclosure were brought to my
> attention.
>
> The reason it has little effect in practice is that the independent claim=
s
> would appear to cover every plan proposal to date (plan a, plan b, unifie=
d
> plan, and even all of the "no plan" variations). Consequently, it does no=
t
> benefit any one approach over the others.
>
> /a
>

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Content-Type: text/html; charset=ISO-8859-1
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Changing the title and the mailing list to the more appropriate.<br><br>The=
 situation would be a bit clearer if patent holders were to provide the lic=
ensing policy regarding this IPR release.=A0Given that Ericsson is actively=
 involved in this working group, I think it would be reasonable to ask them=
 for this.<br>
<br>Regards,<br>
<div>_____________<br>Roman Shpount</div>
<br><br><div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 11:51 AM, Adam R=
oach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_b=
lank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">


 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div>
    <div>On 7/23/13 09:20, tim panton wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <pre>On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK <a href=3D"mailt=
o:stefan.lk.hakansson@ericsson.com" target=3D"_blank">&lt;stefan.lk.hakanss=
on@ericsson.com&gt;</a> wrote:

</pre>
      <blockquote type=3D"cite">
        <pre>On 7/23/13 5:03 AM, Justin Uberti wrote:
</pre>
        <blockquote type=3D"cite">
          <pre>With the compromise reached in the Unified Plan document,
</pre>
        </blockquote>
        <pre>Presumably <a href=3D"http://www.ietf.org/id/draft-roach-mmusi=
c-unified-plan-00.txt" target=3D"_blank">http://www.ietf.org/id/draft-roach=
-mmusic-unified-plan-00.txt</a>
</pre>
      </blockquote>
      <pre>I&#39;ll draw folks attention to this IPR claim

<a href=3D"https://datatracker.ietf.org/ipr/2141/" target=3D"_blank">https:=
//datatracker.ietf.org/ipr/2141/</a>

What does that mean in practice?

</pre>
    </blockquote>
    <br></div></div>
    In practice, it has very little effect.<br>
    <br>
    For those of you unfamiliar with IETF IPR policy, it is documented
    in RFC 3979:<br>
    <br>
   =20
    <a href=3D"http://www.ietf.org/rfc/rfc3979.txt" target=3D"_blank">http:=
//www.ietf.org/rfc/rfc3979.txt</a><br>
    <br>
    The disclosure Tim points to is made pursuant to section 6.1.3 of
    that document. During the course of developing the unified plan, the
    applications that are mentioned in that disclosure were brought to
    my attention.<br>
    <br>
    The reason it has little effect in practice is that the independent
    claims would appear to cover every plan proposal to date (plan a,
    plan b, unified plan, and even all of the &quot;no plan&quot; variation=
s).
    Consequently, it does not benefit any one approach over the others.<spa=
n><font color=3D"#888888"><br>
    <br>
    /a<br>
  </font></span></div>

</blockquote></div><br>

--f46d0442808ee7904104e2303130--

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Subject: Re: [rtcweb] Unified plan IPR
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On 7/23/13 11:22, Roman Shpount wrote:
> The situation would be a bit clearer if patent holders were to provide 
> the licensing policy regarding this IPR release. Given that Ericsson 
> is actively involved in this working group, I think it would be 
> reasonable to ask them for this.

If process has been properly followed, the IPR holder has already been 
notified by the IETF executive director. See 
http://www.ietf.org/rfc/rfc4879.txt (section 1 paragraph 1)

I doubt agitating for action on these mailing lists is likely to produce 
useful results.

/a

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--_788a735f-7672-45a0-80b1-0a9af35b1860_
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I do not think it is necessary to "support all forms of scalability for all=
 codecs".   In fact=2C I would make that an explicit "non-goal".  All that =
was suggested is to try to create a single extension that supports a few co=
mmon cases.   If it is possible to handle VP8=2C VP9 and H.264/SVC in a sin=
gle extension that would be sufficient.  So why not limit it to that?=20

Date: Tue=2C 23 Jul 2013 08:53:45 -0700
From: fineberg@vline.me
To: stewe@stewe.org
CC: juberti@google.com=3B bernard_aboba@hotmail.com=3B avtext@ietf.org=3B r=
tcweb@ietf.org
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

=0A=
  =0A=
    =0A=
  =0A=
  =0A=
    I've been thinking about this and given the ease at which RFC5285=0A=
    allows for the specification of a header extension and the=0A=
    complexity introduced by trying to generalize the header extension=0A=
    to support all forms of scalability for all codecs that the=0A=
    generalization might not be the best approach.  I'm not sure what we=0A=
    really gain by trying to capture all this in a single header=0A=
    extension rather than one per that can succinctly explain the fields=0A=
    without the complexity of multiplexing the bits.
=0A=
   =20
=0A=
    Thoughts?
=0A=
   =20
=0A=
    Regards=2C
=0A=
    Adam
=0A=
   =20
=0A=
    On 7/19/13 3:44 PM=2C Stephan Wenger=0A=
      wrote:
=0A=
    =0A=
    =0A=
      =0A=
      Hi=2C=0A=
      =0A=
       =20
=0A=
      =0A=
      =0A=
        =0A=
          From: Adam Fineberg=0A=
          <fineberg@vline.me>
=0A=
          Date: Friday=2C 19 July=2C=0A=
          2013 15:12=20
=0A=
          To: Stephan Wenger <stewe@stewe.org>
=0A=
          Cc: Justin Uberti <juberti@google.com>=2C=0A=
          Bernard Aboba <bernard_aboba@hotmail.com>=2C=0A=
          "avtext@ietf.org"=0A=
          <avtext@ietf.org>=2C=0A=
          "rtcweb@ietf.org"=0A=
          <rtcweb@ietf.org>
=0A=
          Subject: Re: [avtext]=0A=
          [rtcweb] Fwd: New Version Notification for=0A=
          draft-fineberg-avtext-temporal-layer-ext-00.txt
=0A=
        =0A=
       =20
=0A=
        =0A=
        =0A=
          Stephan=2C
=0A=
           =20
=0A=
            Thanks for the info and the reference.  I'm not sure I=0A=
            follow as I'm not at all familiar with H.265.  I'll review=0A=
            the reference and see what I can figure. =0A=
        =0A=
      =0A=
      =0A=
       =20
=0A=
      =0A=
      StW: Good luck :-)  =0A=
      =0A=
       =20
=0A=
      =0A=
      =0A=
        =0A=
          It seems though to me=0A=
            that you are suggesting that except in the simple case=2C that=
=0A=
            the data for H.265 would not be well suited to a header=0A=
            extension=2C am I understanding you correctly?  There is no=0A=
            reason the middlebox couldn't get out of band signaling of=0A=
            the VPS as you mention but that would not be within the=0A=
            scope of this header extension.=0A=
        =0A=
      =0A=
      =0A=
       =20
=0A=
      =0A=
      StW:=0A=
            well=2C if you would copy the layer_id into your header=0A=
            extension (just as you need to do for the simple case)=2C a=0A=
            really smart middle box could use this information just as a=0A=
            decoder uses it=2C assuming that it intercepted the VPS in the=
=0A=
            first place.  Insofar=2C I wouldn't rule out the second option=
=0A=
            on technical grounds.  Whether any of the actual products=0A=
            would bother to do that=2C ever=2C is another question.  I thin=
k=0A=
            the case ought to be documented=2C though.  I can help=0A=
            drafting text.=0A=
      While=0A=
            we are at it: doing this right could mean that you need=0A=
            multiple specs.  First=2C a generic header extension mechanism=
=0A=
            dedicated to side information required for pruning of RTP=0A=
            packet streams=97ideally not only for scalable video=2C althoug=
h=0A=
            that is the main customer today.  And second=2C for each=0A=
            "payload" (at present we are talking about H.264/SVC=2C=0A=
            H.265v1 (HEVC)=2C H.265v2 (including scalable and 3D=0A=
            extensions=2C which are not yet finalized)=2C VP8=2C and VP9 (I=
=0A=
            know little about the latter)=2C plus Daala=2C and whatnot) a=
=0A=
            mapping of the bits available in the generic header=0A=
            extension to the bits in the payload itself (NAL header and=0A=
            VPS in case of H.265=2C NALU header in SVC=2C and the fields yo=
u=0A=
            mention for VP8).=0A=
     =20
=0A=
      =0A=
      Stephan=0A=
      =0A=
        =0A=
         =20
=0A=
           =20
=0A=
            Any insights are appreciated.
=0A=
           =20
=0A=
            Regards=2C
=0A=
            Adam
=0A=
           =20
=0A=
            On 7/19/13 8:33 AM=2C Stephan=0A=
              Wenger wrote:
=0A=
            =0A=
            =0A=
              Hi=2C=0A=
              I also believe that 16 bits should be enough.  For=0A=
                H.264 and VP8 that has already been demonstrated.  For=0A=
                H.265=2C some initial thoughts below.  Apologies for the=0A=
                word-count.=0A=
             =20
=0A=
              =0A=
              The scalable version of H265 (called SHVC) is=0A=
                currently under development.  The current working draft=0A=
                can be found here: http://phenix.int-evry.fr/jct/doc_end_us=
er/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.=0A=
                 Therein=2C the options for defining layering structures=0A=
                are considerably more complex.  To start=2C we have 3 bits=
=0A=
                for the temporal ID in the NAL unit header of the H.265=0A=
                version 1 (HEVC) base specification (temporal=0A=
                scalability is already nicely supported in version 1).=0A=
                 Just like in SVC.  In the scalable extension=2C the NAL=0A=
                unit header contains a six bit field that points into a=0A=
                data structure known as "Video Parameter Set" (VPS).=0A=
                 Inside the VPS=2C those six bits are mapped to to a=0A=
                position in a directed graph (specified through=0A=
                "dimension_id[][]")=2C which tells you about the reference=
=0A=
                relationship of the layer in question and its parent=0A=
                layer.  One can recursively follow the graph to=0A=
                determine what used to be called dependency_id=2C=0A=
                quality_id=2C view_id=2C and whatnot.  The six bit pointer=
=0A=
                field can (or: is to be when possible) organized by the=0A=
                encoder such that it is prudent for a middle box to=0A=
                throw away NAL units (belonging to layers) with higher=0A=
                values of the six bit field first=2C before throwing away=
=0A=
                NAL units with lower values.  Relying on this feature=2C=0A=
                3+6 bits =3D=3D 9 bits should be fine for the header=0A=
                extension.=0A=
             =20
=0A=
              =0A=
              That said=2C the ordering by the encoder is just a=0A=
                recommendation=2C and there may well be cases where=0A=
                different pruning strategies may be advisable.  For=0A=
                example=2C a layering structure could be constructed that=
=0A=
                expands into two branches=2C one using 2D scalable tools=0A=
                only=2C the other including view_id for multi view coding.=
=0A=
                 By looking at the six bit field alone=2C a middle box=0A=
                will not be able to meaningfully remove NAL units=0A=
                belonging to one of the branches completely while=0A=
                pruning the other branch.  In order to meaningfully deal=0A=
                with that scenario=2C there would be two options: one to=0A=
                represent the dimension_id[][] (and associated control=0A=
                info) in the header extension=2C or require the middle box=
=0A=
                to have access to the VPS and be able to interpret its=0A=
                content.  The further could take considerably more than=0A=
                16 bits and we would be talking about a variable length=0A=
                data structure.  The latter requires the middle box to=0A=
                have state and a mechanism to intercept the VPS (through=0A=
                signaling=97as the encrypted in-band VPS would not be=0A=
                useful under the assumption that the middle box does not=0A=
                have the key to the media=97which is the motivation of the=
=0A=
                draft in the first place).  I personally don't mind at=0A=
                all the second mechanism=2C as I'm a big fan of=0A=
                out-of-band parameter set transmission and any middle=0A=
                box must be in the signaling path anyway to meaningfully=0A=
                manipulate RTP.  I do not like the first option due to=0A=
                its variable=2C and possibly substantial=2C overhead.=0A=
             =20
=0A=
              =0A=
              Stephan   =0A=
             =20
=0A=
              =0A=
             =20
=0A=
              =0A=
              =0A=
                =0A=
                  From: Justin=0A=
                  Uberti <juberti@google.com>
=0A=
                  Date: Friday=2C=0A=
                  19 July=2C 2013 06:32=20
=0A=
                  To: Bernard=0A=
                  Aboba <bernard_aboba@hotmail.com>
=0A=
                  Cc: "avtext@ietf.org"=0A=
                  <avtext@ietf.org>=2C=0A=
                  "rtcweb@ietf.org"=0A=
                  <rtcweb@ietf.org>
=0A=
                  Subject: Re:=0A=
                  [rtcweb] Fwd: New Version Notification for=0A=
                  draft-fineberg-avtext-temporal-layer-ext-00.txt
=0A=
                =0A=
               =20
=0A=
                =0A=
                =0A=
                  =0A=
                    Agree those are the right codecs to=0A=
                      design for. Since in each case there are fairly=0A=
                      low limits on the number of supported layers (i.e.=0A=
                      3 spatial layers for SVC)=2C I think it should be=0A=
                      possible to pack the temporal=2C spatial=2C quality=
=0A=
                      layer ids into 16 bits.=0A=
                     =20
=0A=
                       =20
=0A=
                        On Fri=2C Jul 19=2C 2013 at=0A=
                          1:56 AM=2C Bernard Aboba =0A=
                            <bernard_aboba@hotmail.com>=0A=
                          wrote:
=0A=
                          =0A=
                            =0A=
                              If we can support VP8/9 as well as=0A=
                                H.264/5 SVC=0A=
                              that would be a start. It seems=0A=
                                doable to me.=0A=
                              =0A=
                                =0A=
                                 =20
=0A=
                                    On Jul 18=2C 2013=2C at 8:34 PM=2C "Ada=
m=0A=
                                    Fineberg" <fineberg@vline.me>=0A=
                                    wrote:
=0A=
                                   =20
=0A=
                                  =0A=
                                  =0A=
                                    =0A=
                                      Bernard=2C
=0A=
                                       =20
=0A=
                                        Are there other codecs you are=0A=
                                        thinking should be supported? =0A=
                                        If it's generalized I would=0A=
                                        think we want to be able to=0A=
                                        cover all known scalable codecs.=0A=
                                        I'll look into the H264/SVC=0A=
                                        fields to see how to encode them=0A=
                                        in a generalized header.
=0A=
                                       =20
=0A=
                                        Regards=2C
=0A=
                                        Adam
=0A=
                                       =20
=0A=
                                        On 7/18/13 7:40 PM=2C Bernard=0A=
                                        Aboba wrote:
=0A=
                                      =0A=
                                      =0A=
                                        I think it may be=0A=
                                          possible to generalize this. =0A=
                                          For example=2C for H.264/SVC=0A=
                                          which can support temporal=2C=0A=
                                          spatial and quality=0A=
                                          scalability=2C you would need=0A=
                                          the quality_id and=0A=
                                          dependency_id in addition to=0A=
                                          the temporal_id (what you call=0A=
                                          the temporal layer index).   =0A=
                                         =20
=0A=
                                         =20
=0A=
                                          =0A=
                                            =0A=
                                            Date: Thu=2C 18 Jul 2013=0A=
                                            08:45:38 -0700
=0A=
                                            From: fineberg@vline.me
=0A=
                                            To: =0A=
                                              bernard_aboba@hotmail.com
=0A=
                                            CC: rtcweb@ietf.org=3B=0A=
                                            avtext@ietf.org
=0A=
                                            Subject: Re: [rtcweb] Fwd:=0A=
                                            New Version Notification for=0A=
draft-fineberg-avtext-temporal-layer-ext-00.txt
=0A=
                                           =20
=0A=
                                            Bernard=2C
=0A=
                                           =20
=0A=
                                            Good question.  I'm not=0A=
                                            familiar enough with the=0A=
                                            parameter requirements of=0A=
                                            all other scalable codecs to=0A=
                                            be able to generalize.  If=0A=
                                            you'd like to help specify=0A=
                                            them=2C I'd be fine revising=0A=
                                            the draft to generalize.
=0A=
                                           =20
=0A=
                                            Regards=2C
=0A=
                                            Adam
=0A=
                                           =20
=0A=
                                            On 7/17/13 8:26 PM=2C=0A=
                                              Bernard Aboba wrote:
=0A=
                                            =0A=
                                            =0A=
                                              Since the=0A=
                                                need is not codec=0A=
                                                specific (e.g. it arises=0A=
                                                with any codec=0A=
                                                supporting temporal=2C=0A=
                                                spatial and quality=0A=
                                                scalability)=2C why=0A=
                                               =20
=0A=
                                                 a VP8-specific RTP=0A=
                                                extension?=20
=0A=
                                                =20
=0A=
                                                =0A=
                                                  =0A=
                                                  Date: Wed=2C 17 Jul 2013=
=0A=
                                                  17:09:46 -0700
=0A=
                                                  From: fineberg@vline.me
=0A=
                                                  To: rtcweb@ietf.org
=0A=
                                                  Subject: [rtcweb] Fwd:=0A=
                                                  New Version=0A=
                                                  Notification for=0A=
                                                  draft-fineberg-avtext-tem=
poral-layer-ext-00.txt
=0A=
                                                 =20
=0A=
                                                  Hi=2C=0A=
                                                  =0A=
                                                  I'm=0A=
                                                    working on WebRTC=0A=
                                                    services and have=0A=
                                                    found that while=0A=
                                                    developing services=0A=
                                                    that forward VP8=0A=
                                                    video streams if we=0A=
                                                    want to take=0A=
                                                    advantage of the VP8=0A=
                                                    temporal scaling we=0A=
                                                    must get the=0A=
                                                    temporal layer=0A=
                                                    information from the=0A=
                                                    RTP header which=0A=
                                                    requires us to=0A=
                                                    decrypt the SRTP=0A=
                                                    packets. This is=0A=
                                                    undesirable both=0A=
                                                    because the=0A=
                                                    middle-box needs to=0A=
                                                    have access to the=0A=
                                                    keys as well as the=0A=
                                                    because of the added=0A=
                                                    overhead of the=0A=
                                                    decrypt/encrypt=0A=
                                                    cycle. This draft=0A=
                                                    proposes an RTP=0A=
                                                    header extension=0A=
                                                    that will allow us=0A=
                                                    to use the VP8=0A=
                                                    temporal layer=0A=
                                                    information included=0A=
                                                    in the header=0A=
                                                    extension and=0A=
                                                    therefore do=0A=
                                                    forwarding without=0A=
                                                    SRTP decryption.=0A=
                                                    Comments welcome.=0A=
                                                  =0A=
                                                  Regards=2C=0A=
                                                  Adam=0A=
                                                    Fineberg=0A=
                                                  fineberg=0A=
                                                      at vline.com
=0A=
                                                   =20
=0A=
                                                    -------- Original=0A=
                                                    Message --------=0A=
                                                    =0A=
                                                      =0A=
                                                        =0A=
                                                          Subject:=0A=
                                                          New=0A=
                                                          Version=0A=
                                                          Notification=0A=
                                                          for=0A=
                                                          draft-fineberg-av=
text-temporal-layer-ext-00.txt=0A=
                                                        =0A=
                                                        =0A=
                                                          Date:=0A=
                                                          Tue=2C 09=0A=
                                                          Jul 2013=0A=
                                                          10:02:05 -0700=0A=
                                                        =0A=
                                                        =0A=
                                                          From:=0A=
                                                          internet-drafts=
=0A=
                                                          at ietf.org=0A=
                                                        =0A=
                                                        =0A=
                                                          To:=0A=
                                                          Adam=0A=
                                                          Fineberg <fineber=
g at vline.com>=0A=
                                                        =0A=
                                                      =0A=
                                                    =0A=
                                                   =20
=0A=
                                                   =20
=0A=
                                                    A new version of I-D=2C=
 draft-fineberg-avtext-temporal-layer-ext-00.txt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt=0A=
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext=0A=
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
=0A=
                                                  =0A=
                                                 =20
=0A=
                                                  _________________________=
______________________=0A=
                                                  rtcweb mailing list =0A=
                                                    rtcweb@ietf.org=0A=
                                                  =0A=
https://www.ietf.org/mailman/listinfo/rtcweb=0A=
                                              =0A=
                                            =0A=
                                           =20
=0A=
                                            -- =0A=
Regards=2C=0A=
Adam=0A=
                                          =0A=
                                        =0A=
                                      =0A=
                                     =20
=0A=
                                    =0A=
                                  =0A=
                                =0A=
                              =0A=
                            =0A=
                           =20
=0A=
_______________________________________________
=0A=
                            rtcweb mailing list
=0A=
                            rtcweb@ietf.org
=0A=
                            https://www.ietf.org/mailman/listinfo/rtcweb
=0A=
                           =20
=0A=
                          =0A=
                        =0A=
                       =20
=0A=
                      =0A=
                    =0A=
                  =0A=
                =0A=
             =20
=0A=
              =0A=
             =20
=0A=
              _______________________________________________=0A=
avtext mailing list=0A=
avtext@ietf.orghttps://www.ietf.org/mailman/listinfo/avtext=0A=
            =0A=
           =20
=0A=
            -- =0A=
Regards=2C=0A=
Adam=0A=
          =0A=
        =0A=
      =0A=
    =0A=
   =20
=0A=
    -- =0A=
Regards=2C=0A=
Adam 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>I do not think it is necessary t=
o "support all forms of scalability for all codecs".&nbsp=3B&nbsp=3B In fac=
t=2C I would make that an explicit "non-goal".&nbsp=3B All that was suggest=
ed is to try to create a single extension that supports a few common cases.=
&nbsp=3B&nbsp=3B If it is possible to handle VP8=2C VP9 and H.264/SVC in a =
single extension that would be sufficient.&nbsp=3B So why not limit it to t=
hat? <br><br><div><hr id=3D"stopSpelling">Date: Tue=2C 23 Jul 2013 08:53:45=
 -0700<br>From: fineberg@vline.me<br>To: stewe@stewe.org<br>CC: juberti@goo=
gle.com=3B bernard_aboba@hotmail.com=3B avtext@ietf.org=3B rtcweb@ietf.org<=
br>Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-f=
ineberg-avtext-temporal-layer-ext-00.txt<br><br>=0A=
  =0A=
    =0A=
  =0A=
  =0A=
    I've been thinking about this and given the ease at which RFC5285=0A=
    allows for the specification of a header extension and the=0A=
    complexity introduced by trying to generalize the header extension=0A=
    to support all forms of scalability for all codecs that the=0A=
    generalization might not be the best approach.&nbsp=3B I'm not sure wha=
t we=0A=
    really gain by trying to capture all this in a single header=0A=
    extension rather than one per that can succinctly explain the fields=0A=
    without the complexity of multiplexing the bits.<br>=0A=
    <br>=0A=
    Thoughts?<br>=0A=
    <br>=0A=
    Regards=2C<br>=0A=
    Adam<br>=0A=
    <br>=0A=
    <div class=3D"ecxmoz-cite-prefix">On 7/19/13 3:44 PM=2C Stephan Wenger=
=0A=
      wrote:<br>=0A=
    </div>=0A=
    <blockquote cite=3D"mid:CE0F0BEB.9F95D%25stewe@stewe.org">=0A=
      =0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3B">=
<font color=3D"#0000ff">Hi=2C</font></div>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3Bco=
lor:rgb(0=2C 0=2C 0)=3B">=0A=
        <br>=0A=
      </div>=0A=
      <span id=3D"ecxOLK_SRC_BODY_SECTION" style=3D"font-family:Calibri=2C =
sans-serif=3Bfont-size:14px=3Bcolor:rgb(0=2C 0=2C 0)=3B">=0A=
        <div style=3D"font-family:Calibri=3Bfont-size:11pt=3Btext-align:lef=
t=3Bcolor:black=3BBORDER-BOTTOM:medium none=3BBORDER-LEFT:medium none=3BPAD=
DING-BOTTOM:0in=3BPADDING-LEFT:0in=3BPADDING-RIGHT:0in=3BBORDER-TOP:#b5c4df=
 1pt solid=3BBORDER-RIGHT:medium none=3BPADDING-TOP:3pt=3B">=0A=
          <span style=3D"font-weight:bold=3B">From: </span>Adam Fineberg=0A=
          &lt=3B<a href=3D"mailto:fineberg@vline.me">fineberg@vline.me</a>&=
gt=3B<br>=0A=
          <span style=3D"font-weight:bold=3B">Date: </span>Friday=2C 19 Jul=
y=2C=0A=
          2013 15:12 <br>=0A=
          <span style=3D"font-weight:bold=3B">To: </span>Stephan Wenger &lt=
=3B<a href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a>&gt=3B<br>=0A=
          <span style=3D"font-weight:bold=3B">Cc: </span>Justin Uberti &lt=
=3B<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt=3B=2C=0A=
          Bernard Aboba &lt=3B<a href=3D"mailto:bernard_aboba@hotmail.com">=
bernard_aboba@hotmail.com</a>&gt=3B=2C=0A=
          "<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>"=0A=
          &lt=3B<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt=
=3B=2C=0A=
          "<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"=0A=
          &lt=3B<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt=
=3B<br>=0A=
          <span style=3D"font-weight:bold=3B">Subject: </span>Re: [avtext]=
=0A=
          [rtcweb] Fwd: New Version Notification for=0A=
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>=0A=
        </div>=0A=
        <div><br>=0A=
        </div>=0A=
        <div>=0A=
          <div>Stephan=2C<br>=0A=
            <br>=0A=
            Thanks for the info and the reference.&nbsp=3B I'm not sure I=
=0A=
            follow as I'm not at all familiar with H.265.&nbsp=3B I'll revi=
ew=0A=
            the reference and see what I can figure.&nbsp=3B</div>=0A=
        </div>=0A=
      </span>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3Bco=
lor:rgb(0=2C 0=2C 0)=3B">=0A=
        <br>=0A=
      </div>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3B">=
<font color=3D"#0000ff">StW: Good luck :-) &nbsp=3B</font></div>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3Bco=
lor:rgb(0=2C 0=2C 0)=3B">=0A=
        <br>=0A=
      </div>=0A=
      <span id=3D"ecxOLK_SRC_BODY_SECTION" style=3D"font-family:Calibri=2C =
sans-serif=3Bfont-size:14px=3Bcolor:rgb(0=2C 0=2C 0)=3B">=0A=
        <div>=0A=
          <div>It seems though to me=0A=
            that you are suggesting that except in the simple case=2C that=
=0A=
            the data for H.265 would not be well suited to a header=0A=
            extension=2C am I understanding you correctly?&nbsp=3B There is=
 no=0A=
            reason the middlebox couldn't get out of band signaling of=0A=
            the VPS as you mention but that would not be within the=0A=
            scope of this header extension.</div>=0A=
        </div>=0A=
      </span>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3Bco=
lor:rgb(0=2C 0=2C 0)=3B">=0A=
        <br>=0A=
      </div>=0A=
      <div><font color=3D"#0000ff"><font face=3D"Calibri=2Csans-serif">StW:=
=0A=
            well=2C if you would copy the layer_id into your header=0A=
            extension (just as you need to do for the simple case)=2C a=0A=
            really smart middle box could use this information just as a=0A=
            decoder uses it=2C&nbsp=3Bassuming&nbsp=3Bthat it intercepted t=
he VPS in the=0A=
            first place. &nbsp=3BInsofar=2C I wouldn't rule out the second =
option=0A=
            on technical grounds. &nbsp=3BWhether any of the actual product=
s=0A=
            would bother to do that=2C ever=2C is another question. &nbsp=
=3BI think=0A=
            the case ought to be documented=2C though. &nbsp=3BI can help=
=0A=
            drafting text.</font></font></div>=0A=
      <div><font color=3D"#0000ff"><font face=3D"Calibri=2Csans-serif">Whil=
e=0A=
            we are at it: doing this right could mean that you need=0A=
            multiple specs. &nbsp=3BFirst=2C a generic header extension mec=
hanism=0A=
            dedicated to side information required for pruning of RTP=0A=
            packet streams=97ideally not only for scalable video=2C althoug=
h=0A=
            that is the main customer today. &nbsp=3BAnd second=2C for each=
=0A=
            "payload" (at present we are talking about H.264/SVC=2C=0A=
            H.265v1 (HEVC)=2C H.265v2 (including scalable and 3D=0A=
            extensions=2C which are not yet finalized)=2C VP8=2C and VP9 (I=
=0A=
            know little about the latter)=2C plus Daala=2C and whatnot) a=
=0A=
            mapping of the bits available in the generic header=0A=
            extension to the bits in the payload itself (NAL header and=0A=
            VPS in case of H.265=2C NALU header in SVC=2C and the fields yo=
u=0A=
            mention for VP8).</font></font></div>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3B">=
<br>=0A=
      </div>=0A=
      <div style=3D"font-family:Calibri=2C sans-serif=3Bfont-size:14px=3B">=
<font color=3D"#0000ff">Stephan</font></div>=0A=
      <span id=3D"ecxOLK_SRC_BODY_SECTION" style=3D"font-family:Calibri=2C =
sans-serif=3Bfont-size:14px=3Bcolor:rgb(0=2C 0=2C 0)=3B">=0A=
        <div>=0A=
          <div><br>=0A=
            <br>=0A=
            Any insights are appreciated.<br>=0A=
            <br>=0A=
            Regards=2C<br>=0A=
            Adam<br>=0A=
            <br>=0A=
            <div class=3D"ecxmoz-cite-prefix">On 7/19/13 8:33 AM=2C Stephan=
=0A=
              Wenger wrote:<br>=0A=
            </div>=0A=
            <blockquote cite=3D"mid:CE0E9F56.9F89B%25stewe@stewe.org">=0A=
              <div>Hi=2C</div>=0A=
              <div>I also believe that 16 bits should be enough. &nbsp=3BFo=
r=0A=
                H.264 and VP8 that has already been demonstrated. &nbsp=3BF=
or=0A=
                H.265=2C some initial thoughts below. &nbsp=3BApologies for=
 the=0A=
                word-count.</div>=0A=
              <div><br>=0A=
              </div>=0A=
              <div>The scalable version of H265 (called SHVC) is=0A=
                currently under development. &nbsp=3BThe current working dr=
aft=0A=
                can be found here:&nbsp=3B<a href=3D"http://phenix.int-evry=
.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip" target=
=3D"_blank">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon=
/wg11/JCTVC-M1008-v3.zip</a>.=0A=
                &nbsp=3BTherein=2C the options for defining layering struct=
ures=0A=
                are considerably more complex. &nbsp=3BTo start=2C we have =
3 bits=0A=
                for the temporal ID in the NAL unit header of the H.265=0A=
                version 1 (HEVC) base specification (temporal=0A=
                scalability is already nicely supported in version 1).=0A=
                &nbsp=3BJust like in SVC. &nbsp=3BIn the scalable extension=
=2C the NAL=0A=
                unit header contains a six bit field that points into a=0A=
                data structure known as "Video Parameter Set" (VPS).=0A=
                &nbsp=3BInside the VPS=2C those six bits are mapped to to a=
=0A=
                position in a directed graph (specified through=0A=
                "dimension_id[][]")=2C which tells you about the reference=
=0A=
                relationship of the layer in question and its parent=0A=
                layer. &nbsp=3BOne can recursively follow the graph to=0A=
                determine what used to be called dependency_id=2C=0A=
                quality_id=2C view_id=2C and whatnot. &nbsp=3BThe six bit p=
ointer=0A=
                field can (or: is to be when possible) organized by the=0A=
                encoder such that it is prudent for a middle box to=0A=
                throw away NAL units (belonging to layers) with higher=0A=
                values of the six bit field first=2C before throwing away=
=0A=
                NAL units with lower values. &nbsp=3BRelying on this featur=
e=2C=0A=
                3+6 bits =3D=3D 9 bits should be fine for the header=0A=
                extension.</div>=0A=
              <div><br>=0A=
              </div>=0A=
              <div>That said=2C the ordering by the encoder is just a=0A=
                recommendation=2C and there may well be cases where=0A=
                different pruning strategies may be advisable. &nbsp=3BFor=
=0A=
                example=2C a layering structure could be constructed that=
=0A=
                expands into two branches=2C one using 2D scalable tools=0A=
                only=2C the other including view_id for multi view coding.=
=0A=
                &nbsp=3BBy looking at the six bit field alone=2C a middle b=
ox=0A=
                will not be able to meaningfully remove NAL units=0A=
                belonging to one of the branches completely while=0A=
                pruning the other branch. &nbsp=3BIn order to meaningfully =
deal=0A=
                with that scenario=2C there would be two options: one to=0A=
                represent the dimension_id[][] (and associated control=0A=
                info) in the header extension=2C or require the middle box=
=0A=
                to have access to the VPS and be able to interpret its=0A=
                content. &nbsp=3BThe further could take considerably more t=
han=0A=
                16 bits and we would be talking about a variable length=0A=
                data structure. &nbsp=3BThe latter requires the middle box =
to=0A=
                have state and a mechanism to intercept the VPS (through=0A=
                signaling=97as the encrypted in-band VPS would not be=0A=
                useful under the assumption that the middle box does not=0A=
                have the key to the media=97which is the motivation of the=
=0A=
                draft in the first place). &nbsp=3BI personally don't mind =
at=0A=
                all the second mechanism=2C as I'm a big fan of=0A=
                out-of-band parameter set transmission and any middle=0A=
                box must be in the signaling path anyway to meaningfully=0A=
                manipulate RTP. &nbsp=3BI do not like the first option due =
to=0A=
                its variable=2C and possibly substantial=2C overhead.</div>=
=0A=
              <div><br>=0A=
              </div>=0A=
              <div>Stephan &nbsp=3B&nbsp=3B</div>=0A=
              <div><br>=0A=
              </div>=0A=
              <div><br>=0A=
              </div>=0A=
              <span id=3D"ecxOLK_SRC_BODY_SECTION">=0A=
                <div style=3D"font-family:Calibri=3Bfont-size:11pt=3Btext-a=
lign:left=3Bcolor:black=3BBORDER-BOTTOM:medium none=3BBORDER-LEFT:medium no=
ne=3BPADDING-BOTTOM:0in=3BPADDING-LEFT:0in=3BPADDING-RIGHT:0in=3BBORDER-TOP=
:#b5c4df 1pt solid=3BBORDER-RIGHT:medium none=3BPADDING-TOP:3pt=3B">=0A=
                  <span style=3D"font-weight:bold=3B">From: </span>Justin=
=0A=
                  Uberti &lt=3B<a href=3D"mailto:juberti@google.com">jubert=
i@google.com</a>&gt=3B<br>=0A=
                  <span style=3D"font-weight:bold=3B">Date: </span>Friday=
=2C=0A=
                  19 July=2C 2013 06:32 <br>=0A=
                  <span style=3D"font-weight:bold=3B">To: </span>Bernard=0A=
                  Aboba &lt=3B<a href=3D"mailto:bernard_aboba@hotmail.com">=
bernard_aboba@hotmail.com</a>&gt=3B<br>=0A=
                  <span style=3D"font-weight:bold=3B">Cc: </span>"<a href=
=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>"=0A=
                  &lt=3B<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org<=
/a>&gt=3B=2C=0A=
                  "<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"=
=0A=
                  &lt=3B<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org<=
/a>&gt=3B<br>=0A=
                  <span style=3D"font-weight:bold=3B">Subject: </span>Re:=
=0A=
                  [rtcweb] Fwd: New Version Notification for=0A=
                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>=0A=
                </div>=0A=
                <div><br>=0A=
                </div>=0A=
                <div>=0A=
                  <div>=0A=
                    <div dir=3D"ltr">Agree those are the right codecs to=0A=
                      design for. Since in each case there are fairly=0A=
                      low limits on the number of supported layers (i.e.=0A=
                      3 spatial layers for SVC)=2C I think it should be=0A=
                      possible to pack the temporal=2C spatial=2C quality=
=0A=
                      layer ids into 16 bits.=0A=
                      <div class=3D"ecxgmail_extra"><br>=0A=
                        <br>=0A=
                        <div class=3D"ecxgmail_quote">On Fri=2C Jul 19=2C 2=
013 at=0A=
                          1:56 AM=2C Bernard Aboba <span dir=3D"ltr">=0A=
                            &lt=3B<a href=3D"mailto:bernard_aboba@hotmail.c=
om" target=3D"_blank">bernard_aboba@hotmail.com</a>&gt=3B</span>=0A=
                          wrote:<br>=0A=
                          <blockquote class=3D"ecxgmail_quote" style=3D"bor=
der-left:1px #ccc solid=3Bpadding-left:1ex=3B">=0A=
                            <div dir=3D"auto">=0A=
                              <div>If we can support VP8/9 as well as=0A=
                                H.264/5 SVC</div>=0A=
                              <div>that would be a start. It seems=0A=
                                doable to me.</div>=0A=
                              <div>=0A=
                                <div>=0A=
                                  <div><br>=0A=
                                    On Jul 18=2C 2013=2C at 8:34 PM=2C "Ada=
m=0A=
                                    Fineberg" &lt=3B<a href=3D"mailto:fineb=
erg@vline.me" target=3D"_blank">fineberg@vline.me</a>&gt=3B=0A=
                                    wrote:<br>=0A=
                                    <br>=0A=
                                  </div>=0A=
                                  <blockquote>=0A=
                                    <div>=0A=
                                      <div>Bernard=2C<br>=0A=
                                        <br>=0A=
                                        Are there other codecs you are=0A=
                                        thinking should be supported?&nbsp=
=3B=0A=
                                        If it's generalized I would=0A=
                                        think we want to be able to=0A=
                                        cover all known scalable codecs.=0A=
                                        I'll look into the H264/SVC=0A=
                                        fields to see how to encode them=0A=
                                        in a generalized header.<br>=0A=
                                        <br>=0A=
                                        Regards=2C<br>=0A=
                                        Adam<br>=0A=
                                        <br>=0A=
                                        On 7/18/13 7:40 PM=2C Bernard=0A=
                                        Aboba wrote:<br>=0A=
                                      </div>=0A=
                                      <blockquote>=0A=
                                        <div dir=3D"ltr">I think it may be=
=0A=
                                          possible to generalize this.&nbsp=
=3B=0A=
                                          For example=2C for H.264/SVC=0A=
                                          which can support temporal=2C=0A=
                                          spatial and quality=0A=
                                          scalability=2C you would need=0A=
                                          the quality_id and=0A=
                                          dependency_id in addition to=0A=
                                          the temporal_id (what you call=0A=
                                          the temporal layer index). &nbsp=
=3B&nbsp=3B=0A=
                                          <br>=0A=
                                          <br>=0A=
                                          <div>=0A=
                                            <hr>=0A=
                                            Date: Thu=2C 18 Jul 2013=0A=
                                            08:45:38 -0700<br>=0A=
                                            From: <a href=3D"mailto:fineber=
g@vline.me" target=3D"_blank">fineberg@vline.me</a><br>=0A=
                                            To: <a href=3D"mailto:bernard_a=
boba@hotmail.com" target=3D"_blank">=0A=
                                              bernard_aboba@hotmail.com</a>=
<br>=0A=
                                            CC: <a href=3D"mailto:rtcweb@ie=
tf.org" target=3D"_blank">rtcweb@ietf.org</a>=3B=0A=
                                            <a href=3D"mailto:avtext@ietf.o=
rg" target=3D"_blank">avtext@ietf.org</a><br>=0A=
                                            Subject: Re: [rtcweb] Fwd:=0A=
                                            New Version Notification for=0A=
draft-fineberg-avtext-temporal-layer-ext-00.txt<br>=0A=
                                            <br>=0A=
                                            Bernard=2C<br>=0A=
                                            <br>=0A=
                                            Good question.&nbsp=3B I'm not=
=0A=
                                            familiar enough with the=0A=
                                            parameter requirements of=0A=
                                            all other scalable codecs to=0A=
                                            be able to generalize.&nbsp=3B =
If=0A=
                                            you'd like to help specify=0A=
                                            them=2C I'd be fine revising=0A=
                                            the draft to generalize.<br>=0A=
                                            <br>=0A=
                                            Regards=2C<br>=0A=
                                            Adam<br>=0A=
                                            <br>=0A=
                                            <div>On 7/17/13 8:26 PM=2C=0A=
                                              Bernard Aboba wrote:<br>=0A=
                                            </div>=0A=
                                            <blockquote>=0A=
                                              <div dir=3D"ltr">Since the=0A=
                                                need is not codec=0A=
                                                specific (e.g. it arises=0A=
                                                with any codec=0A=
                                                supporting temporal=2C=0A=
                                                spatial and quality=0A=
                                                scalability)=2C why=0A=
                                                <br>=0A=
                                                &nbsp=3Ba VP8-specific RTP=
=0A=
                                                extension? <br>=0A=
                                                &nbsp=3B<br>=0A=
                                                <div>=0A=
                                                  <hr>=0A=
                                                  Date: Wed=2C 17 Jul 2013=
=0A=
                                                  17:09:46 -0700<br>=0A=
                                                  From: <a href=3D"mailto:f=
ineberg@vline.me" target=3D"_blank">fineberg@vline.me</a><br>=0A=
                                                  To: <a href=3D"mailto:rtc=
web@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=0A=
                                                  Subject: [rtcweb] Fwd:=0A=
                                                  New Version=0A=
                                                  Notification for=0A=
                                                  draft-fineberg-avtext-tem=
poral-layer-ext-00.txt<br>=0A=
                                                  <br>=0A=
                                                  <span style=3D"text-inden=
t:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3B=
display:inline !important=3Bword-spacing:0px=3B">Hi=2C</span><br style=3D"t=
ext-indent:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:=
normal=3Bword-spacing:0px=3B">=0A=
                                                  <br style=3D"text-indent:=
0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3Bwo=
rd-spacing:0px=3B">=0A=
                                                  <span style=3D"text-inden=
t:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3B=
display:inline !important=3Bword-spacing:0px=3B">I'm=0A=
                                                    working on WebRTC=0A=
                                                    services and have=0A=
                                                    found that while=0A=
                                                    developing services=0A=
                                                    that forward VP8=0A=
                                                    video streams if we=0A=
                                                    want to take=0A=
                                                    advantage of the VP8=0A=
                                                    temporal scaling we=0A=
                                                    must get the=0A=
                                                    temporal layer=0A=
                                                    information from the=0A=
                                                    RTP header which=0A=
                                                    requires us to=0A=
                                                    decrypt the SRTP=0A=
                                                    packets. This is=0A=
                                                    undesirable both=0A=
                                                    because the=0A=
                                                    middle-box needs to=0A=
                                                    have access to the=0A=
                                                    keys as well as the=0A=
                                                    because of the added=0A=
                                                    overhead of the=0A=
                                                    decrypt/encrypt=0A=
                                                    cycle. This draft=0A=
                                                    proposes an RTP=0A=
                                                    header extension=0A=
                                                    that will allow us=0A=
                                                    to use the VP8=0A=
                                                    temporal layer=0A=
                                                    information included=0A=
                                                    in the header=0A=
                                                    extension and=0A=
                                                    therefore do=0A=
                                                    forwarding without=0A=
                                                    SRTP decryption.=0A=
                                                    Comments welcome.</span=
><br style=3D"text-indent:0px=3Bletter-spacing:normal=3Btext-transform:none=
=3Bwhite-space:normal=3Bword-spacing:0px=3B">=0A=
                                                  <br style=3D"text-indent:=
0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3Bwo=
rd-spacing:0px=3B">=0A=
                                                  <span style=3D"text-inden=
t:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3B=
display:inline !important=3Bword-spacing:0px=3B">Regards=2C</span><br style=
=3D"text-indent:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-s=
pace:normal=3Bword-spacing:0px=3B">=0A=
                                                  <span style=3D"text-inden=
t:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3B=
display:inline !important=3Bword-spacing:0px=3B">Adam=0A=
                                                    Fineberg</span><br styl=
e=3D"text-indent:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-=
space:normal=3Bword-spacing:0px=3B">=0A=
                                                  <div style=3D"text-indent=
:0px=3Bletter-spacing:normal=3Btext-transform:none=3Bwhite-space:normal=3Bw=
ord-spacing:0px=3B"><a href=3D"mailto:fineberg%20at%20vline.com" rel=3D"nof=
ollow" target=3D"_blank">fineberg=0A=
                                                      at vline.com</a><br>=
=0A=
                                                    <br>=0A=
                                                    -------- Original=0A=
                                                    Message --------=0A=
                                                    <table border=3D"0" cel=
lpadding=3D"0" cellspacing=3D"0">=0A=
                                                      <tbody>=0A=
                                                        <tr>=0A=
                                                          <th align=3D"RIGH=
T" nowrap=3D"nowrap" valign=3D"BASELINE">Subject:</th>=0A=
                                                          <td>New=0A=
                                                          Version=0A=
                                                          Notification=0A=
                                                          for=0A=
                                                          draft-fineberg-av=
text-temporal-layer-ext-00.txt</td>=0A=
                                                        </tr>=0A=
                                                        <tr>=0A=
                                                          <th align=3D"RIGH=
T" nowrap=3D"nowrap" valign=3D"BASELINE">Date:</th>=0A=
                                                          <td>Tue=2C 09=0A=
                                                          Jul 2013=0A=
                                                          10:02:05 -0700</t=
d>=0A=
                                                        </tr>=0A=
                                                        <tr>=0A=
                                                          <th align=3D"RIGH=
T" nowrap=3D"nowrap" valign=3D"BASELINE">From:</th>=0A=
                                                          <td><a href=3D"ma=
ilto:internet-drafts%20at%20ietf.org" rel=3D"nofollow" target=3D"_blank">in=
ternet-drafts=0A=
                                                          at ietf.org</a></=
td>=0A=
                                                        </tr>=0A=
                                                        <tr>=0A=
                                                          <th align=3D"RIGH=
T" nowrap=3D"nowrap" valign=3D"BASELINE">To:</th>=0A=
                                                          <td>Adam=0A=
                                                          Fineberg<span>&nb=
sp=3B</span><a href=3D"mailto:fineberg%20at%20vline.com" rel=3D"nofollow" t=
arget=3D"_blank">&lt=3Bfineberg at vline.com&gt=3B</a></td>=0A=
                                                        </tr>=0A=
                                                      </tbody>=0A=
                                                    </table>=0A=
                                                    <br>=0A=
                                                    <br>=0A=
                                                    <pre style=3D"width:939=
.59px=3Bwhite-space:pre-wrap=3B">A new version of I-D=2C draft-fineberg-avt=
ext-temporal-layer-ext-00.txt=0A=
has been successfully submitted by Adam Fineberg and posted to the=0A=
IETF repository.=0A=
=0A=
Filename:	 draft-fineberg-avtext-temporal-layer-ext=0A=
Revision:	 00=0A=
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information=0A=
Creation date:	 2013-07-08=0A=
Group:		 Individual Submission=0A=
Number of pages: 6=0A=
URL:             <a href=3D"http://www.ietf.org/internet-drafts/draft-fineb=
erg-avtext-temporal-layer-ext-00.txt" rel=3D"nofollow" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-=
00.txt</a>=0A=
Status:          <a href=3D"http://datatracker.ietf.org/doc/draft-fineberg-=
avtext-temporal-layer-ext" rel=3D"nofollow" target=3D"_blank">http://datatr=
acker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>=0A=
Htmlized:        <a href=3D"http://tools.ietf.org/html/draft-fineberg-avtex=
t-temporal-layer-ext-00" rel=3D"nofollow" target=3D"_blank">http://tools.ie=
tf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>=0A=
=0A=
=0A=
Abstract:=0A=
   This document defines a mechanism by which packets of Real-Time=0A=
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can=0A=
   indicate=2C in an RTP header extension=2C the temporal layer information=
=0A=
   about the frame encoded in the RTP packet.  This information can be=0A=
   used in a middlebox performing bandwidth management of streams=0A=
   without requiring it to decrypt the streams.=0A=
</pre>=0A=
                                                  </div>=0A=
                                                  <br>=0A=
                                                  _________________________=
______________________=0A=
                                                  rtcweb mailing list <a hr=
ef=3D"mailto:rtcweb@ietf.org" target=3D"_blank">=0A=
                                                    rtcweb@ietf.org</a>=0A=
                                                  <a href=3D"https://www.ie=
tf.org/mailman/listinfo/rtcweb" target=3D"_blank">=0A=
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>=0A=
                                              </div>=0A=
                                            </blockquote>=0A=
                                            <br>=0A=
                                            <pre>-- =0A=
Regards=2C=0A=
Adam</pre>=0A=
                                          </div>=0A=
                                        </div>=0A=
                                      </blockquote>=0A=
                                      <br>=0A=
                                    </div>=0A=
                                  </blockquote>=0A=
                                </div>=0A=
                              </div>=0A=
                            </div>=0A=
                            <br>=0A=
_______________________________________________<br>=0A=
                            rtcweb mailing list<br>=0A=
                            <a href=3D"mailto:rtcweb@ietf.org" target=3D"_b=
lank">rtcweb@ietf.org</a><br>=0A=
                            <a href=3D"https://www.ietf.org/mailman/listinf=
o/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a=
><br>=0A=
                            <br>=0A=
                          </blockquote>=0A=
                        </div>=0A=
                        <br>=0A=
                      </div>=0A=
                    </div>=0A=
                  </div>=0A=
                </div>=0A=
              </span><br>=0A=
              <fieldset class=3D"ecxmimeAttachmentHeader"></fieldset>=0A=
              <br>=0A=
              <pre>_______________________________________________=0A=
avtext mailing list=0A=
<a class=3D"ecxmoz-txt-link-abbreviated" href=3D"mailto:avtext@ietf.org">av=
text@ietf.org</a><a class=3D"ecxmoz-txt-link-freetext" href=3D"https://www.=
ietf.org/mailman/listinfo/avtext" target=3D"_blank">https://www.ietf.org/ma=
ilman/listinfo/avtext</a></pre>=0A=
            </blockquote>=0A=
            <br>=0A=
            <pre class=3D"ecxmoz-signature">-- =0A=
Regards=2C=0A=
Adam</pre>=0A=
          </div>=0A=
        </div>=0A=
      </span>=0A=
    </blockquote>=0A=
    <br>=0A=
    <pre class=3D"ecxmoz-signature">-- =0A=
Regards=2C=0A=
Adam</pre></div> 		 	   		  </div></body>
</html>=

--_788a735f-7672-45a0-80b1-0a9af35b1860_--

From hadriel.kaplan@oracle.com  Tue Jul 23 12:37:27 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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On Jul 20, 2013, at 10:26 AM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:

> 2013/7/20 Cullen Jennings <fluffy@iii.ca>:
>> I will note that over half of that very small sample set is folks =
that have been long termed involved in SIP and VoIP so, uh, might not be =
a representative sample of type of people you are portraying it as.
>=20
> It is even worse if people interested and expert in SIP tells you that
> the current SDP-blob-based API is really wrong for using SIP in
> WebRTC, don't you think?

No, not at all - we're all jaded from past experience and we know how =
bad it is already, so Cullen's right we're all tainted.  Hopefully if we =
ask new people, they won't know the truth and give us much more positive =
results instead.  Or new people might even see the SDP-based "API" and =
think: "Oh cool, the API is a string blob of stuff in some archaic =
inconsistent syntax format with hidden state-based dependencies and =
rules... awesome!  I always wanted to party like it's 1999, and here's =
my chance!"

-hadriel


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Adam Roach <adam@nostrum.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Unified plan IPR
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Date: Tue, 23 Jul 2013 20:18:39 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, tim panton <thp@westhawk.co.uk>
Subject: Re: [rtcweb] Unified plan IPR
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Adam Roach:
>
> I doubt agitating for action on these mailing lists is likely to produce =
useful
> results.

This appears to be generally true, yes.

One wonders why we have them.

Matthew Kaufman


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From: Justin Uberti <juberti@google.com>
Date: Tue, 23 Jul 2013 16:45:12 -0400
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--047d7b2e429a59343f04e233f183
Content-Type: text/plain; charset=UTF-8

So there are two parts to the proposal - the HTTP request for the
credentials and interactions with WebRTC, and the stateless implementation
that avoids the need for communication between web and TURN servers. And it
is true that other implementations besides the suggested stateless one are
possible, e.g. OAuth2.

Would you prefer that I address the protocol and implementation parts
separately in the document, and mention that other implementations with the
same HTTP protocol are possible? RFC 5766 does something similar, where it
defines the TURN protocol, but in Section 6.2 talks about how a "stateless
stack" implementation can be used to simplify the processing on the TURN
server.


On Mon, Jul 22, 2013 at 5:37 PM, Oleg Moskalenko <mom040267@gmail.com>wrote:

> The overall proposal by Justin:
>
> http://tools.ietf.org/html/draft-uberti-rtcweb-turn-rest-00
>
> has been implemented in our rfc5766-turn-server for months, and I see
> nothing proprietary in the proposal. I liked the proposal from the
> beginning. Its been discussed and its been quite handy for the users - I
> hear stories of success from the users.  I believe that Philipp shares the
> same view.
>
> One especially good thing about the proposal is that is does not
> contradict by any means to the RFC 5766. It just specifies the way how the
> passwords are stored/generated - and the RFC 5766 does not say anything
> about that matter. It means that Justin's proposal is 100% compatible with
> RFC 5766.
>
> Thanks
> Oleg
> http://code.google.com/p/rfc5766-turn-server/
>
>
>
>
> On Mon, Jul 22, 2013 at 1:46 PM, Philipp Hancke <fippo@goodadvice.pages.de
> > wrote:
>
>> Am 22.07.2013 22:16, schrieb Adam Roach:
>>
>>> What Justin proposes gets us this inter-vendor interop cheaply and
>>> easily. What you're counterproposing forces people into a single-vendor
>>> (or, at least, partnered) solution, which kinda defeats the purpose of
>>> defining this in an SDO in the first place.
>>>
>>
>> Right. What Justin proposed was originally located at
>> https://code.google.com/p/**webrtc/issues/detail?id=1197<https://code.google.com/p/webrtc/issues/detail?id=1197>
>> I liked the plan, implemented it in restund and improved it a litle. It
>> was discussed at https://groups.google.com/**
>> forum/#!topic/discuss-webrtc/**nn8b6UboqRA/discussion<https://groups.google.com/forum/#!topic/discuss-webrtc/nn8b6UboqRA/discussion>and also implemented in rfc-5766-turn-server.
>>
>> Rough consensus and running code...
>>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www.ietf.org/mailman/listinfo/behave
>
>

--047d7b2e429a59343f04e233f183
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<div dir=3D"ltr">So there are two parts to the proposal - the HTTP request =
for the credentials and interactions with WebRTC, and the stateless impleme=
ntation that avoids the need for communication between web and TURN servers=
. And it is true that other implementations besides the suggested stateless=
 one are possible, e.g. OAuth2.<br>

<div><br></div><div>Would you prefer that I address the protocol and implem=
entation parts separately in the document, and mention that other implement=
ations with the same HTTP protocol are possible? RFC 5766 does something si=
milar, where it defines the TURN protocol, but in Section 6.2 talks about h=
ow a &quot;stateless stack&quot; implementation can be used to simplify the=
 processing on the TURN server.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Jul 22, 2013 at 5:37 PM, Oleg Moskalenko <span dir=3D"ltr">&lt;<a href=3D"=
mailto:mom040267@gmail.com" target=3D"_blank">mom040267@gmail.com</a>&gt;</=
span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>The overall propo=
sal by Justin:<br><br><a href=3D"http://tools.ietf.org/html/draft-uberti-rt=
cweb-turn-rest-00" target=3D"_blank">http://tools.ietf.org/html/draft-ubert=
i-rtcweb-turn-rest-00</a><br>

<br></div>has been implemented in our rfc5766-turn-server for months, and I=
 see nothing proprietary in the proposal. I liked the proposal from the beg=
inning. Its been discussed and its been quite handy for the users - I hear =
stories of success from the users.=C2=A0 I believe that Philipp shares the =
same view.<br>


<br></div><div>One especially good thing about the proposal is that is does=
 not contradict by any means to the RFC 5766. It just specifies the way how=
 the passwords are stored/generated - and the RFC 5766 does not say anythin=
g about that matter. It means that Justin&#39;s proposal is 100% compatible=
 with RFC 5766.<br>


</div><div><br></div>Thanks<br>Oleg<br><a href=3D"http://code.google.com/p/=
rfc5766-turn-server/" target=3D"_blank">http://code.google.com/p/rfc5766-tu=
rn-server/</a><br><br><br></div><div class=3D"HOEnZb"><div class=3D"h5"><di=
v class=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Mon, Jul 22, 2013 at 1:46 PM, Philipp=
 Hancke <span dir=3D"ltr">&lt;<a href=3D"mailto:fippo@goodadvice.pages.de" =
target=3D"_blank">fippo@goodadvice.pages.de</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Am 22.07.2013 22:16, schrieb Adam Roach:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
What Justin proposes gets us this inter-vendor interop cheaply and<br>
easily. What you&#39;re counterproposing forces people into a single-vendor=
<br>
(or, at least, partnered) solution, which kinda defeats the purpose of<br>
defining this in an SDO in the first place.<br>
</blockquote>
<br>
Right. What Justin proposed was originally located at<br>
<a href=3D"https://code.google.com/p/webrtc/issues/detail?id=3D1197" target=
=3D"_blank">https://code.google.com/p/<u></u>webrtc/issues/detail?id=3D1197=
</a><br>
I liked the plan, implemented it in restund and improved it a litle. It was=
 discussed at <a href=3D"https://groups.google.com/forum/#!topic/discuss-we=
brtc/nn8b6UboqRA/discussion" target=3D"_blank">https://groups.google.com/<u=
></u>forum/#!topic/discuss-webrtc/<u></u>nn8b6UboqRA/discussion</a> and als=
o implemented in rfc-5766-turn-server.<br>



<br>
Rough consensus and running code...<br>
</blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
Behave mailing list<br>
<a href=3D"mailto:Behave@ietf.org">Behave@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/behave" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/behave</a><br>
<br></blockquote></div><br></div>

--047d7b2e429a59343f04e233f183--

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On 23/07/2013 12:51 PM, Adam Roach wrote:
> On 7/23/13 11:22, Roman Shpount wrote:
>> The situation would be a bit clearer if patent holders were to 
>> provide the licensing policy regarding this IPR release. Given that 
>> Ericsson is actively involved in this working group, I think it would 
>> be reasonable to ask them for this.
>
> If process has been properly followed, the IPR holder has already been 
> notified by the IETF executive director. See 
> http://www.ietf.org/rfc/rfc4879.txt (section 1 paragraph 1)
>
> I doubt agitating for action on these mailing lists is likely to 
> produce useful results.
>
> /a

Hi Adam,

     I'm a bit concerned about the optics of what just happened.

  * The Working Group has been pushing for the use of SDP since 2011
    (see http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
  * The first post related to the use of SDP in WebRTC came from
    Christer Holmberg of Ericsson on September 14th, 2011.
  * One of the Chairs of the Working Group and one of the Specification
    editors are from Ericsson.
  * There has been a substantial push against the use of SDP by some
    mailing list participants, but this was rejected by the Working Group.
  * Suddenly we find out that Ericsson has filed two patents related to
    the use of SDP in WebRTC and these were filed *after* Ericsson
    actively pushed for the use of SDP.

     Isn't there a conflict of interest here?

     As a Web Developer who doesn't want/need SDP to begin with, I am 
finding this a bitter pill to swallow. I have no problem with other 
people using SDP (all the power to them) but, with this IPR discovery, 
forcing their preference on me will have real-world consequences (no 
less than had we mandated the use H264 in WebRTC).

 1. Do the patents imply that Web Developers will have to pay patents
    when deploying application on top of the Browser or Native APIs?
 2. Is there a way to retrofit the API so those of us who do not
    want/need to use SDP are not forced to license this IPR? For
    example, the specification states that the initial offer/answer
    mechanism is out of scope. Could we do the same for SDP?

Thank you,
Gili

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  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 23/07/2013 12:51 PM, Adam Roach
      wrote:<br>
    </div>
    <blockquote cite="mid:51EEB495.4070404@nostrum.com" type="cite">On
      7/23/13 11:22, Roman Shpount wrote:
      <br>
      <blockquote type="cite">The situation would be a bit clearer if
        patent holders were to provide the licensing policy regarding
        this IPR release. Given that Ericsson is actively involved in
        this working group, I think it would be reasonable to ask them
        for this.
        <br>
      </blockquote>
      <br>
      If process has been properly followed, the IPR holder has already
      been notified by the IETF executive director. See
      <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfc/rfc4879.txt">http://www.ietf.org/rfc/rfc4879.txt</a> (section 1 paragraph 1)
      <br>
      <br>
      I doubt agitating for action on these mailing lists is likely to
      produce useful results.
      <br>
      <br>
      /a
      <br>
    </blockquote>
    <br>
    Hi Adam,<br>
    <br>
    &nbsp;&nbsp;&nbsp; I'm a bit concerned about the optics of what just happened.<br>
    <ul>
      <li>The Working Group has been pushing for the use of SDP since
        2011 (see <a
          href="http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html">http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html</a>).</li>
      <li>The first post related to the use of SDP in WebRTC came from
        Christer Holmberg of Ericsson on September 14th, 2011.</li>
      <li>One of the Chairs of the Working Group and one of the
        Specification editors are from Ericsson.</li>
      <li>There has been a substantial push against the use of SDP by
        some mailing list participants, but this was rejected by the
        Working Group.</li>
      <li>Suddenly we find out that Ericsson has filed two patents
        related to the use of SDP in WebRTC and these were filed *after*
        Ericsson actively pushed for the use of SDP.</li>
    </ul>
    <p>&nbsp;&nbsp;&nbsp; Isn't there a conflict of interest here?<br>
    </p>
    &nbsp;&nbsp;&nbsp; As a Web Developer who doesn't want/need SDP to begin with, I am
    finding this a bitter pill to swallow. I have no problem with other
    people using SDP (all the power to them) but, with this IPR
    discovery, forcing their preference on me will have real-world
    consequences (no less than had we mandated the use H264 in WebRTC).<br>
    <ol>
      <li>Do the patents imply that Web Developers will have to pay
        patents when deploying application on top of the Browser or
        Native APIs?</li>
      <li>Is there a way to retrofit the API so those of us who do not
        want/need to use SDP are not forced to license this IPR? For
        example, the specification states that the initial offer/answer
        mechanism is out of scope. Could we do the same for SDP?<br>
      </li>
    </ol>
    Thank you,<br>
    Gili<br>
  </body>
</html>

--------------060605090502050100050201--

From ted.ietf@gmail.com  Tue Jul 23 15:08:35 2013
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Date: Tue, 23 Jul 2013 15:08:33 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: cowwoc <cowwoc@bbs.darktech.org>
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Subject: Re: [rtcweb] Unified plan IPR
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Hi Gili,

I've snipped some of this, but I want to note that it is not really
possible for Adam to clarify this at the moment; like the rest of us, he
will have to wait for the folks who filed the patent to clarify the
situation.   He is listed in the IPR declaration only because he brought
the patent to the attention of the IETF, not because he is an inventor or
assignee.

The IETF patent process has already pointed out (in RFC 4879); please check
it for the next steps.

thanks,

Ted Hardie

On Tue, Jul 23, 2013 at 2:58 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:

>
> Hi Adam,
>
>     I'm a bit concerned about the optics of what just happened.
>
>    - The Working Group has been pushing for the use of SDP since 2011
>    (see http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>    - The first post related to the use of SDP in WebRTC came from
>    Christer Holmberg of Ericsson on September 14th, 2011.
>    - One of the Chairs of the Working Group and one of the Specification
>    editors are from Ericsson.
>    - There has been a substantial push against the use of SDP by some
>    mailing list participants, but this was rejected by the Working Group.
>    - Suddenly we find out that Ericsson has filed two patents related to
>    the use of SDP in WebRTC and these were filed *after* Ericsson actively
>    pushed for the use of SDP.
>
>     Isn't there a conflict of interest here?
>      As a Web Developer who doesn't want/need SDP to begin with, I am
> finding this a bitter pill to swallow. I have no problem with other people
> using SDP (all the power to them) but, with this IPR discovery, forcing
> their preference on me will have real-world consequences (no less than had
> we mandated the use H264 in WebRTC).
>
>    1. Do the patents imply that Web Developers will have to pay patents
>    when deploying application on top of the Browser or Native APIs?
>    2. Is there a way to retrofit the API so those of us who do not
>    want/need to use SDP are not forced to license this IPR? For example, the
>    specification states that the initial offer/answer mechanism is out of
>    scope. Could we do the same for SDP?
>
> Thank you,
> Gili
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e0122a7fca93cdf04e235084c
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Hi Gili,<br><br>I&#39;ve snipped some of this, but I want to note that it i=
s not really possible for Adam to clarify this at the moment; like the rest=
 of us, he will have to wait for the folks who filed the patent to clarify =
the situation.=A0=A0 He is listed in the IPR declaration only because he br=
ought the patent to the attention of the IETF, not because he is an invento=
r or assignee.<br>
<br>The IETF patent process has already pointed out (in RFC 4879); please c=
heck it for the next steps.<br><br>thanks,<br><br>Ted Hardie<br><br>On Tue,=
 Jul 23, 2013 at 2:58 PM, cowwoc <span dir=3D"ltr">&lt;<a href=3D"mailto:co=
wwoc@bbs.darktech.org" target=3D"_blank">cowwoc@bbs.darktech.org</a>&gt;</s=
pan> wrote:<br>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"im"><br></div>
    Hi Adam,<br>
    <br>
    =A0=A0=A0 I&#39;m a bit concerned about the optics of what just happene=
d.<br>
    <ul>
      <li>The Working Group has been pushing for the use of SDP since
        2011 (see <a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/cu=
rrent/mail15.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/r=
tcweb/current/mail15.html</a>).</li>
      <li>The first post related to the use of SDP in WebRTC came from
        Christer Holmberg of Ericsson on September 14th, 2011.</li>
      <li>One of the Chairs of the Working Group and one of the
        Specification editors are from Ericsson.</li>
      <li>There has been a substantial push against the use of SDP by
        some mailing list participants, but this was rejected by the
        Working Group.</li>
      <li>Suddenly we find out that Ericsson has filed two patents
        related to the use of SDP in WebRTC and these were filed *after*
        Ericsson actively pushed for the use of SDP.</li>
    </ul>
    <p>=A0=A0=A0 Isn&#39;t there a conflict of interest here?<br>
    </p>
    =A0=A0=A0 As a Web Developer who doesn&#39;t want/need SDP to begin wit=
h, I am
    finding this a bitter pill to swallow. I have no problem with other
    people using SDP (all the power to them) but, with this IPR
    discovery, forcing their preference on me will have real-world
    consequences (no less than had we mandated the use H264 in WebRTC).<br>
    <ol>
      <li>Do the patents imply that Web Developers will have to pay
        patents when deploying application on top of the Browser or
        Native APIs?</li>
      <li>Is there a way to retrofit the API so those of us who do not
        want/need to use SDP are not forced to license this IPR? For
        example, the specification states that the initial offer/answer
        mechanism is out of scope. Could we do the same for SDP?<br>
      </li>
    </ol>
    Thank you,<br>
    Gili<br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--089e0122a7fca93cdf04e235084c--

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Subject: Re: [rtcweb] Unified plan IPR
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Hi Ted,

     Thank you for the clarification.

     In the meantime, Adam (and others) can comment question #2.

Gili

On 23/07/2013 6:08 PM, Ted Hardie wrote:
> Hi Gili,
>
> I've snipped some of this, but I want to note that it is not really 
> possible for Adam to clarify this at the moment; like the rest of us, 
> he will have to wait for the folks who filed the patent to clarify the 
> situation.   He is listed in the IPR declaration only because he 
> brought the patent to the attention of the IETF, not because he is an 
> inventor or assignee.
>
> The IETF patent process has already pointed out (in RFC 4879); please 
> check it for the next steps.
>
> thanks,
>
> Ted Hardie
>
> On Tue, Jul 23, 2013 at 2:58 PM, cowwoc <cowwoc@bbs.darktech.org 
> <mailto:cowwoc@bbs.darktech.org>> wrote:
>
>
>     Hi Adam,
>
>         I'm a bit concerned about the optics of what just happened.
>
>       * The Working Group has been pushing for the use of SDP since
>         2011 (see
>         http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>       * The first post related to the use of SDP in WebRTC came from
>         Christer Holmberg of Ericsson on September 14th, 2011.
>       * One of the Chairs of the Working Group and one of the
>         Specification editors are from Ericsson.
>       * There has been a substantial push against the use of SDP by
>         some mailing list participants, but this was rejected by the
>         Working Group.
>       * Suddenly we find out that Ericsson has filed two patents
>         related to the use of SDP in WebRTC and these were filed
>         *after* Ericsson actively pushed for the use of SDP.
>
>         Isn't there a conflict of interest here?
>
>         As a Web Developer who doesn't want/need SDP to begin with, I
>     am finding this a bitter pill to swallow. I have no problem with
>     other people using SDP (all the power to them) but, with this IPR
>     discovery, forcing their preference on me will have real-world
>     consequences (no less than had we mandated the use H264 in WebRTC).
>
>      1. Do the patents imply that Web Developers will have to pay
>         patents when deploying application on top of the Browser or
>         Native APIs?
>      2. Is there a way to retrofit the API so those of us who do not
>         want/need to use SDP are not forced to license this IPR? For
>         example, the specification states that the initial
>         offer/answer mechanism is out of scope. Could we do the same
>         for SDP?
>
>     Thank you,
>     Gili
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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    <div class="moz-cite-prefix">Hi Ted,<br>
      <br>
      &nbsp;&nbsp;&nbsp; Thank you for the clarification.<br>
      <br>
      &nbsp;&nbsp;&nbsp; In the meantime, Adam (and others) can comment question #2.<br>
      <br>
      Gili<br>
      <br>
      On 23/07/2013 6:08 PM, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMBwBP2p5UG95h7rY9CAwUpXpRiLKjne-bEn0pX2gooS7w@mail.gmail.com"
      type="cite">Hi Gili,<br>
      <br>
      I've snipped some of this, but I want to note that it is not
      really possible for Adam to clarify this at the moment; like the
      rest of us, he will have to wait for the folks who filed the
      patent to clarify the situation.&nbsp;&nbsp; He is listed in the IPR
      declaration only because he brought the patent to the attention of
      the IETF, not because he is an inventor or assignee.<br>
      <br>
      The IETF patent process has already pointed out (in RFC 4879);
      please check it for the next steps.<br>
      <br>
      thanks,<br>
      <br>
      Ted Hardie<br>
      <br>
      On Tue, Jul 23, 2013 at 2:58 PM, cowwoc <span dir="ltr">&lt;<a
          moz-do-not-send="true" href="mailto:cowwoc@bbs.darktech.org"
          target="_blank">cowwoc@bbs.darktech.org</a>&gt;</span> wrote:<br>
      <div class="gmail_quote">
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div text="#000000" bgcolor="#FFFFFF">
            <div class="im"><br>
            </div>
            Hi Adam,<br>
            <br>
            &nbsp;&nbsp;&nbsp; I'm a bit concerned about the optics of what just
            happened.<br>
            <ul>
              <li>The Working Group has been pushing for the use of SDP
                since 2011 (see <a moz-do-not-send="true"
                  href="http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html"
                  target="_blank">http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html</a>).</li>
              <li>The first post related to the use of SDP in WebRTC
                came from Christer Holmberg of Ericsson on September
                14th, 2011.</li>
              <li>One of the Chairs of the Working Group and one of the
                Specification editors are from Ericsson.</li>
              <li>There has been a substantial push against the use of
                SDP by some mailing list participants, but this was
                rejected by the Working Group.</li>
              <li>Suddenly we find out that Ericsson has filed two
                patents related to the use of SDP in WebRTC and these
                were filed *after* Ericsson actively pushed for the use
                of SDP.</li>
            </ul>
            <p>&nbsp;&nbsp;&nbsp; Isn't there a conflict of interest here?<br>
            </p>
            &nbsp;&nbsp;&nbsp; As a Web Developer who doesn't want/need SDP to begin
            with, I am finding this a bitter pill to swallow. I have no
            problem with other people using SDP (all the power to them)
            but, with this IPR discovery, forcing their preference on me
            will have real-world consequences (no less than had we
            mandated the use H264 in WebRTC).<br>
            <ol>
              <li>Do the patents imply that Web Developers will have to
                pay patents when deploying application on top of the
                Browser or Native APIs?</li>
              <li>Is there a way to retrofit the API so those of us who
                do not want/need to use SDP are not forced to license
                this IPR? For example, the specification states that the
                initial offer/answer mechanism is out of scope. Could we
                do the same for SDP?<br>
              </li>
            </ol>
            Thank you,<br>
            Gili<br>
          </div>
          <br>
          _______________________________________________<br>
          rtcweb mailing list<br>
          <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          <a moz-do-not-send="true"
            href="https://www.ietf.org/mailman/listinfo/rtcweb"
            target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          <br>
        </blockquote>
      </div>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------030708030001000801050905--

From adam@nostrum.com  Tue Jul 23 15:21:54 2013
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Subject: Re: [rtcweb] Unified plan IPR
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The chairs have asked us not to cross-post. As this pertains to an IETF 
IPR declaration, I'll speak to it here.

I'll note that your response is a vast overreaction at this juncture, as 
(1) these are merely applications, not granted patents; and (2) Ericsson 
has not yet indicated their intentions regarding the licensing terms of 
any patents that may result.

In terms of your two questions: I'm not a laywer, so I'm not able to 
speak to the applicability of the patents in any authoritative fashion. 
All that I can really say is that I have a reasonable belief that the 
claims of these applications, if granted, would apply to the draft in 
question.

I will make one factual observation, without any interpretation, from 
which you can draw your own conclusions: the independent claims of the 
patent applications in question do not mention SDP.

/a



On 7/23/13 16:58, cowwoc wrote:
>
>     I'm a bit concerned about the optics of what just happened.
>
>   * The Working Group has been pushing for the use of SDP since 2011
>     (see http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>   * The first post related to the use of SDP in WebRTC came from
>     Christer Holmberg of Ericsson on September 14th, 2011.
>   * One of the Chairs of the Working Group and one of the
>     Specification editors are from Ericsson.
>   * There has been a substantial push against the use of SDP by some
>     mailing list participants, but this was rejected by the Working Group.
>   * Suddenly we find out that Ericsson has filed two patents related
>     to the use of SDP in WebRTC and these were filed *after* Ericsson
>     actively pushed for the use of SDP.
>
>     Isn't there a conflict of interest here?
>
>     As a Web Developer who doesn't want/need SDP to begin with, I am 
> finding this a bitter pill to swallow. I have no problem with other 
> people using SDP (all the power to them) but, with this IPR discovery, 
> forcing their preference on me will have real-world consequences (no 
> less than had we mandated the use H264 in WebRTC).
>
>  1. Do the patents imply that Web Developers will have to pay patents
>     when deploying application on top of the Browser or Native APIs?
>  2. Is there a way to retrofit the API so those of us who do not
>     want/need to use SDP are not forced to license this IPR? For
>     example, the specification states that the initial offer/answer
>     mechanism is out of scope. Could we do the same for SDP?
>
> Thank you,
> Gili
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">The chairs have asked us not to
      cross-post. As this pertains to an IETF IPR declaration, I'll
      speak to it here.<br>
      <br>
      I'll note that your response is a vast overreaction at this
      juncture, as (1) these are merely applications, not granted
      patents; and (2) Ericsson has not yet indicated their intentions
      regarding the licensing terms of any patents that may result.<br>
      <br>
      In terms of your two questions: I'm not a laywer, so I'm not able
      to speak to the applicability of the patents in any authoritative
      fashion. All that I can really say is that I have a reasonable
      belief that the claims of these applications, if granted, would
      apply to the draft in question.<br>
      <br>
      I will make one factual observation, without any interpretation,
      from which you can draw your own conclusions: the independent
      claims of the patent applications in question do not mention SDP.<br>
      <br>
      /a<br>
      <br>
      <br>
      <br>
      On 7/23/13 16:58, cowwoc wrote:<br>
    </div>
    <blockquote cite="mid:51EEFC6B.9090503@bbs.darktech.org" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <br>
      &nbsp;&nbsp;&nbsp; I'm a bit concerned about the optics of what just happened.<br>
      <ul>
        <li>The Working Group has been pushing for the use of SDP since
          2011 (see <a moz-do-not-send="true"
            href="http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html">http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html</a>).</li>
        <li>The first post related to the use of SDP in WebRTC came from
          Christer Holmberg of Ericsson on September 14th, 2011.</li>
        <li>One of the Chairs of the Working Group and one of the
          Specification editors are from Ericsson.</li>
        <li>There has been a substantial push against the use of SDP by
          some mailing list participants, but this was rejected by the
          Working Group.</li>
        <li>Suddenly we find out that Ericsson has filed two patents
          related to the use of SDP in WebRTC and these were filed
          *after* Ericsson actively pushed for the use of SDP.</li>
      </ul>
      <p>&nbsp;&nbsp;&nbsp; Isn't there a conflict of interest here?<br>
      </p>
      &nbsp;&nbsp;&nbsp; As a Web Developer who doesn't want/need SDP to begin with, I
      am finding this a bitter pill to swallow. I have no problem with
      other people using SDP (all the power to them) but, with this IPR
      discovery, forcing their preference on me will have real-world
      consequences (no less than had we mandated the use H264 in
      WebRTC).<br>
      <ol>
        <li>Do the patents imply that Web Developers will have to pay
          patents when deploying application on top of the Browser or
          Native APIs?</li>
        <li>Is there a way to retrofit the API so those of us who do not
          want/need to use SDP are not forced to license this IPR? For
          example, the specification states that the initial
          offer/answer mechanism is out of scope. Could we do the same
          for SDP?<br>
        </li>
      </ol>
      Thank you,<br>
      Gili<br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------030705050605070309010906--

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Subject: Re: [rtcweb] Unified plan IPR
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--f46d0442808ee757b304e235b4ec
Content-Type: text/plain; charset=ISO-8859-1

I did not want to blow this out of proportion. All I wanted to point out
that that licensing terms are missing and until the licensing terms are
clear this does present a certain risk. I do agree that given current
active participation of Ericsson in this list, this risk is likely low. One
other thing that I wanted to mention was that some aspects of the unified
draft can be adjusted to limit the exposure to the declared IPR.

Regards,
_____________
Roman Shpount


On Tue, Jul 23, 2013 at 6:21 PM, Adam Roach <adam@nostrum.com> wrote:

>  The chairs have asked us not to cross-post. As this pertains to an IETF
> IPR declaration, I'll speak to it here.
>
> I'll note that your response is a vast overreaction at this juncture, as
> (1) these are merely applications, not granted patents; and (2) Ericsson
> has not yet indicated their intentions regarding the licensing terms of any
> patents that may result.
>
> In terms of your two questions: I'm not a laywer, so I'm not able to speak
> to the applicability of the patents in any authoritative fashion. All that
> I can really say is that I have a reasonable belief that the claims of
> these applications, if granted, would apply to the draft in question.
>
> I will make one factual observation, without any interpretation, from
> which you can draw your own conclusions: the independent claims of the
> patent applications in question do not mention SDP.
>
> /a
>
>
>
>
> On 7/23/13 16:58, cowwoc wrote:
>
>
>     I'm a bit concerned about the optics of what just happened.
>
>    - The Working Group has been pushing for the use of SDP since 2011
>    (see http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>    - The first post related to the use of SDP in WebRTC came from
>    Christer Holmberg of Ericsson on September 14th, 2011.
>    - One of the Chairs of the Working Group and one of the Specification
>    editors are from Ericsson.
>    - There has been a substantial push against the use of SDP by some
>    mailing list participants, but this was rejected by the Working Group.
>    - Suddenly we find out that Ericsson has filed two patents related to
>    the use of SDP in WebRTC and these were filed *after* Ericsson actively
>    pushed for the use of SDP.
>
>     Isn't there a conflict of interest here?
>      As a Web Developer who doesn't want/need SDP to begin with, I am
> finding this a bitter pill to swallow. I have no problem with other people
> using SDP (all the power to them) but, with this IPR discovery, forcing
> their preference on me will have real-world consequences (no less than had
> we mandated the use H264 in WebRTC).
>
>    1. Do the patents imply that Web Developers will have to pay patents
>    when deploying application on top of the Browser or Native APIs?
>    2. Is there a way to retrofit the API so those of us who do not
>    want/need to use SDP are not forced to license this IPR? For example, the
>    specification states that the initial offer/answer mechanism is out of
>    scope. Could we do the same for SDP?
>
> Thank you,
> Gili
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--f46d0442808ee757b304e235b4ec
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I did not want to blow this out of proportion. All I wanted to point out th=
at that licensing terms are missing and until the licensing terms are clear=
 this does present a certain risk. I do agree that given current active par=
ticipation of Ericsson in this list, this risk is likely low. One other thi=
ng that I wanted to mention was that some aspects of the unified draft can =
be adjusted to limit the exposure to the declared IPR.<br>
<br>Regards,<br clear=3D"all"><div>_____________<br>Roman Shpount</div>
<br><br><div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 6:21 PM, Adam Ro=
ach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_bl=
ank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">

 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>The chairs have asked us not to
      cross-post. As this pertains to an IETF IPR declaration, I&#39;ll
      speak to it here.<br>
      <br>
      I&#39;ll note that your response is a vast overreaction at this
      juncture, as (1) these are merely applications, not granted
      patents; and (2) Ericsson has not yet indicated their intentions
      regarding the licensing terms of any patents that may result.<br>
      <br>
      In terms of your two questions: I&#39;m not a laywer, so I&#39;m not =
able
      to speak to the applicability of the patents in any authoritative
      fashion. All that I can really say is that I have a reasonable
      belief that the claims of these applications, if granted, would
      apply to the draft in question.<br>
      <br>
      I will make one factual observation, without any interpretation,
      from which you can draw your own conclusions: the independent
      claims of the patent applications in question do not mention SDP.<spa=
n class=3D"HOEnZb"><font color=3D"#888888"><br>
      <br>
      /a</font></span><div class=3D"im"><br>
      <br>
      <br>
      <br>
      On 7/23/13 16:58, cowwoc wrote:<br>
    </div></div>
    <blockquote type=3D"cite"><div class=3D"im">
     =20
      <br>
      =A0=A0=A0 I&#39;m a bit concerned about the optics of what just happe=
ned.<br>
      <ul>
        <li>The Working Group has been pushing for the use of SDP since
          2011 (see <a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/=
current/mail15.html" target=3D"_blank">http://www.ietf.org/mail-archive/web=
/rtcweb/current/mail15.html</a>).</li>
        <li>The first post related to the use of SDP in WebRTC came from
          Christer Holmberg of Ericsson on September 14th, 2011.</li>
        <li>One of the Chairs of the Working Group and one of the
          Specification editors are from Ericsson.</li>
        <li>There has been a substantial push against the use of SDP by
          some mailing list participants, but this was rejected by the
          Working Group.</li>
        <li>Suddenly we find out that Ericsson has filed two patents
          related to the use of SDP in WebRTC and these were filed
          *after* Ericsson actively pushed for the use of SDP.</li>
      </ul>
      <p>=A0=A0=A0 Isn&#39;t there a conflict of interest here?<br>
      </p>
      =A0=A0=A0 As a Web Developer who doesn&#39;t want/need SDP to begin w=
ith, I
      am finding this a bitter pill to swallow. I have no problem with
      other people using SDP (all the power to them) but, with this IPR
      discovery, forcing their preference on me will have real-world
      consequences (no less than had we mandated the use H264 in
      WebRTC).<br>
      <ol>
        <li>Do the patents imply that Web Developers will have to pay
          patents when deploying application on top of the Browser or
          Native APIs?</li>
        <li>Is there a way to retrofit the API so those of us who do not
          want/need to use SDP are not forced to license this IPR? For
          example, the specification states that the initial
          offer/answer mechanism is out of scope. Could we do the same
          for SDP?<br>
        </li>
      </ol>
      Thank you,<br>
      Gili<br>
      <br>
      <fieldset></fieldset>
      <br>
      </div><div class=3D"im"><pre>________________________________________=
_______
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </div></blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--f46d0442808ee757b304e235b4ec--

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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 23 Jul 2013 16:07:55 -0700
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Subject: Re: [rtcweb] Unified plan IPR
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--e89a8f6475cf5b97c104e235dffe
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On Tue, Jul 23, 2013 at 3:56 PM, Roman Shpount <roman@telurix.com> wrote:

> I did not want to blow this out of proportion. All I wanted to point out
> that that licensing terms are missing and until the licensing terms are
> clear this does present a certain risk. I do agree that given current
> active participation of Ericsson in this list, this risk is likely low. One
> other thing that I wanted to mention was that some aspects of the unified
> draft can be adjusted to limit the exposure to the declared IPR.
>

Roman,

That seems like something that would be useful if you could lay out in more
detail.

Best,
-Ekr

Regards,
> _____________
> Roman Shpount
>
>
> On Tue, Jul 23, 2013 at 6:21 PM, Adam Roach <adam@nostrum.com> wrote:
>
>>  The chairs have asked us not to cross-post. As this pertains to an IETF
>> IPR declaration, I'll speak to it here.
>>
>> I'll note that your response is a vast overreaction at this juncture, as
>> (1) these are merely applications, not granted patents; and (2) Ericsson
>> has not yet indicated their intentions regarding the licensing terms of any
>> patents that may result.
>>
>> In terms of your two questions: I'm not a laywer, so I'm not able to
>> speak to the applicability of the patents in any authoritative fashion. All
>> that I can really say is that I have a reasonable belief that the claims of
>> these applications, if granted, would apply to the draft in question.
>>
>> I will make one factual observation, without any interpretation, from
>> which you can draw your own conclusions: the independent claims of the
>> patent applications in question do not mention SDP.
>>
>> /a
>>
>>
>>
>>
>> On 7/23/13 16:58, cowwoc wrote:
>>
>>
>>     I'm a bit concerned about the optics of what just happened.
>>
>>    - The Working Group has been pushing for the use of SDP since 2011
>>    (see http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>>    - The first post related to the use of SDP in WebRTC came from
>>    Christer Holmberg of Ericsson on September 14th, 2011.
>>    - One of the Chairs of the Working Group and one of the Specification
>>    editors are from Ericsson.
>>    - There has been a substantial push against the use of SDP by some
>>    mailing list participants, but this was rejected by the Working Group.
>>    - Suddenly we find out that Ericsson has filed two patents related to
>>    the use of SDP in WebRTC and these were filed *after* Ericsson actively
>>    pushed for the use of SDP.
>>
>>     Isn't there a conflict of interest here?
>>      As a Web Developer who doesn't want/need SDP to begin with, I am
>> finding this a bitter pill to swallow. I have no problem with other people
>> using SDP (all the power to them) but, with this IPR discovery, forcing
>> their preference on me will have real-world consequences (no less than had
>> we mandated the use H264 in WebRTC).
>>
>>    1. Do the patents imply that Web Developers will have to pay patents
>>    when deploying application on top of the Browser or Native APIs?
>>    2. Is there a way to retrofit the API so those of us who do not
>>    want/need to use SDP are not forced to license this IPR? For example, the
>>    specification states that the initial offer/answer mechanism is out of
>>    scope. Could we do the same for SDP?
>>
>> Thank you,
>> Gili
>>
>>
>> _______________________________________________
>> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--e89a8f6475cf5b97c104e235dffe
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Jul 23, 2013 at 3:56 PM, Roman Shpount <span dir=3D"ltr">&l=
t;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com<=
/a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I did not want to blow this out of proportio=
n. All I wanted to point out that that licensing terms are missing and unti=
l the licensing terms are clear this does present a certain risk. I do agre=
e that given current active participation of Ericsson in this list, this ri=
sk is likely low. One other thing that I wanted to mention was that some as=
pects of the unified draft can be adjusted to limit the exposure to the dec=
lared IPR.<br>

</blockquote><div><br></div><div>Roman,</div><div><br></div><div>That seems=
 like something that would be useful if you could lay out in more detail.</=
div><div><br></div><div>Best,</div><div>-Ekr</div><div><br></div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">

Regards,<br clear=3D"all"><div>_____________<span class=3D"HOEnZb"><font co=
lor=3D"#888888"><br>Roman Shpount</font></span></div><div class=3D"HOEnZb">=
<div class=3D"h5">
<br><br><div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 6:21 PM, Adam Ro=
ach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_bl=
ank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">



 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>The chairs have asked us not to
      cross-post. As this pertains to an IETF IPR declaration, I&#39;ll
      speak to it here.<br>
      <br>
      I&#39;ll note that your response is a vast overreaction at this
      juncture, as (1) these are merely applications, not granted
      patents; and (2) Ericsson has not yet indicated their intentions
      regarding the licensing terms of any patents that may result.<br>
      <br>
      In terms of your two questions: I&#39;m not a laywer, so I&#39;m not =
able
      to speak to the applicability of the patents in any authoritative
      fashion. All that I can really say is that I have a reasonable
      belief that the claims of these applications, if granted, would
      apply to the draft in question.<br>
      <br>
      I will make one factual observation, without any interpretation,
      from which you can draw your own conclusions: the independent
      claims of the patent applications in question do not mention SDP.<spa=
n><font color=3D"#888888"><br>
      <br>
      /a</font></span><div><br>
      <br>
      <br>
      <br>
      On 7/23/13 16:58, cowwoc wrote:<br>
    </div></div>
    <blockquote type=3D"cite"><div>
     =20
      <br>
      =A0=A0=A0 I&#39;m a bit concerned about the optics of what just happe=
ned.<br>
      <ul>
        <li>The Working Group has been pushing for the use of SDP since
          2011 (see <a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/=
current/mail15.html" target=3D"_blank">http://www.ietf.org/mail-archive/web=
/rtcweb/current/mail15.html</a>).</li>
        <li>The first post related to the use of SDP in WebRTC came from
          Christer Holmberg of Ericsson on September 14th, 2011.</li>
        <li>One of the Chairs of the Working Group and one of the
          Specification editors are from Ericsson.</li>
        <li>There has been a substantial push against the use of SDP by
          some mailing list participants, but this was rejected by the
          Working Group.</li>
        <li>Suddenly we find out that Ericsson has filed two patents
          related to the use of SDP in WebRTC and these were filed
          *after* Ericsson actively pushed for the use of SDP.</li>
      </ul>
      <p>=A0=A0=A0 Isn&#39;t there a conflict of interest here?<br>
      </p>
      =A0=A0=A0 As a Web Developer who doesn&#39;t want/need SDP to begin w=
ith, I
      am finding this a bitter pill to swallow. I have no problem with
      other people using SDP (all the power to them) but, with this IPR
      discovery, forcing their preference on me will have real-world
      consequences (no less than had we mandated the use H264 in
      WebRTC).<br>
      <ol>
        <li>Do the patents imply that Web Developers will have to pay
          patents when deploying application on top of the Browser or
          Native APIs?</li>
        <li>Is there a way to retrofit the API so those of us who do not
          want/need to use SDP are not forced to license this IPR? For
          example, the specification states that the initial
          offer/answer mechanism is out of scope. Could we do the same
          for SDP?<br>
        </li>
      </ol>
      Thank you,<br>
      Gili<br>
      <br>
      <fieldset></fieldset>
      <br>
      </div><div><pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </div></blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--e89a8f6475cf5b97c104e235dffe--

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Subject: Re: [rtcweb] Unified plan IPR
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Hi Adam,

On 23/07/2013 6:21 PM, Adam Roach wrote:
> The chairs have asked us not to cross-post. As this pertains to an 
> IETF IPR declaration, I'll speak to it here.

     I cross-posted because I think we need to have a separate 
discussion regarding how to mitigate IPR risk in the API. It's good that 
you removed public-webrtc@w3c.org from the reply (I should have 
explained as much in my original post).

> I'll note that your response is a vast overreaction at this juncture, 
> as (1) these are merely applications, not granted patents; and (2) 
> Ericsson has not yet indicated their intentions regarding the 
> licensing terms of any patents that may result.

     I am not a lawyer either so I apologize if this seems like an 
overreaction. Based on my limited knowledge of software patents, I view 
this as a matter of concern. Furthermore, based on my limited 
understanding of http://www.ietf.org/rfc/rfc3979.txt I believe that IETF 
members are required to make IPR disclosures "as soon as reasonably 
possible after the Contribution is published". My understanding is that 
Ericsson was responsible to disclose this over 4 months ago (but did 
not). Again, I apologize if I am misunderstanding the document.

     Anyway, I'll wait to see what the community has to say. Hopefully 
we can come up with a reasonable solution.

Thanks,
Gili

>
> /a
>
>
>
> On 7/23/13 16:58, cowwoc wrote:
>>
>>     I'm a bit concerned about the optics of what just happened.
>>
>>   * The Working Group has been pushing for the use of SDP since 2011
>>     (see
>>     http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
>>   * The first post related to the use of SDP in WebRTC came from
>>     Christer Holmberg of Ericsson on September 14th, 2011.
>>   * One of the Chairs of the Working Group and one of the
>>     Specification editors are from Ericsson.
>>   * There has been a substantial push against the use of SDP by some
>>     mailing list participants, but this was rejected by the Working
>>     Group.
>>   * Suddenly we find out that Ericsson has filed two patents related
>>     to the use of SDP in WebRTC and these were filed *after* Ericsson
>>     actively pushed for the use of SDP.
>>
>>     Isn't there a conflict of interest here?
>>
>>     As a Web Developer who doesn't want/need SDP to begin with, I am 
>> finding this a bitter pill to swallow. I have no problem with other 
>> people using SDP (all the power to them) but, with this IPR 
>> discovery, forcing their preference on me will have real-world 
>> consequences (no less than had we mandated the use H264 in WebRTC).
>>
>>  1. Do the patents imply that Web Developers will have to pay patents
>>     when deploying application on top of the Browser or Native APIs?
>>  2. Is there a way to retrofit the API so those of us who do not
>>     want/need to use SDP are not forced to license this IPR? For
>>     example, the specification states that the initial offer/answer
>>     mechanism is out of scope. Could we do the same for SDP?
>>
>> Thank you,
>> Gili
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


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    <meta content="text/html; charset=ISO-8859-1"
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    <div class="moz-cite-prefix">Hi Adam,<br>
      <br>
      On 23/07/2013 6:21 PM, Adam Roach wrote:<br>
    </div>
    <blockquote cite="mid:51EF01E5.7090701@nostrum.com" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">The chairs have asked us not to
        cross-post. As this pertains to an IETF IPR declaration, I'll
        speak to it here.<br>
      </div>
    </blockquote>
    <br>
    &nbsp;&nbsp;&nbsp; I cross-posted because I think we need to have a separate
    discussion regarding how to mitigate IPR risk in the API. It's good
    that you removed <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3c.org">public-webrtc@w3c.org</a> from the reply (I should have
    explained as much in my original post).<br>
    <br>
    <blockquote cite="mid:51EF01E5.7090701@nostrum.com" type="cite">
      <div class="moz-cite-prefix"> I'll note that your response is a
        vast overreaction at this juncture, as (1) these are merely
        applications, not granted patents; and (2) Ericsson has not yet
        indicated their intentions regarding the licensing terms of any
        patents that may result.<br>
      </div>
    </blockquote>
    <br>
    &nbsp;&nbsp;&nbsp; I am not a lawyer either so I apologize if this seems like an
    overreaction. Based on my limited knowledge of software patents, I
    view this as a matter of concern. Furthermore, based on my limited
    understanding of <a href="http://www.ietf.org/rfc/rfc3979.txt">http://www.ietf.org/rfc/rfc3979.txt</a>
    I believe that IETF members are required to make IPR disclosures "as
    soon as reasonably possible after the Contribution is published". My
    understanding is that Ericsson was responsible to disclose this over
    4 months ago (but did not). Again, I apologize if I am
    misunderstanding the document.<br>
    <br>
    &nbsp;&nbsp;&nbsp; Anyway, I'll wait to see what the community has to say.
    Hopefully we can come up with a reasonable solution.<br>
    <br>
    Thanks,<br>
    Gili<br>
    <br>
    <blockquote cite="mid:51EF01E5.7090701@nostrum.com" type="cite">
      <div class="moz-cite-prefix"> <br>
        /a<br>
        <br>
        <br>
        <br>
        On 7/23/13 16:58, cowwoc wrote:<br>
      </div>
      <blockquote cite="mid:51EEFC6B.9090503@bbs.darktech.org"
        type="cite">
        <meta content="text/html; charset=ISO-8859-1"
          http-equiv="Content-Type">
        <br>
        &nbsp;&nbsp;&nbsp; I'm a bit concerned about the optics of what just happened.<br>
        <ul>
          <li>The Working Group has been pushing for the use of SDP
            since 2011 (see <a moz-do-not-send="true"
              href="http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html">http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html</a>).</li>
          <li>The first post related to the use of SDP in WebRTC came
            from Christer Holmberg of Ericsson on September 14th, 2011.</li>
          <li>One of the Chairs of the Working Group and one of the
            Specification editors are from Ericsson.</li>
          <li>There has been a substantial push against the use of SDP
            by some mailing list participants, but this was rejected by
            the Working Group.</li>
          <li>Suddenly we find out that Ericsson has filed two patents
            related to the use of SDP in WebRTC and these were filed
            *after* Ericsson actively pushed for the use of SDP.</li>
        </ul>
        <p>&nbsp;&nbsp;&nbsp; Isn't there a conflict of interest here?<br>
        </p>
        &nbsp;&nbsp;&nbsp; As a Web Developer who doesn't want/need SDP to begin with,
        I am finding this a bitter pill to swallow. I have no problem
        with other people using SDP (all the power to them) but, with
        this IPR discovery, forcing their preference on me will have
        real-world consequences (no less than had we mandated the use
        H264 in WebRTC).<br>
        <ol>
          <li>Do the patents imply that Web Developers will have to pay
            patents when deploying application on top of the Browser or
            Native APIs?</li>
          <li>Is there a way to retrofit the API so those of us who do
            not want/need to use SDP are not forced to license this IPR?
            For example, the specification states that the initial
            offer/answer mechanism is out of scope. Could we do the same
            for SDP?<br>
          </li>
        </ol>
        Thank you,<br>
        Gili<br>
        <br>
        <fieldset class="mimeAttachmentHeader"></fieldset>
        <br>
        <pre wrap="">_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------080309060409070206060200--

From keith.drage@alcatel-lucent.com  Tue Jul 23 17:59:37 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: cowwoc <cowwoc@bbs.darktech.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Unified plan IPR
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Date: Wed, 24 Jul 2013 00:59:23 +0000
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In regard to your questions, I suggest you go read RFC 3979 (BCP 79) and it=
s updates (as referred to by Adam) and then continue the discussion.

In regard to 1, noone in the IETF can tell you anything about the licensing=
 conditions beyond what is stated in the IPR declaration. RFC 4879 says how=
 you go about getting more clarity in the declaration if it is required. (B=
ut note the requirements of the declaration are limited).

In regard to 2, the IETF WG is expected to make its decisions based on know=
ledge of declared IPR, and the declared licensing policy.

W3C has (very) different IPR rules and you need to have that different disc=
ussion there for any requirements specified by W3C.

Regards

Keith

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 cowwoc
Sent: 23 July 2013 22:58
To: rtcweb@ietf.org
Cc: public-webrtc@w3.org
Subject: Re: [rtcweb] Unified plan IPR

On 23/07/2013 12:51 PM, Adam Roach wrote:
On 7/23/13 11:22, Roman Shpount wrote:

The situation would be a bit clearer if patent holders were to provide the =
licensing policy regarding this IPR release. Given that Ericsson is activel=
y involved in this working group, I think it would be reasonable to ask the=
m for this.

If process has been properly followed, the IPR holder has already been noti=
fied by the IETF executive director. See http://www.ietf.org/rfc/rfc4879.tx=
t (section 1 paragraph 1)

I doubt agitating for action on these mailing lists is likely to produce us=
eful results.

/a

Hi Adam,

    I'm a bit concerned about the optics of what just happened.

  *   The Working Group has been pushing for the use of SDP since 2011 (see=
 http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html).
  *   The first post related to the use of SDP in WebRTC came from Christer=
 Holmberg of Ericsson on September 14th, 2011.
  *   One of the Chairs of the Working Group and one of the Specification e=
ditors are from Ericsson.
  *   There has been a substantial push against the use of SDP by some mail=
ing list participants, but this was rejected by the Working Group.
  *   Suddenly we find out that Ericsson has filed two patents related to t=
he use of SDP in WebRTC and these were filed *after* Ericsson actively push=
ed for the use of SDP.

    Isn't there a conflict of interest here?
    As a Web Developer who doesn't want/need SDP to begin with, I am findin=
g this a bitter pill to swallow. I have no problem with other people using =
SDP (all the power to them) but, with this IPR discovery, forcing their pre=
ference on me will have real-world consequences (no less than had we mandat=
ed the use H264 in WebRTC).

  1.  Do the patents imply that Web Developers will have to pay patents whe=
n deploying application on top of the Browser or Native APIs?
  2.  Is there a way to retrofit the API so those of us who do not want/nee=
d to use SDP are not forced to license this IPR? For example, the specifica=
tion states that the initial offer/answer mechanism is out of scope. Could =
we do the same for SDP?
Thank you,
Gili

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<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">In regard to your questions, I suggest=
 you go read RFC 3979 (<st1:stockticker w:st=3D"on">BCP</st1:stockticker> 7=
9) and its updates (as
 referred to by Adam) and then continue the discussion.<o:p></o:p></span></=
font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">In regard to 1, noone in the IETF can =
tell you anything about the licensing conditions beyond what is stated in t=
he
<st1:stockticker w:st=3D"on">IPR</st1:stockticker> declaration. RFC 4879 sa=
ys how you go about getting more clarity in the declaration if it is requir=
ed. (But note the requirements of the declaration are limited).<o:p></o:p><=
/span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">In regard to 2, the IETF WG is expecte=
d to make its decisions based on knowledge of declared
<st1:stockticker w:st=3D"on">IPR</st1:stockticker>, and the declared licens=
ing policy.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">W3C has (very) different
<st1:stockticker w:st=3D"on">IPR</st1:stockticker> rules and you need to ha=
ve that different discussion there for any requirements specified by W3C.<o=
:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Regards<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Keith<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
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<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
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tyle=3D"font-size:12.0pt;
color:windowtext">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" color=3D"black" face=3D"Tahoma">=
<span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:Tahoma;color:win=
dowtext;font-weight:bold">From:</span></font></b><font size=3D"2" color=3D"=
black" face=3D"Tahoma"><span lang=3D"EN-US" style=3D"font-size:10.0pt;
font-family:Tahoma;color:windowtext">
 rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b><span style=3D=
"font-weight:bold">On Behalf Of
</span></b>cowwoc<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> 23 July 2013 22:58<br>
<b><span style=3D"font-weight:bold">To:</span></b> rtcweb@ietf.org<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> public-webrtc@w3.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [rtcweb] Unifie=
d plan <st1:stockticker w:st=3D"on">
IPR</st1:stockticker></span></font><font color=3D"black"><span lang=3D"EN-U=
S" style=3D"color:windowtext"><o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt">On 23/07/2013
<st1:time Minute=3D"51" Hour=3D"12" w:st=3D"on">12:51 PM</st1:time>, Adam R=
oach wrote:<o:p></o:p></span></font></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt" cite=3D"mid:51EE=
B495.4070404@nostrum.com" type=3D"cite">
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt">On
<st1:date Year=3D"13" Day=3D"23" Month=3D"7" ls=3D"trans" w:st=3D"on">7/23/=
13</st1:date> 11:22, Roman Shpount wrote:
<br>
<br>
<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt">The situation would be a bit clearer=
 if patent holders were to provide the licensing policy regarding this
<st1:stockticker w:st=3D"on">IPR</st1:stockticker> release. Given that Eric=
sson is actively involved in this working group, I think it would be reason=
able to ask them for this.
<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt"><br>
If process has been properly followed, the <st1:stockticker w:st=3D"on">IPR=
</st1:stockticker> holder has already been notified by the IETF executive d=
irector. See
<a href=3D"http://www.ietf.org/rfc/rfc4879.txt">http://www.ietf.org/rfc/rfc=
4879.txt</a> (section 1 paragraph 1)
<br>
<br>
I doubt agitating for action on these mailing lists is likely to produce us=
eful results.
<br>
<br>
/a <o:p></o:p></span></font></p>
</blockquote>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt"><br>
Hi Adam,<br>
<br>
&nbsp;&nbsp;&nbsp; I'm a bit concerned about the optics of what just happen=
ed.<o:p></o:p></span></font></p>
<ul type=3D"disc">
<li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-=
alt:auto;
     mso-list:l0 level1 lfo1">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">The Working Group has been pushing for the use of SDP since=
 2011 (see
<a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html"=
>http://www.ietf.org/mail-archive/web/rtcweb/current/mail15.html</a>).<o:p>=
</o:p></span></font>
</li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bo=
ttom-alt:auto;
     mso-list:l0 level1 lfo1">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">The first post related to the use of SDP in WebRTC came fro=
m Christer Holmberg of Ericsson on September 14th, 2011.<o:p></o:p></span><=
/font>
</li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bo=
ttom-alt:auto;
     mso-list:l0 level1 lfo1">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">One of the Chairs of the Working Group and one of the Speci=
fication editors are from Ericsson.<o:p></o:p></span></font>
</li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bo=
ttom-alt:auto;
     mso-list:l0 level1 lfo1">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">There has been a substantial push against the use of SDP by=
 some mailing list participants, but this was rejected by the Working Group=
.<o:p></o:p></span></font>
</li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bo=
ttom-alt:auto;
     mso-list:l0 level1 lfo1">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">Suddenly we find out that Ericsson has filed two patents re=
lated to the use of SDP in WebRTC and these were filed *after* Ericsson act=
ively pushed for the use of SDP.<o:p></o:p></span></font>
</li></ul>
<p><font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D=
"font-size:12.0pt">&nbsp;&nbsp;&nbsp; Isn't there a conflict of interest he=
re?<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt">&nbsp;&nbsp;&nbsp; As a Web Develope=
r who doesn't want/need SDP to begin with, I am finding this a bitter pill =
to swallow. I have no problem with other people using SDP
 (all the power to them) but, with this <st1:stockticker w:st=3D"on">IPR</s=
t1:stockticker> discovery, forcing their preference on me will have real-wo=
rld consequences (no less than had we mandated the use H264 in WebRTC).<o:p=
></o:p></span></font></p>
<ol start=3D"1" type=3D"1">
<li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-=
alt:auto;
     mso-list:l1 level1 lfo2">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">Do the patents imply that Web Developers will have to pay p=
atents when deploying application on top of the Browser or Native APIs?<o:p=
></o:p></span></font>
</li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bo=
ttom-alt:auto;
     mso-list:l1 level1 lfo2">
<font size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"fo=
nt-size:12.0pt">Is there a way to retrofit the
<st1:stockticker w:st=3D"on">API</st1:stockticker> so those of us who do no=
t want/need to use SDP are not forced to license this
<st1:stockticker w:st=3D"on">IPR</st1:stockticker>? For example, the specif=
ication states that the initial offer/answer mechanism is out of scope. Cou=
ld we do the same for SDP?<o:p></o:p></span></font>
</li></ol>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt">Thank you,<br>
Gili<o:p></o:p></span></font></p>
</div>
</div>
</body>
</html>

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: cowwoc <cowwoc@bbs.darktech.org>
Thread-Topic: [rtcweb] Unified plan IPR
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On Jul 23, 2013, at 4:14 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:

>> 	=95 Is there a way to retrofit the API so those of us who do not want/n=
eed to use SDP are not forced to license this IPR? For example, the specifi=
cation states that the initial offer/answer mechanism is out of scope. Coul=
d we do the same for SDP?

I think that you should have a read of the patent before deciding what the =
best mitigation is. The IETF generally does prefer to  avoid IPR where it c=
an and some people have been working towards that.=20

I'd also like to note that I far as I can tell right now, Ericsson did make=
 all the required disclosures promptly. They did disclose all this long ago=
 when they published their relevant draft.=20





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--047d7bdc119adb319c04e23c619a
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> So there are two parts to the proposal - the HTTP request for the
> credentials and interactions with WebRTC, and the stateless implementation
> that avoids the need for communication between web and TURN servers. And it
> is true that other implementations besides the suggested stateless one are
> possible, e.g. OAuth2.
>
> Would you prefer that I address the protocol and implementation parts
> separately in the document, and mention that other implementations with the
> same HTTP protocol are possible? RFC 5766 does something similar, where it
> defines the TURN protocol, but in Section 6.2 talks about how a "stateless
> stack" implementation can be used to simplify the processing on the TURN
> server.
>
> As you pointed out, the draft essentially has 2 parts - the REST API and
the stateless stack implementation which are completely independent. So
yes, separating them would help because we want other implementations to
also be compliant to the spec. Having said that, I would like opinions and
consensus from others as well.

Just to make my earlier point clearer - The credentials are generated by
TURN server and validated by TURN server. None of the other components in
the chain - web server, web application and the webrtc client, are
concerned with it (sort of like a blob). So the interoperability is not an
issue at all irrespective of what implementation approach you take.

I had some other comments earlier, which are collated below.

   1. What does the web server do with the shared (with TURN server) secret
   key? This is not clear from the text.
   2. Sec 4.2 Server - "Note that the REALM value supplied by the server is
   not meaningful in this context, and can be set to any valid value". Which
   realm value is being referred here? The REALM attribute value in the 401
   challenge response sent by the TURN server (in response to initial ALLOCATE
   request)?
   3. The 2 statements seem to be contradictory: sec 1 "However, as a relay
   service, it imposes a nontrivial cost on the service provider". sec 5.1
   "The assumption is that TURN services are of low enough value that waiting
   for the timeout to expire is a valid approach for dealing with
   possibly-compromised credentials". I am OK with the text, but would prefer
   if this could be reworded in anyway.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div dir=3D"ltr">So there are two parts to the proposal - the HTTP request =
for the credentials and interactions with WebRTC, and the stateless impleme=
ntation that avoids the need for communication between web and TURN servers=
. And it is true that other implementations besides the suggested stateless=
 one are possible, e.g. OAuth2.<br>


<div><br></div><div>Would you prefer that I address the protocol and implem=
entation parts separately in the document, and mention that other implement=
ations with the same HTTP protocol are possible? RFC 5766 does something si=
milar, where it defines the TURN protocol, but in Section 6.2 talks about h=
ow a &quot;stateless stack&quot; implementation can be used to simplify the=
 processing on the TURN server.</div>


</div><div class=3D"gmail_extra"><br></div></blockquote></div>As you pointe=
d out, the draft essentially has 2 parts - the REST API and the stateless s=
tack implementation which are completely independent. So yes, separating th=
em would help because we want other implementations to also be compliant to=
 the spec. Having said that, I would like opinions and consensus from other=
s as well.</div>
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Just to mak=
e my earlier point clearer - The credentials are generated by TURN server a=
nd validated by TURN server. None of the other components in the chain - we=
b server, web application and the webrtc client, are concerned with it (sor=
t of like a blob). So the interoperability is not an issue at all irrespect=
ive of what implementation approach you take.</div>
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">I had some =
other comments earlier, which are collated below.</div><div class=3D"gmail_=
extra"><ol><li>What does the web server do with the shared (with TURN serve=
r) secret key? This is not clear from the text.<br>
</li><li>Sec 4.2 Server - &quot;Note that the REALM value supplied by the s=
erver is not meaningful in this context, and can be set to any valid value&=
quot;. Which realm value is being referred here? The REALM attribute value =
in the 401 challenge response sent by the TURN server (in response to initi=
al ALLOCATE request)?<br>
</li><li>The 2 statements seem to be contradictory: sec 1 &quot;However, as=
 a relay service, it imposes a nontrivial cost on the service provider&quot=
;. sec 5.1 &quot;The assumption is that TURN services are of low enough val=
ue that waiting for the timeout to expire is a valid approach for dealing w=
ith possibly-compromised credentials&quot;. I am OK with the text, but woul=
d prefer if this could be reworded in anyway.</li>
</ol></div></div>

--047d7bdc119adb319c04e23c619a--

From rajmohanbanavi@gmail.com  Wed Jul 24 01:47:00 2013
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Date: Wed, 24 Jul 2013 14:16:59 +0530
Message-ID: <CAJWm+fGM1hNNnzj+LRgObKYGf=C0RXebEFpEjG4pn463NM6P+Q@mail.gmail.com>
From: Rajmohan Banavi <rajmohanbanavi@gmail.com>
To: Oleg Moskalenko <mom040267@gmail.com>
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Cc: behave <behave@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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--bcaec511e1b6e4e6be04e23df36a
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This is the draft (BEHAVE WG) I am referring to -
http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00


> This is not the case. It is not the TURN server who generates the
> credentials. The web server must generate the temporary password, and to be
> able to do that the web server must have the shared secret - the same as
> TURN server has. How they share the same shared secret I'd leave outside
> the proposed specs.
>
> OK fine.


> It is rather clear - the web server takes the shared secret and it
> generates the temporary password for long-term TURN credentials. The TURN
> server can reproduce that generation process and obtain the same temporary
> password - because the TURN server knows the same shared secret as the web
> server.
>

OK fine.

Thanks,
Rajmohan

--bcaec511e1b6e4e6be04e23df36a
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<div dir=3D"ltr"><span style=3D"font-family:arial,sans-serif;font-size:13px=
">This is the draft (BEHAVE WG) I am referring to -=A0</span><a href=3D"htt=
p://tools.ietf.org/html/draft-uberti-behave-turn-rest-00" target=3D"_blank"=
 style=3D"font-family:arial,sans-serif;font-size:13px">http://tools.ietf.or=
g/html/draft-uberti-behave-turn-rest-00</a><br style=3D"font-family:arial,s=
ans-serif;font-size:13px">
<div class=3D"gmail_extra" style=3D"font-family:arial,sans-serif;font-size:=
13px"><br><div class=3D"gmail_quote"><div class=3D"im"><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><br></div><div>This is not the case. It is not the TURN server who generat=
es the credentials. The web server must generate the temporary password, an=
d to be able to do that the web server must have the shared secret - the sa=
me as TURN server has. How they share the same shared secret I&#39;d leave =
outside the proposed specs.<br>
</div><div><br></div></div></div></div></blockquote></div><div>OK fine.</di=
v><div class=3D"im"><div>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>It is rather clear - the web server takes the shared secret and it generat=
es the temporary password for long-term TURN credentials. The TURN server c=
an reproduce that generation process and obtain the same temporary password=
 - because the TURN server knows the same shared secret as the web server.<=
br>
</div><div><div></div></div></div></div></div></blockquote><div><br></div><=
/div><div>OK fine.</div></div></div><div class=3D"gmail_extra"><br>Thanks,<=
/div><div class=3D"gmail_extra">Rajmohan</div></div>

--bcaec511e1b6e4e6be04e23df36a--

From stefan.lk.hakansson@ericsson.com  Wed Jul 24 02:22:20 2013
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Unified plan IPR
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Date: Wed, 24 Jul 2013 09:22:09 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, tim panton <thp@westhawk.co.uk>
Subject: Re: [rtcweb] Unified plan IPR
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FYI,=0A=
=0A=
I've asked internally in my company for advice on what needs to be done =0A=
(if anything) from our side. IPRs and handling of them is outside my =0A=
competence.=0A=
=0A=
Stefan=0A=
=0A=
=0A=
On 7/23/13 6:22 PM, Roman Shpount wrote:=0A=
> Changing the title and the mailing list to the more appropriate.=0A=
>=0A=
> The situation would be a bit clearer if patent holders were to provide=0A=
> the licensing policy regarding this IPR release. Given that Ericsson is=
=0A=
> actively involved in this working group, I think it would be reasonable=
=0A=
> to ask them for this.=0A=
>=0A=
> Regards,=0A=
> _____________=0A=
> Roman Shpount=0A=
>=0A=
>=0A=
> On Tue, Jul 23, 2013 at 11:51 AM, Adam Roach <adam@nostrum.com=0A=
> <mailto:adam@nostrum.com>> wrote:=0A=
>=0A=
>     On 7/23/13 09:20, tim panton wrote:=0A=
>>     On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK<stefan.lk.hakansson@=
ericsson.com>  <mailto:stefan.lk.hakansson@ericsson.com>  wrote:=0A=
>>=0A=
>>>     On 7/23/13 5:03 AM, Justin Uberti wrote:=0A=
>>>>     With the compromise reached in the Unified Plan document,=0A=
>>>     Presumablyhttp://www.ietf.org/id/draft-roach-mmusic-unified-plan-00=
.txt=0A=
>>     I'll draw folks attention to this IPR claim=0A=
>>=0A=
>>     https://datatracker.ietf.org/ipr/2141/=0A=
>>=0A=
>>     What does that mean in practice?=0A=
>>=0A=
>=0A=
>     In practice, it has very little effect.=0A=
>=0A=
>     For those of you unfamiliar with IETF IPR policy, it is documented=0A=
>     in RFC 3979:=0A=
>=0A=
>     http://www.ietf.org/rfc/rfc3979.txt=0A=
>=0A=
>     The disclosure Tim points to is made pursuant to section 6.1.3 of=0A=
>     that document. During the course of developing the unified plan, the=
=0A=
>     applications that are mentioned in that disclosure were brought to=0A=
>     my attention.=0A=
>=0A=
>     The reason it has little effect in practice is that the independent=
=0A=
>     claims would appear to cover every plan proposal to date (plan a,=0A=
>     plan b, unified plan, and even all of the "no plan" variations).=0A=
>     Consequently, it does not benefit any one approach over the others.=
=0A=
>=0A=
>     /a=0A=
>=0A=
>=0A=
=0A=

From ibc@aliax.net  Wed Jul 24 07:09:16 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 24 Jul 2013 16:08:17 +0200
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Subject: [rtcweb] About unified Plan and multiple m lines
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Hi, let's suppose WebRTC 1.0 borns with "Unified Plan" built-in, and
my browser receives a session invitation from a conference server, and
there are 8 audio participants in the conference.

So the SDP I receive has 8 "m=3Daudio" lines, maybe with different audio
codecs in each line. And following SDP O/A rules my browser will
generate SDP answer that will also contain 8 "m=3Daudio" lines.

Now, where it is supposed my browser will send my single audio track
to the conference server? What does it mean that I receive 8 "m=3Daudio"
lines while I'm just able to send a single audio track? which m=3Daudio
line should I use for sending RTP?

Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Hi, let's suppose WebRTC 1.0 borns with "Unified Plan" built-in, and
> my browser receives a session invitation from a conference server, and
> there are 8 audio participants in the conference.
>
> So the SDP I receive has 8 "m=3Daudio" lines, maybe with different audio
> codecs in each line. And following SDP O/A rules my browser will
> generate SDP answer that will also contain 8 "m=3Daudio" lines.
>
> Now, where it is supposed my browser will send my single audio track
> to the conference server? What does it mean that I receive 8 "m=3Daudio"
> lines while I'm just able to send a single audio track? which m=3Daudio
> line should I use for sending RTP?
>

One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.
The sendrecv line is the one you use to send media.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo <span dir=3D"=
ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</=
a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi, let&#39;s suppose WebRTC 1.0 borns with =
&quot;Unified Plan&quot; built-in, and<br>
my browser receives a session invitation from a conference server, and<br>
there are 8 audio participants in the conference.<br>
<br>
So the SDP I receive has 8 &quot;m=3Daudio&quot; lines, maybe with differen=
t audio<br>
codecs in each line. And following SDP O/A rules my browser will<br>
generate SDP answer that will also contain 8 &quot;m=3Daudio&quot; lines.<b=
r>
<br>
Now, where it is supposed my browser will send my single audio track<br>
to the conference server? What does it mean that I receive 8 &quot;m=3Daudi=
o&quot;<br>
lines while I&#39;m just able to send a single audio track? which m=3Daudio=
<br>
line should I use for sending RTP?<br></blockquote><div><br></div><div>One =
of them will be a=3Dsendrecv. The others will be a=3Dsendonly.</div><div>Th=
e sendrecv line is the one you use to send media.</div><div><br></div><div>

-Ekr</div><div>=A0</div></div></div></div>

--047d7b671fa678c41e04e242bd65--

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From: Ted Hardie <ted.ietf@gmail.com>
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Howdy,

Just a reminder to folks that review of the updated documents prior to the
meeting will help us make progress.  Please make sure that you have read
the unified plan document before the MMUSIC session, the data channel docs
prior to our session, and the updates to the security docs.  List
discussion on the documents is, of course, welcome now.

https://datatracker.ietf.org/doc/draft-roach-mmusic-unified-plan/

https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/

https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

Thanks,

Ted

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Howdy,<br><br>Just a reminder to folks that review of the updated documents=
 prior to the meeting will help us make progress.=A0 Please make sure that =
you have read the unified plan document before the MMUSIC session, the data=
 channel docs prior to our session, and the updates to the security docs.=
=A0 List discussion on the documents is, of course, welcome now.<br>
<br><a href=3D"https://datatracker.ietf.org/doc/draft-roach-mmusic-unified-=
plan/">https://datatracker.ietf.org/doc/draft-roach-mmusic-unified-plan/</a=
><br><pre><a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-sec=
urity-arch/">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-ar=
ch/</a><br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/">ht=
tps://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/</a><br><br><a hr=
ef=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/">htt=
ps://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/</a>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol=
/">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/</a></p=
re>Thanks,<br><br>Ted<br>

--20cf3036361f6cef7404e244a044--

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Bernard,

I apologize if I come across as being difficult here but I stil am not 
seeing the benefits.  Since the fields are not the same for the codecs, 
we will be multiplexing the bits and that seems to me to add complexity 
rather than add clarity.  Also, I can't find an IETF VP9 document for 
the payload format to reference.  If the group thinks generalization is 
the right approach I would appreciate some collaboration on getting the 
right bit definitions for the other codecs.

Regards,
Adam

On 7/23/13 12:07 PM, Bernard Aboba wrote:
> I do not think it is necessary to "support all forms of scalability 
> for all codecs".   In fact, I would make that an explicit "non-goal".  
> All that was suggested is to try to create a single extension that 
> supports a few common cases. If it is possible to handle VP8, VP9 and 
> H.264/SVC in a single extension that would be sufficient.  So why not 
> limit it to that?
>
> ------------------------------------------------------------------------
> Date: Tue, 23 Jul 2013 08:53:45 -0700
> From: fineberg@vline.me
> To: stewe@stewe.org
> CC: juberti@google.com; bernard_aboba@hotmail.com; avtext@ietf.org; 
> rtcweb@ietf.org
> Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> I've been thinking about this and given the ease at which RFC5285 
> allows for the specification of a header extension and the complexity 
> introduced by trying to generalize the header extension to support all 
> forms of scalability for all codecs that the generalization might not 
> be the best approach.  I'm not sure what we really gain by trying to 
> capture all this in a single header extension rather than one per that 
> can succinctly explain the fields without the complexity of 
> multiplexing the bits.
>
> Thoughts?
>
> Regards,
> Adam
>
> On 7/19/13 3:44 PM, Stephan Wenger wrote:
>
>     Hi,
>
>     From: Adam Fineberg <fineberg@vline.me <mailto:fineberg@vline.me>>
>     Date: Friday, 19 July, 2013 15:12
>     To: Stephan Wenger <stewe@stewe.org <mailto:stewe@stewe.org>>
>     Cc: Justin Uberti <juberti@google.com
>     <mailto:juberti@google.com>>, Bernard Aboba
>     <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>>,
>     "avtext@ietf.org <mailto:avtext@ietf.org>" <avtext@ietf.org
>     <mailto:avtext@ietf.org>>, "rtcweb@ietf.org
>     <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>     Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for
>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>     Stephan,
>
>     Thanks for the info and the reference.  I'm not sure I follow as
>     I'm not at all familiar with H.265.  I'll review the reference and
>     see what I can figure.
>
>     StW: Good luck :-)
>
>     It seems though to me that you are suggesting that except in the
>     simple case, that the data for H.265 would not be well suited to a
>     header extension, am I understanding you correctly?  There is no
>     reason the middlebox couldn't get out of band signaling of the VPS
>     as you mention but that would not be within the scope of this
>     header extension.
>
>     StW: well, if you would copy the layer_id into your header
>     extension (just as you need to do for the simple case), a really
>     smart middle box could use this information just as a decoder uses
>     it, assuming that it intercepted the VPS in the first place.
>      Insofar, I wouldn't rule out the second option on technical
>     grounds.  Whether any of the actual products would bother to do
>     that, ever, is another question.  I think the case ought to be
>     documented, though.  I can help drafting text.
>     While we are at it: doing this right could mean that you need
>     multiple specs.  First, a generic header extension mechanism
>     dedicated to side information required for pruning of RTP packet
>     streams—ideally not only for scalable video, although that is the
>     main customer today.  And second, for each "payload" (at present
>     we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 (including
>     scalable and 3D extensions, which are not yet finalized), VP8, and
>     VP9 (I know little about the latter), plus Daala, and whatnot) a
>     mapping of the bits available in the generic header extension to
>     the bits in the payload itself (NAL header and VPS in case of
>     H.265, NALU header in SVC, and the fields you mention for VP8).
>
>     Stephan
>
>
>     Any insights are appreciated.
>
>     Regards,
>     Adam
>
>     On 7/19/13 8:33 AM, Stephan Wenger wrote:
>
>         Hi,
>         I also believe that 16 bits should be enough.  For H.264 and
>         VP8 that has already been demonstrated.  For H.265, some
>         initial thoughts below.  Apologies for the word-count.
>
>         The scalable version of H265 (called SHVC) is currently under
>         development.  The current working draft can be found here:
>         http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.
>          Therein, the options for defining layering structures are
>         considerably more complex.  To start, we have 3 bits for the
>         temporal ID in the NAL unit header of the H.265 version 1
>         (HEVC) base specification (temporal scalability is already
>         nicely supported in version 1).  Just like in SVC.  In the
>         scalable extension, the NAL unit header contains a six bit
>         field that points into a data structure known as "Video
>         Parameter Set" (VPS).  Inside the VPS, those six bits are
>         mapped to to a position in a directed graph (specified through
>         "dimension_id[][]"), which tells you about the reference
>         relationship of the layer in question and its parent layer.
>          One can recursively follow the graph to determine what used
>         to be called dependency_id, quality_id, view_id, and whatnot.
>          The six bit pointer field can (or: is to be when possible)
>         organized by the encoder such that it is prudent for a middle
>         box to throw away NAL units (belonging to layers) with higher
>         values of the six bit field first, before throwing away NAL
>         units with lower values.  Relying on this feature, 3+6 bits ==
>         9 bits should be fine for the header extension.
>
>         That said, the ordering by the encoder is just a
>         recommendation, and there may well be cases where different
>         pruning strategies may be advisable.  For example, a layering
>         structure could be constructed that expands into two branches,
>         one using 2D scalable tools only, the other including view_id
>         for multi view coding.  By looking at the six bit field alone,
>         a middle box will not be able to meaningfully remove NAL units
>         belonging to one of the branches completely while pruning the
>         other branch.  In order to meaningfully deal with that
>         scenario, there would be two options: one to represent the
>         dimension_id[][] (and associated control info) in the header
>         extension, or require the middle box to have access to the VPS
>         and be able to interpret its content.  The further could take
>         considerably more than 16 bits and we would be talking about a
>         variable length data structure.  The latter requires the
>         middle box to have state and a mechanism to intercept the VPS
>         (through signaling—as the encrypted in-band VPS would not be
>         useful under the assumption that the middle box does not have
>         the key to the media—which is the motivation of the draft in
>         the first place).  I personally don't mind at all the second
>         mechanism, as I'm a big fan of out-of-band parameter set
>         transmission and any middle box must be in the signaling path
>         anyway to meaningfully manipulate RTP.  I do not like the
>         first option due to its variable, and possibly substantial,
>         overhead.
>
>         Stephan
>
>
>         From: Justin Uberti <juberti@google.com
>         <mailto:juberti@google.com>>
>         Date: Friday, 19 July, 2013 06:32
>         To: Bernard Aboba <bernard_aboba@hotmail.com
>         <mailto:bernard_aboba@hotmail.com>>
>         Cc: "avtext@ietf.org <mailto:avtext@ietf.org>"
>         <avtext@ietf.org <mailto:avtext@ietf.org>>, "rtcweb@ietf.org
>         <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org
>         <mailto:rtcweb@ietf.org>>
>         Subject: Re: [rtcweb] Fwd: New Version Notification for
>         draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>         Agree those are the right codecs to design for. Since in each
>         case there are fairly low limits on the number of supported
>         layers (i.e. 3 spatial layers for SVC), I think it should be
>         possible to pack the temporal, spatial, quality layer ids into
>         16 bits.
>
>
>         On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba
>         <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>>
>         wrote:
>
>             If we can support VP8/9 as well as H.264/5 SVC
>             that would be a start. It seems doable to me.
>
>             On Jul 18, 2013, at 8:34 PM, "Adam Fineberg"
>             <fineberg@vline.me <mailto:fineberg@vline.me>> wrote:
>
>                 Bernard,
>
>                 Are there other codecs you are thinking should be
>                 supported?  If it's generalized I would think we want
>                 to be able to cover all known scalable codecs. I'll
>                 look into the H264/SVC fields to see how to encode
>                 them in a generalized header.
>
>                 Regards,
>                 Adam
>
>                 On 7/18/13 7:40 PM, Bernard Aboba wrote:
>
>                     I think it may be possible to generalize this. 
>                     For example, for H.264/SVC which can support
>                     temporal, spatial and quality scalability, you
>                     would need the quality_id and dependency_id in
>                     addition to the temporal_id (what you call the
>                     temporal layer index).
>
>                     ------------------------------------------------------------------------
>                     Date: Thu, 18 Jul 2013 08:45:38 -0700
>                     From: fineberg@vline.me <mailto:fineberg@vline.me>
>                     To: bernard_aboba@hotmail.com
>                     <mailto:bernard_aboba@hotmail.com>
>                     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>;
>                     avtext@ietf.org <mailto:avtext@ietf.org>
>                     Subject: Re: [rtcweb] Fwd: New Version
>                     Notification for
>                     draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                     Bernard,
>
>                     Good question.  I'm not familiar enough with the
>                     parameter requirements of all other scalable
>                     codecs to be able to generalize.  If you'd like to
>                     help specify them, I'd be fine revising the draft
>                     to generalize.
>
>                     Regards,
>                     Adam
>
>                     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>
>                         Since the need is not codec specific (e.g. it
>                         arises with any codec supporting temporal,
>                         spatial and quality scalability), why
>                          a VP8-specific RTP extension?
>
>                         ------------------------------------------------------------------------
>                         Date: Wed, 17 Jul 2013 17:09:46 -0700
>                         From: fineberg@vline.me <mailto:fineberg@vline.me>
>                         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>                         Subject: [rtcweb] Fwd: New Version
>                         Notification for
>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                         Hi,
>
>                         I'm working on WebRTC services and have found
>                         that while developing services that forward
>                         VP8 video streams if we want to take advantage
>                         of the VP8 temporal scaling we must get the
>                         temporal layer information from the RTP header
>                         which requires us to decrypt the SRTP packets.
>                         This is undesirable both because the
>                         middle-box needs to have access to the keys as
>                         well as the because of the added overhead of
>                         the decrypt/encrypt cycle. This draft proposes
>                         an RTP header extension that will allow us to
>                         use the VP8 temporal layer information
>                         included in the header extension and therefore
>                         do forwarding without SRTP decryption.
>                         Comments welcome.
>
>                         Regards,
>                         Adam Fineberg
>                         fineberg at vline.com
>                         <mailto:fineberg%20at%20vline.com>
>
>                         -------- Original Message --------
>                         Subject: 	New Version Notification for
>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>                         Date: 	Tue, 09 Jul 2013 10:02:05 -0700
>                         From: 	internet-drafts at ietf.org
>                         <mailto:internet-drafts%20at%20ietf.org>
>                         To: 	Adam Fineberg<fineberg at vline.com>
>                         <mailto:fineberg%20at%20vline.com>
>
>
>
>                         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>                         has been successfully submitted by Adam Fineberg and posted to the
>                         IETF repository.
>
>                         Filename:	 draft-fineberg-avtext-temporal-layer-ext
>                         Revision:	 00
>                         Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>                         Creation date:	 2013-07-08
>                         Group:		 Individual Submission
>                         Number of pages: 6
>                         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>                         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>                         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>
>                         Abstract:
>                             This document defines a mechanism by which packets of Real-Time
>                             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>                             indicate, in an RTP header extension, the temporal layer information
>                             about the frame encoded in the RTP packet.  This information can be
>                             used in a middlebox performing bandwidth management of streams
>                             without requiring it to decrypt the streams.
>
>
>                         _______________________________________________ rtcweb
>                         mailing list rtcweb@ietf.org
>                         <mailto:rtcweb@ietf.org>
>                         https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>                     -- 
>                     Regards,
>                     Adam
>
>
>
>             _______________________________________________
>             rtcweb mailing list
>             rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>             https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>         _______________________________________________
>         avtext mailing list
>         avtext@ietf.org  <mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinfo/avtext
>
>
>     -- 
>     Regards,
>     Adam
>
>
> -- 
> Regards,
> Adam

-- 
Regards,
Adam


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    Bernard,<br>
    <br>
    I apologize if I come across as being difficult here but I stil am
    not seeing the benefits.  Since the fields are not the same for the
    codecs, we will be multiplexing the bits and that seems to me to add
    complexity rather than add clarity.  Also, I can't find an IETF VP9
    document for the payload format to reference.  If the group thinks
    generalization is the right approach I would appreciate some
    collaboration on getting the right bit definitions for the other
    codecs.<br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/23/13 12:07 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W20CACC8554C875802188A3936F0@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">I do not think it is necessary to "support all
        forms of scalability for all codecs".   In fact, I would make
        that an explicit "non-goal".  All that was suggested is to try
        to create a single extension that supports a few common cases.  
        If it is possible to handle VP8, VP9 and H.264/SVC in a single
        extension that would be sufficient.  So why not limit it to
        that? <br>
        <br>
        <div>
          <hr id="stopSpelling">Date: Tue, 23 Jul 2013 08:53:45 -0700<br>
          From: <a class="moz-txt-link-abbreviated" href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
          To: <a class="moz-txt-link-abbreviated" href="mailto:stewe@stewe.org">stewe@stewe.org</a><br>
          CC: <a class="moz-txt-link-abbreviated" href="mailto:juberti@google.com">juberti@google.com</a>; <a class="moz-txt-link-abbreviated" href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>;
          <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification
          for draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
          <br>
          I've been thinking about this and given the ease at which
          RFC5285 allows for the specification of a header extension and
          the complexity introduced by trying to generalize the header
          extension to support all forms of scalability for all codecs
          that the generalization might not be the best approach.  I'm
          not sure what we really gain by trying to capture all this in
          a single header extension rather than one per that can
          succinctly explain the fields without the complexity of
          multiplexing the bits.<br>
          <br>
          Thoughts?<br>
          <br>
          Regards,<br>
          Adam<br>
          <br>
          <div class="ecxmoz-cite-prefix">On 7/19/13 3:44 PM, Stephan
            Wenger wrote:<br>
          </div>
          <blockquote cite="mid:CE0F0BEB.9F95D%25stewe@stewe.org">
            <div style="font-family:Calibri, sans-serif;font-size:14px;"><font
                color="#0000ff">Hi,</font></div>
            <div style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);"> <br>
            </div>
            <span id="ecxOLK_SRC_BODY_SECTION"
              style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);">
              <div
                style="font-family:Calibri;font-size:11pt;text-align:left;color:black;BORDER-BOTTOM:medium
                none;BORDER-LEFT:medium
                none;PADDING-BOTTOM:0in;PADDING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df
                1pt solid;BORDER-RIGHT:medium none;PADDING-TOP:3pt;"> <span
                  style="font-weight:bold;">From: </span>Adam Fineberg
                &lt;<a moz-do-not-send="true"
                  href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
                <span style="font-weight:bold;">Date: </span>Friday, 19
                July, 2013 15:12 <br>
                <span style="font-weight:bold;">To: </span>Stephan
                Wenger &lt;<a moz-do-not-send="true"
                  href="mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
                <span style="font-weight:bold;">Cc: </span>Justin
                Uberti &lt;<a moz-do-not-send="true"
                  href="mailto:juberti@google.com">juberti@google.com</a>&gt;,

                Bernard Aboba &lt;<a moz-do-not-send="true"
                  href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,

                "<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                &lt;<a moz-do-not-send="true"
                  href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                "<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                &lt;<a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                <span style="font-weight:bold;">Subject: </span>Re:
                [avtext] [rtcweb] Fwd: New Version Notification for
                draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
              </div>
              <div><br>
              </div>
              <div>
                <div>Stephan,<br>
                  <br>
                  Thanks for the info and the reference.  I'm not sure I
                  follow as I'm not at all familiar with H.265.  I'll
                  review the reference and see what I can figure. </div>
              </div>
            </span>
            <div style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);"> <br>
            </div>
            <div style="font-family:Calibri, sans-serif;font-size:14px;"><font
                color="#0000ff">StW: Good luck :-)  </font></div>
            <div style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);"> <br>
            </div>
            <span id="ecxOLK_SRC_BODY_SECTION"
              style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);">
              <div>
                <div>It seems though to me that you are suggesting that
                  except in the simple case, that the data for H.265
                  would not be well suited to a header extension, am I
                  understanding you correctly?  There is no reason the
                  middlebox couldn't get out of band signaling of the
                  VPS as you mention but that would not be within the
                  scope of this header extension.</div>
              </div>
            </span>
            <div style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);"> <br>
            </div>
            <div><font color="#0000ff"><font face="Calibri,sans-serif">StW:

                  well, if you would copy the layer_id into your header
                  extension (just as you need to do for the simple
                  case), a really smart middle box could use this
                  information just as a decoder uses it, assuming that
                  it intercepted the VPS in the first place.  Insofar, I
                  wouldn't rule out the second option on technical
                  grounds.  Whether any of the actual products would
                  bother to do that, ever, is another question.  I think
                  the case ought to be documented, though.  I can help
                  drafting text.</font></font></div>
            <div><font color="#0000ff"><font face="Calibri,sans-serif">While

                  we are at it: doing this right could mean that you
                  need multiple specs.  First, a generic header
                  extension mechanism dedicated to side information
                  required for pruning of RTP packet streams—ideally not
                  only for scalable video, although that is the main
                  customer today.  And second, for each "payload" (at
                  present we are talking about H.264/SVC, H.265v1
                  (HEVC), H.265v2 (including scalable and 3D extensions,
                  which are not yet finalized), VP8, and VP9 (I know
                  little about the latter), plus Daala, and whatnot) a
                  mapping of the bits available in the generic header
                  extension to the bits in the payload itself (NAL
                  header and VPS in case of H.265, NALU header in SVC,
                  and the fields you mention for VP8).</font></font></div>
            <div style="font-family:Calibri, sans-serif;font-size:14px;"><br>
            </div>
            <div style="font-family:Calibri, sans-serif;font-size:14px;"><font
                color="#0000ff">Stephan</font></div>
            <span id="ecxOLK_SRC_BODY_SECTION"
              style="font-family:Calibri,
              sans-serif;font-size:14px;color:rgb(0, 0, 0);">
              <div>
                <div><br>
                  <br>
                  Any insights are appreciated.<br>
                  <br>
                  Regards,<br>
                  Adam<br>
                  <br>
                  <div class="ecxmoz-cite-prefix">On 7/19/13 8:33 AM,
                    Stephan Wenger wrote:<br>
                  </div>
                  <blockquote
                    cite="mid:CE0E9F56.9F89B%25stewe@stewe.org">
                    <div>Hi,</div>
                    <div>I also believe that 16 bits should be enough.
                       For H.264 and VP8 that has already been
                      demonstrated.  For H.265, some initial thoughts
                      below.  Apologies for the word-count.</div>
                    <div><br>
                    </div>
                    <div>The scalable version of H265 (called SHVC) is
                      currently under development.  The current working
                      draft can be found here: <a moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip"
                        target="_blank">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
                       Therein, the options for defining layering
                      structures are considerably more complex.  To
                      start, we have 3 bits for the temporal ID in the
                      NAL unit header of the H.265 version 1 (HEVC) base
                      specification (temporal scalability is already
                      nicely supported in version 1).  Just like in SVC.
                       In the scalable extension, the NAL unit header
                      contains a six bit field that points into a data
                      structure known as "Video Parameter Set" (VPS).
                       Inside the VPS, those six bits are mapped to to a
                      position in a directed graph (specified through
                      "dimension_id[][]"), which tells you about the
                      reference relationship of the layer in question
                      and its parent layer.  One can recursively follow
                      the graph to determine what used to be called
                      dependency_id, quality_id, view_id, and whatnot.
                       The six bit pointer field can (or: is to be when
                      possible) organized by the encoder such that it is
                      prudent for a middle box to throw away NAL units
                      (belonging to layers) with higher values of the
                      six bit field first, before throwing away NAL
                      units with lower values.  Relying on this feature,
                      3+6 bits == 9 bits should be fine for the header
                      extension.</div>
                    <div><br>
                    </div>
                    <div>That said, the ordering by the encoder is just
                      a recommendation, and there may well be cases
                      where different pruning strategies may be
                      advisable.  For example, a layering structure
                      could be constructed that expands into two
                      branches, one using 2D scalable tools only, the
                      other including view_id for multi view coding.  By
                      looking at the six bit field alone, a middle box
                      will not be able to meaningfully remove NAL units
                      belonging to one of the branches completely while
                      pruning the other branch.  In order to
                      meaningfully deal with that scenario, there would
                      be two options: one to represent the
                      dimension_id[][] (and associated control info) in
                      the header extension, or require the middle box to
                      have access to the VPS and be able to interpret
                      its content.  The further could take considerably
                      more than 16 bits and we would be talking about a
                      variable length data structure.  The latter
                      requires the middle box to have state and a
                      mechanism to intercept the VPS (through
                      signaling—as the encrypted in-band VPS would not
                      be useful under the assumption that the middle box
                      does not have the key to the media—which is the
                      motivation of the draft in the first place).  I
                      personally don't mind at all the second mechanism,
                      as I'm a big fan of out-of-band parameter set
                      transmission and any middle box must be in the
                      signaling path anyway to meaningfully manipulate
                      RTP.  I do not like the first option due to its
                      variable, and possibly substantial, overhead.</div>
                    <div><br>
                    </div>
                    <div>Stephan   </div>
                    <div><br>
                    </div>
                    <div><br>
                    </div>
                    <span id="ecxOLK_SRC_BODY_SECTION">
                      <div
                        style="font-family:Calibri;font-size:11pt;text-align:left;color:black;BORDER-BOTTOM:medium
                        none;BORDER-LEFT:medium
                        none;PADDING-BOTTOM:0in;PADDING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df
                        1pt solid;BORDER-RIGHT:medium
                        none;PADDING-TOP:3pt;"> <span
                          style="font-weight:bold;">From: </span>Justin
                        Uberti &lt;<a moz-do-not-send="true"
                          href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
                        <span style="font-weight:bold;">Date: </span>Friday,

                        19 July, 2013 06:32 <br>
                        <span style="font-weight:bold;">To: </span>Bernard

                        Aboba &lt;<a moz-do-not-send="true"
                          href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
                        <span style="font-weight:bold;">Cc: </span>"<a
                          moz-do-not-send="true"
                          href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                        &lt;<a moz-do-not-send="true"
                          href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,

                        "<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                        &lt;<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                        <span style="font-weight:bold;">Subject: </span>Re:

                        [rtcweb] Fwd: New Version Notification for
                        draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                      </div>
                      <div><br>
                      </div>
                      <div>
                        <div>
                          <div dir="ltr">Agree those are the right
                            codecs to design for. Since in each case
                            there are fairly low limits on the number of
                            supported layers (i.e. 3 spatial layers for
                            SVC), I think it should be possible to pack
                            the temporal, spatial, quality layer ids
                            into 16 bits.
                            <div class="ecxgmail_extra"><br>
                              <br>
                              <div class="ecxgmail_quote">On Fri, Jul
                                19, 2013 at 1:56 AM, Bernard Aboba <span
                                  dir="ltr"> &lt;<a
                                    moz-do-not-send="true"
                                    href="mailto:bernard_aboba@hotmail.com"
                                    target="_blank">bernard_aboba@hotmail.com</a>&gt;</span>
                                wrote:<br>
                                <blockquote class="ecxgmail_quote"
                                  style="border-left:1px #ccc
                                  solid;padding-left:1ex;">
                                  <div dir="auto">
                                    <div>If we can support VP8/9 as well
                                      as H.264/5 SVC</div>
                                    <div>that would be a start. It seems
                                      doable to me.</div>
                                    <div>
                                      <div>
                                        <div><br>
                                          On Jul 18, 2013, at 8:34 PM,
                                          "Adam Fineberg" &lt;<a
                                            moz-do-not-send="true"
                                            href="mailto:fineberg@vline.me"
                                            target="_blank">fineberg@vline.me</a>&gt;

                                          wrote:<br>
                                          <br>
                                        </div>
                                        <blockquote>
                                          <div>
                                            <div>Bernard,<br>
                                              <br>
                                              Are there other codecs you
                                              are thinking should be
                                              supported?  If it's
                                              generalized I would think
                                              we want to be able to
                                              cover all known scalable
                                              codecs. I'll look into the
                                              H264/SVC fields to see how
                                              to encode them in a
                                              generalized header.<br>
                                              <br>
                                              Regards,<br>
                                              Adam<br>
                                              <br>
                                              On 7/18/13 7:40 PM,
                                              Bernard Aboba wrote:<br>
                                            </div>
                                            <blockquote>
                                              <div dir="ltr">I think it
                                                may be possible to
                                                generalize this.  For
                                                example, for H.264/SVC
                                                which can support
                                                temporal, spatial and
                                                quality scalability, you
                                                would need the
                                                quality_id and
                                                dependency_id in
                                                addition to the
                                                temporal_id (what you
                                                call the temporal layer
                                                index).    <br>
                                                <br>
                                                <div>
                                                  <hr> Date: Thu, 18 Jul
                                                  2013 08:45:38 -0700<br>
                                                  From: <a
                                                    moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                  To: <a
                                                    moz-do-not-send="true"
href="mailto:bernard_aboba@hotmail.com" target="_blank">
                                                    bernard_aboba@hotmail.com</a><br>
                                                  CC: <a
                                                    moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>; <a
                                                    moz-do-not-send="true"
href="mailto:avtext@ietf.org" target="_blank">avtext@ietf.org</a><br>
                                                  Subject: Re: [rtcweb]
                                                  Fwd: New Version
                                                  Notification for
draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                  <br>
                                                  Bernard,<br>
                                                  <br>
                                                  Good question.  I'm
                                                  not familiar enough
                                                  with the parameter
                                                  requirements of all
                                                  other scalable codecs
                                                  to be able to
                                                  generalize.  If you'd
                                                  like to help specify
                                                  them, I'd be fine
                                                  revising the draft to
                                                  generalize.<br>
                                                  <br>
                                                  Regards,<br>
                                                  Adam<br>
                                                  <br>
                                                  <div>On 7/17/13 8:26
                                                    PM, Bernard Aboba
                                                    wrote:<br>
                                                  </div>
                                                  <blockquote>
                                                    <div dir="ltr">Since
                                                      the need is not
                                                      codec specific
                                                      (e.g. it arises
                                                      with any codec
                                                      supporting
                                                      temporal, spatial
                                                      and quality
                                                      scalability), why
                                                      <br>
                                                       a VP8-specific
                                                      RTP extension? <br>
                                                       <br>
                                                      <div>
                                                        <hr> Date: Wed,
                                                        17 Jul 2013
                                                        17:09:46 -0700<br>
                                                        From: <a
                                                          moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                        To: <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                        Subject:
                                                        [rtcweb] Fwd:
                                                        New Version
                                                        Notification for
draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                        <br>
                                                        <span
                                                          style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline
!important;word-spacing:0px;">Hi,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <span
                                                          style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline
!important;word-spacing:0px;">I'm working on WebRTC services and have
                                                          found that
                                                          while
                                                          developing
                                                          services that
                                                          forward VP8
                                                          video streams
                                                          if we want to
                                                          take advantage
                                                          of the VP8
                                                          temporal
                                                          scaling we
                                                          must get the
                                                          temporal layer
                                                          information
                                                          from the RTP
                                                          header which
                                                          requires us to
                                                          decrypt the
                                                          SRTP packets.
                                                          This is
                                                          undesirable
                                                          both because
                                                          the middle-box
                                                          needs to have
                                                          access to the
                                                          keys as well
                                                          as the because
                                                          of the added
                                                          overhead of
                                                          the
                                                          decrypt/encrypt
                                                          cycle. This
                                                          draft proposes
                                                          an RTP header
                                                          extension that
                                                          will allow us
                                                          to use the VP8
                                                          temporal layer
                                                          information
                                                          included in
                                                          the header
                                                          extension and
                                                          therefore do
                                                          forwarding
                                                          without SRTP
                                                          decryption.
                                                          Comments
                                                          welcome.</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <span
                                                          style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline
!important;word-spacing:0px;">Regards,</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <span
                                                          style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;display:inline
!important;word-spacing:0px;">Adam Fineberg</span><br
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;">
                                                        <div
style="text-indent:0px;letter-spacing:normal;text-transform:none;white-space:normal;word-spacing:0px;"><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          rel="nofollow"
target="_blank">fineberg at vline.com</a><br>
                                                          <br>
                                                          --------
                                                          Original
                                                          Message
                                                          --------
                                                          <table
                                                          border="0"
                                                          cellpadding="0"
cellspacing="0">
                                                          <tbody>
                                                          <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Subject:</th>
                                                          <td>New
                                                          Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt</td>
                                                          </tr>
                                                          <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">Date:</th>
                                                          <td>Tue, 09
                                                          Jul 2013
                                                          10:02:05 -0700</td>
                                                          </tr>
                                                          <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">From:</th>
                                                          <td><a
                                                          moz-do-not-send="true"
href="mailto:internet-drafts%20at%20ietf.org" rel="nofollow"
                                                          target="_blank">internet-drafts

                                                          at ietf.org</a></td>
                                                          </tr>
                                                          <tr>
                                                          <th
                                                          align="RIGHT"
nowrap="nowrap" valign="BASELINE">To:</th>
                                                          <td>Adam
                                                          Fineberg<span> </span><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          rel="nofollow"
target="_blank">&lt;fineberg at vline.com&gt;</a></td>
                                                          </tr>
                                                          </tbody>
                                                          </table>
                                                          <br>
                                                          <br>
                                                          <pre style="width:939.59px;white-space:pre-wrap;">A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" rel="nofollow" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a>
Status:          <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" rel="nofollow" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a>
Htmlized:        <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" rel="nofollow" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a>


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.
</pre>
                                                        </div>
                                                        <br>
                                                        _______________________________________________

                                                        rtcweb mailing
                                                        list <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank"> rtcweb@ietf.org</a> <a
                                                          moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
                                                    </div>
                                                  </blockquote>
                                                  <br>
                                                  <pre>-- 
Regards,
Adam</pre>
                                                </div>
                                              </div>
                                            </blockquote>
                                            <br>
                                          </div>
                                        </blockquote>
                                      </div>
                                    </div>
                                  </div>
                                  <br>
_______________________________________________<br>
                                  rtcweb mailing list<br>
                                  <a moz-do-not-send="true"
                                    href="mailto:rtcweb@ietf.org"
                                    target="_blank">rtcweb@ietf.org</a><br>
                                  <a moz-do-not-send="true"
                                    href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                    target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                                  <br>
                                </blockquote>
                              </div>
                              <br>
                            </div>
                          </div>
                        </div>
                      </div>
                    </span><br>
                    <fieldset class="ecxmimeAttachmentHeader"></fieldset>
                    <br>
                    <pre>_______________________________________________
avtext mailing list
<a moz-do-not-send="true" class="ecxmoz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send="true" class="ecxmoz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/avtext" target="_blank">https://www.ietf.org/mailman/listinfo/avtext</a></pre>
                  </blockquote>
                  <br>
                  <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
                </div>
              </div>
            </span> </blockquote>
          <br>
          <pre class="ecxmoz-signature">-- 
Regards,
Adam</pre>
        </div>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
  </body>
</html>

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From: Oleg Moskalenko <mom040267@gmail.com>
To: Rajmohan Banavi <rajmohanbanavi@gmail.com>
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--047d7b5d9ff7df488e04e23c9d1f
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Please see my comments in the text:


On Tue, Jul 23, 2013 at 11:54 PM, Rajmohan Banavi
<rajmohanbanavi@gmail.com>wrote:

>
> Just to make my earlier point clearer - The credentials are generated by
> TURN server and validated by TURN server. None of the other components in
> the chain - web server, web application and the webrtc client, are
> concerned with it (sort of like a blob). So the interoperability is not an
> issue at all irrespective of what implementation approach you take.
>

This is not the case. It is not the TURN server who generates the
credentials. The web server must generate the temporary password, and to be
able to do that the web server must have the shared secret - the same as
TURN server has. How they share the same shared secret I'd leave outside
the proposed specs.



>
> I had some other comments earlier, which are collated below.
>
>    1. What does the web server do with the shared (with TURN server)
>    secret key? This is not clear from the text.
>
> It is rather clear - the web server takes the shared secret and it
generates the temporary password for long-term TURN credentials. The TURN
server can reproduce that generation process and obtain the same temporary
password - because the TURN server knows the same shared secret as the web
server.


>
>    1.
>    2. Sec 4.2 Server - "Note that the REALM value supplied by the server
>    is not meaningful in this context, and can be set to any valid value".
>    Which realm value is being referred here? The REALM attribute value in the
>    401 challenge response sent by the TURN server (in response to initial
>    ALLOCATE request)?
>
>
I do not see those words in the draft text on the Internet:

http://tools.ietf.org/html/draft-uberti-rtcweb-turn-rest-00

is there a newer version somewhere ?


>
>    1.
>    2. The 2 statements seem to be contradictory: sec 1 "However, as a
>    relay service, it imposes a nontrivial cost on the service provider". sec
>    5.1 "The assumption is that TURN services are of low enough value that
>    waiting for the timeout to expire is a valid approach for dealing with
>    possibly-compromised credentials". I am OK with the text, but would prefer
>    if this could be reworded in anyway.
>
>
I'd really like to have an access to the fresh draft,it seems like it was
not posted publicly - I cannot find it.

Thanks
Oleg

--047d7b5d9ff7df488e04e23c9d1f
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Please see my comments in the text:<br><div class=3D"gmail=
_extra"><br><br><div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 11:54 PM=
, Rajmohan Banavi <span dir=3D"ltr">&lt;<a href=3D"mailto:rajmohanbanavi@gm=
ail.com" target=3D"_blank">rajmohanbanavi@gmail.com</a>&gt;</span> wrote:<b=
r>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><br>Just=
 to make my earlier point clearer - The credentials are generated by TURN s=
erver and validated by TURN server. None of the other components in the cha=
in - web server, web application and the webrtc client, are concerned with =
it (sort of like a blob). So the interoperability is not an issue at all ir=
respective of what implementation approach you take.
</div></blockquote><div><br></div><div>This is not the case. It is not the =
TURN server who generates the credentials. The web server must generate the=
 temporary password, and to be able to do that the web server must have the=
 shared secret - the same as TURN server has. How they share the same share=
d secret I&#39;d leave outside the proposed specs.<br>
</div><div><br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
div dir=3D"ltr"><div class=3D"gmail_extra"><br></div><div class=3D"gmail_ex=
tra">I had some other comments earlier, which are collated below.</div>
<div class=3D"gmail_extra"><ol><li>What does the web server do with the sha=
red (with TURN server) secret key? This is not clear from the text.<br></li=
></ol></div></div></blockquote><div>It is rather clear - the web server tak=
es the shared secret and it generates the temporary password for long-term =
TURN credentials. The TURN server can reproduce that generation process and=
 obtain the same temporary password - because the TURN server knows the sam=
e shared secret as the web server.<br>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div =
dir=3D"ltr"><div class=3D"gmail_extra"><ol><li>
</li><li>Sec 4.2 Server - &quot;Note that the REALM value supplied by the s=
erver is not meaningful in this context, and can be set to any valid value&=
quot;. Which realm value is being referred here? The REALM attribute value =
in the 401 challenge response sent by the TURN server (in response to initi=
al ALLOCATE request)?<br>
</li></ol></div></div></blockquote><div><br></div><div>I do not see those w=
ords in the draft text on the Internet:<br><br><a href=3D"http://tools.ietf=
.org/html/draft-uberti-rtcweb-turn-rest-00">http://tools.ietf.org/html/draf=
t-uberti-rtcweb-turn-rest-00</a><br>
<br></div><div>is there a newer version somewhere ?<br></div><div>=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div cla=
ss=3D"gmail_extra">
<ol><li>
</li><li>The 2 statements seem to be contradictory: sec 1 &quot;However, as=
 a relay service, it imposes a nontrivial cost on the service provider&quot=
;. sec 5.1 &quot;The assumption is that TURN services are of low enough val=
ue that waiting for the timeout to expire is a valid approach for dealing w=
ith possibly-compromised credentials&quot;. I am OK with the text, but woul=
d prefer if this could be reworded in anyway.</li>

</ol></div></div>
</blockquote></div><br></div><div class=3D"gmail_extra">I&#39;d really like=
 to have an access to the fresh draft,it seems like it was not posted publi=
cly - I cannot find it.<br><br>Thanks<br>Oleg<br><br></div></div>

--047d7b5d9ff7df488e04e23c9d1f--

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Date: Wed, 24 Jul 2013 07:06:27 -0700
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From: Oleg Moskalenko <mom040267@gmail.com>
To: Rajmohan Banavi <rajmohanbanavi@gmail.com>
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Subject: Re: [rtcweb] [BEHAVE] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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Thank you for the new link.

I checked the new version of the draft and I personally see no problem in
the text. There is no proprietary software requirements in the draft. It
simply defines the logic how the TURN server and web server can organize
the temporary password generation, without imposing any proprietary
requirements and specs on the software. It is mentioning a possible
communication channel between web server and TURN server without defining
any specs and as it is written that channel is not required. As it is
written, I do not think that it has to be separated into two pieces - it is
a single solid logical functionality definition.

Thanks
Oleg


On Wed, Jul 24, 2013 at 1:46 AM, Rajmohan Banavi
<rajmohanbanavi@gmail.com>wrote:

> This is the draft (BEHAVE WG) I am referring to -
> http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
>
>
>> This is not the case. It is not the TURN server who generates the
>> credentials. The web server must generate the temporary password, and to be
>> able to do that the web server must have the shared secret - the same as
>> TURN server has. How they share the same shared secret I'd leave outside
>> the proposed specs.
>>
>> OK fine.
>
>
>> It is rather clear - the web server takes the shared secret and it
>> generates the temporary password for long-term TURN credentials. The TURN
>> server can reproduce that generation process and obtain the same temporary
>> password - because the TURN server knows the same shared secret as the web
>> server.
>>
>
> OK fine.
>
> Thanks,
> Rajmohan
>

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<div dir=3D"ltr"><div><div>Thank you for the new link. <br><br>I checked th=
e new version of the draft and I personally see no problem in the text. The=
re is no proprietary software requirements in the draft. It simply defines =
the logic how the TURN server and web server can organize the temporary pas=
sword generation, without imposing any proprietary requirements and specs o=
n the software. It is mentioning a possible communication channel between w=
eb server and TURN server without defining any specs and as it is written t=
hat channel is not required. As it is written, I do not think that it has t=
o be separated into two pieces - it is a single solid logical functionality=
 definition.<br>
<br></div>Thanks<br></div>Oleg <br><div class=3D"gmail_extra"><br><br><div =
class=3D"gmail_quote">On Wed, Jul 24, 2013 at 1:46 AM, Rajmohan Banavi <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:rajmohanbanavi@gmail.com" target=3D"_bl=
ank">rajmohanbanavi@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span style=3D"font-family:=
arial,sans-serif;font-size:13px">This is the draft (BEHAVE WG) I am referri=
ng to -=A0</span><a href=3D"http://tools.ietf.org/html/draft-uberti-behave-=
turn-rest-00" style=3D"font-family:arial,sans-serif;font-size:13px" target=
=3D"_blank">http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00</a>=
<br style=3D"font-family:arial,sans-serif;font-size:13px">

<div class=3D"gmail_extra" style=3D"font-family:arial,sans-serif;font-size:=
13px"><br><div class=3D"gmail_quote"><div class=3D"im"><div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><br></div><div>This is not the case. It is not the TURN server who generat=
es the credentials. The web server must generate the temporary password, an=
d to be able to do that the web server must have the shared secret - the sa=
me as TURN server has. How they share the same shared secret I&#39;d leave =
outside the proposed specs.<br>

</div><div><br></div></div></div></div></blockquote></div></div><div>OK fin=
e.</div><div class=3D"im"><div><div>=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>It is rather clear - the web server takes the shared secret and it generat=
es the temporary password for long-term TURN credentials. The TURN server c=
an reproduce that generation process and obtain the same temporary password=
 - because the TURN server knows the same shared secret as the web server.<=
br>

</div><div><div></div></div></div></div></div></blockquote><div><br></div><=
/div></div><div>OK fine.</div></div></div><div class=3D"gmail_extra"><br>Th=
anks,</div><div class=3D"gmail_extra">Rajmohan</div></div>
</blockquote></div><br></div></div>

--047d7b6d7a6466913a04e2426a0e--

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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Adam Fineberg <fineberg@vline.me>, Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [avtext] [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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If the group is to specify something generic, naturally it should be generi=
c enough to cover at least all existing standard scalable video codecs if p=
ossible. And I personally think that is possible and not difficult at all. =
Thus, why limit to only a few scalable video codecs?

I could provide some help here too if needed.

BR, YK

From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf Of=
 Adam Fineberg
Sent: Wednesday, July 24, 2013 10:08 AM
To: Bernard Aboba
Cc: avtext@ietf.org; rtcweb@ietf.org; Justin Uberti
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Bernard,

I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.  Since the fields are not the same for the codecs, we will=
 be multiplexing the bits and that seems to me to add complexity rather tha=
n add clarity.  Also, I can't find an IETF VP9 document for the payload for=
mat to reference.  If the group thinks generalization is the right approach=
 I would appreciate some collaboration on getting the right bit definitions=
 for the other codecs.

Regards,
Adam
On 7/23/13 12:07 PM, Bernard Aboba wrote:
I do not think it is necessary to "support all forms of scalability for all=
 codecs".   In fact, I would make that an explicit "non-goal".  All that wa=
s suggested is to try to create a single extension that supports a few comm=
on cases.   If it is possible to handle VP8, VP9 and H.264/SVC in a single =
extension that would be sufficient.  So why not limit it to that?
________________________________
Date: Tue, 23 Jul 2013 08:53:45 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: stewe@stewe.org<mailto:stewe@stewe.org>
CC: juberti@google.com<mailto:juberti@google.com>; bernard_aboba@hotmail.co=
m<mailto:bernard_aboba@hotmail.com>; avtext@ietf.org<mailto:avtext@ietf.org=
>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization might not be the best approach.  =
I'm not sure what we really gain by trying to capture all this in a single =
header extension rather than one per that can succinctly explain the fields=
 without the complexity of multiplexing the bits.

Thoughts?

Regards,
Adam
On 7/19/13 3:44 PM, Stephan Wenger wrote:
Hi,

From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Friday, 19 July, 2013 15:12
To: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Cc: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, Bernard =
Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>>, "avtex=
t@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtext@ietf.org=
>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not at=
 all familiar with H.265.  I'll review the reference and see what I can fig=
ure.

StW: Good luck :-)

It seems though to me that you are suggesting that except in the simple cas=
e, that the data for H.265 would not be well suited to a header extension, =
am I understanding you correctly?  There is no reason the middlebox couldn'=
t get out of band signaling of the VPS as you mention but that would not be=
 within the scope of this header extension.

StW: well, if you would copy the layer_id into your header extension (just =
as you need to do for the simple case), a really smart middle box could use=
 this information just as a decoder uses it, assuming that it intercepted t=
he VPS in the first place.  Insofar, I wouldn't rule out the second option =
on technical grounds.  Whether any of the actual products would bother to d=
o that, ever, is another question.  I think the case ought to be documented=
, though.  I can help drafting text.
While we are at it: doing this right could mean that you need multiple spec=
s.  First, a generic header extension mechanism dedicated to side informati=
on required for pruning of RTP packet streams-ideally not only for scalable=
 video, although that is the main customer today.  And second, for each "pa=
yload" (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 =
(including scalable and 3D extensions, which are not yet finalized), VP8, a=
nd VP9 (I know little about the latter), plus Daala, and whatnot) a mapping=
 of the bits available in the generic header extension to the bits in the p=
ayload itself (NAL header and VPS in case of H.265, NALU header in SVC, and=
 the fields you mention for VP8).

Stephan


Any insights are appreciated.

Regards,
Adam
On 7/19/13 8:33 AM, Stephan Wenger wrote:
Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling-as the encrypted in-band VPS would not be useful under the assump=
tion that the middle box does not have the key to the media-which is the mo=
tivation of the draft in the first place).  I personally don't mind at all =
the second mechanism, as I'm a big fan of out-of-band parameter set transmi=
ssion and any middle box must be in the signaling path anyway to meaningful=
ly manipulate RTP.  I do not like the first option due to its variable, and=
 possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.

On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:
Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).
________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam
On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:

New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.tx=
t

Date:

Tue, 09 Jul 2013 10:02:05 -0700

From:

internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>

To:

Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.com>





A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt

has been successfully submitted by Adam Fineberg and posted to the

IETF repository.



Filename:         draft-fineberg-avtext-temporal-layer-ext

Revision:         00

Title:            A Real-Time Transport Protocol (RTP) Header Extension for=
 VP8 Temporal Layer Information

Creation date:    2013-07-08

Group:            Individual Submission

Number of pages: 6

URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt

Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext

Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00





Abstract:

   This document defines a mechanism by which packets of Real-Time

   Tranport Protocol (RTP) video streams encoded with the VP8 codec can

   indicate, in an RTP header extension, the temporal layer information

   about the frame encoded in the RTP packet.  This information can be

   used in a middlebox performing bandwidth management of streams

   without requiring it to decrypt the streams.

_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--

Regards,

Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb




_______________________________________________

avtext mailing list

avtext@ietf.org<mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinf=
o/avtext


--

Regards,

Adam


--

Regards,

Adam



--

Regards,

Adam

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</head>
<body bgcolor=3D"white" lang=3D"EN-CA" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">If the group is to specif=
y something generic, naturally it should be generic enough to cover at leas=
t all existing standard scalable video codecs if possible.
 And I personally think that is possible and not difficult at all. Thus, wh=
y limit to only a few scalable video codecs?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I could provide some help=
 here too if needed.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR, YK<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> avtext-bounces@ietf=
.org [mailto:avtext-bounces@ietf.org]
<b>On Behalf Of </b>Adam Fineberg<br>
<b>Sent:</b> Wednesday, July 24, 2013 10:08 AM<br>
<b>To:</b> Bernard Aboba<br>
<b>Cc:</b> avtext@ietf.org; rtcweb@ietf.org; Justin Uberti<br>
<b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Bernard,<br>
<br>
I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.&nbsp; Since the fields are not the same for the codecs, we=
 will be multiplexing the bits and that seems to me to add complexity rathe=
r than add clarity.&nbsp; Also, I can't find
 an IETF VP9 document for the payload format to reference.&nbsp; If the gro=
up thinks generalization is the right approach I would appreciate some coll=
aboration on getting the right bit definitions for the other codecs.<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/23/13 12:07 PM, Bernard Aboba wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I do not think it is =
necessary to &quot;support all forms of scalability for all codecs&quot;.&n=
bsp;&nbsp; In fact, I would make that an explicit &quot;non-goal&quot;.&nbs=
p; All that was suggested is to try to create a single extension that suppo=
rts
 a few common cases.&nbsp;&nbsp; If it is possible to handle VP8, VP9 and H=
.264/SVC in a single extension that would be sufficient.&nbsp; So why not l=
imit it to that?
<o:p></o:p></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center" id=3D"stopSpelling">
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Date: Tue, 23 Jul 201=
3 08:53:45 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me">fineberg@vline.me</a><br>
To: <a href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a><br>
CC: <a href=3D"mailto:juberti@google.com">juberti@google.com</a>; <a href=
=3D"mailto:bernard_aboba@hotmail.com">
bernard_aboba@hotmail.com</a>; <a href=3D"mailto:avtext@ietf.org">avtext@ie=
tf.org</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt<br>
<br>
I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization
 might not be the best approach.&nbsp; I'm not sure what we really gain by =
trying to capture all this in a single header extension rather than one per=
 that can succinctly explain the fields without the complexity of multiplex=
ing the bits.<br>
<br>
Thoughts?<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/19/13 3:44 PM, Stephan Wenger wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">Hi,</span><span style=3D"fon=
t-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>=
</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;">Adam Fineberg &lt;<a href=3D"mailto:fineberg@vline.=
me">fineberg@vline.me</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 15:12 <br>
<b>To: </b>Stephan Wenger &lt;<a href=3D"mailto:stewe@stewe.org">stewe@stew=
e.org</a>&gt;<br>
<b>Cc: </b>Justin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@=
google.com</a>&gt;, Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotma=
il.com">bernard_aboba@hotmail.com</a>&gt;, &quot;<a href=3D"mailto:avtext@i=
etf.org">avtext@ietf.org</a>&quot; &lt;<a href=3D"mailto:avtext@ietf.org">a=
vtext@ietf.org</a>&gt;,
 &quot;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Stephan,<br>
<br>
Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm n=
ot at all familiar with H.265.&nbsp; I'll review the reference and see what=
 I can figure.&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">StW: Good luck :-) &nbsp;</s=
pan><span style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">It seems though to me that you are sugg=
esting that except in the simple case, that the data for H.265 would not be=
 well suited to a header extension, am I understanding you
 correctly?&nbsp; There is no reason the middlebox couldn't get out of band=
 signaling of the VPS as you mention but that would not be within the scope=
 of this header extension.<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:blue">StW: well, if you would copy the layer_id int=
o your header extension (just as you need to do for the simple case), a rea=
lly smart middle box could use this information just as
 a decoder uses it,&nbsp;assuming&nbsp;that it intercepted the VPS in the f=
irst place. &nbsp;Insofar, I wouldn't rule out the second option on technic=
al grounds. &nbsp;Whether any of the actual products would bother to do tha=
t, ever, is another question. &nbsp;I think the case ought
 to be documented, though. &nbsp;I can help drafting text.</span><o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:blue">While we are at it: doing this right could me=
an that you need multiple specs. &nbsp;First, a generic header extension me=
chanism dedicated to side information required for pruning of
 RTP packet streams&#8212;ideally not only for scalable video, although tha=
t is the main customer today. &nbsp;And second, for each &quot;payload&quot=
; (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 (incl=
uding scalable and 3D extensions, which are not yet finalized),
 VP8, and VP9 (I know little about the latter), plus Daala, and whatnot) a =
mapping of the bits available in the generic header extension to the bits i=
n the payload itself (NAL header and VPS in case of H.265, NALU header in S=
VC, and the fields you mention for
 VP8).</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">Stephan</span><span style=3D=
"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><=
o:p></o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
<br>
Any insights are appreciated.<br>
<br>
Regards,<br>
Adam<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On 7/19/13 8:33 AM, Stephan Wenger wrot=
e:<o:p></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">I also believe that 16 bits should be e=
nough. &nbsp;For H.264 and VP8 that has already been demonstrated. &nbsp;Fo=
r H.265, some initial thoughts below. &nbsp;Apologies for the word-count.<o=
:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">The scalable version of H265 (called SH=
VC) is currently under development. &nbsp;The current working draft can be =
found here:&nbsp;<a href=3D"http://phenix.int-evry.fr/jct/doc_end_user/docu=
ments/13_Incheon/wg11/JCTVC-M1008-v3.zip" target=3D"_blank">http://phenix.i=
nt-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a=
>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">That said, the ordering by the encoder =
is just a recommendation, and there may well be cases where different pruni=
ng strategies may be advisable. &nbsp;For example, a layering
 structure could be constructed that expands into two branches, one using 2=
D scalable tools only, the other including view_id for multi view coding. &=
nbsp;By looking at the six bit field alone, a middle box will not be able t=
o meaningfully remove NAL units belonging
 to one of the branches completely while pruning the other branch. &nbsp;In=
 order to meaningfully deal with that scenario, there would be two options:=
 one to represent the dimension_id[][] (and associated control info) in the=
 header extension, or require the middle
 box to have access to the VPS and be able to interpret its content. &nbsp;=
The further could take considerably more than 16 bits and we would be talki=
ng about a variable length data structure. &nbsp;The latter requires the mi=
ddle box to have state and a mechanism to
 intercept the VPS (through signaling&#8212;as the encrypted in-band VPS wo=
uld not be useful under the assumption that the middle box does not have th=
e key to the media&#8212;which is the motivation of the draft in the first =
place). &nbsp;I personally don't mind at all the
 second mechanism, as I'm a big fan of out-of-band parameter set transmissi=
on and any middle box must be in the signaling path anyway to meaningfully =
manipulate RTP. &nbsp;I do not like the first option due to its variable, a=
nd possibly substantial, overhead.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Stephan &nbsp;&nbsp;<o:p></o:p></span><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;">Justin Uberti &lt;<a href=3D"mailto:juberti@google.=
com">juberti@google.com</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 06:32 <br>
<b>To: </b>Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.com">b=
ernard_aboba@hotmail.com</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quo=
t; &lt;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, &quot;<a=
 href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"m=
ailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [rtcweb] Fwd: New Version Notification for draft-finebe=
rg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Agree those are the right codecs to des=
ign for. Since in each case there are fairly low limits on the number of su=
pported layers (i.e. 3 spatial layers for SVC), I think
 it should be possible to pack the temporal, spatial, quality layer ids int=
o 16 bits.
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nb=
sp;</o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On Fri, Jul 19, 2013 at 1:56 AM, Bernar=
d Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">=
bernard_aboba@hotmail.com</a>&gt; wrote:<o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If we can support VP8/9 as well as H.26=
4/5 SVC<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">that would be a start. It seems doable =
to me.<o:p></o:p></span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a href=3D"mailt=
o:fineberg@vline.me" target=3D"_blank">fineberg@vline.me</a>&gt; wrote:<o:p=
></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:<o:p></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">I think =
it may be possible to generalize this.&nbsp; For example, for H.264/SVC whi=
ch can support temporal, spatial and quality scalability, you would
 need the quality_id and dependency_id in addition to the temporal_id (what=
 you call the temporal layer index). &nbsp;&nbsp;
<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Date: Th=
u, 18 Jul 2013 08:45:38 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_=
aboba@hotmail.com</a><br>
CC: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
>; <a href=3D"mailto:avtext@ietf.org" target=3D"_blank">
avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On 7/17/13 8:26 PM, Bernard Aboba wrote=
:<o:p></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Since the need is not codec specific (e=
.g. it arises with any codec supporting temporal, spatial and quality scala=
bility), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Date: Wed, 17 Jul 2013 17:09:46 -0700<b=
r>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
Hi,<br>
<br>
I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt
 the SRTP packets. This is undesirable both because the middle-box needs to=
 have access to the keys as well as the because of the added overhead of th=
e decrypt/encrypt cycle. This draft proposes an RTP header extension that w=
ill allow us to use the VP8 temporal
 layer information included in the header extension and therefore do forwar=
ding without SRTP decryption. Comments welcome.<br>
<br>
Regards,<br>
Adam Fineberg<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><a href=3D"mailto:fineberg%20at%20vline=
.com" target=3D"_blank">fineberg at vline.com</a><br>
<br>
-------- Original Message -------- <o:p></o:p></span></p>
<table class=3D"MsoNormalTable" border=3D"0" cellspacing=3D"0" cellpadding=
=3D"0">
<tbody>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Subjec=
t:<o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Date:<=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">Tue, 09 Jul 2013 10:02:05 -0700<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>From:<=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal"><a href=3D"mailto:internet-drafts%20at%20ietf.org" t=
arget=3D"_blank">internet-drafts at ietf.org</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>To:<o:=
p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">Adam Fineberg&nbsp;<a href=3D"mailto:fineberg%20at%2=
0vline.com" target=3D"_blank">&lt;fineberg at vline.com&gt;</a><o:p></o:p><=
/p>
</td>
</tr>
</tbody>
</table>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
<br>
<br>
<o:p></o:p></span></p>
<pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<=
o:p></o:p></pre>
<pre>has been successfully submitted by Adam Fineberg and posted to the<o:p=
></o:p></pre>
<pre>IETF repository.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  draft-fineberg-av=
text-temporal-layer-ext<o:p></o:p></pre>
<pre>Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  00<o:p></o:p></pr=
e>
<pre>Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  A =
Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer =
Information<o:p></o:p></pre>
<pre>Creation date:&nbsp;&nbsp;  2013-07-08<o:p></o:p></pre>
<pre>Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  In=
dividual Submission<o:p></o:p></pre>
<pre>Number of pages: 6<o:p></o:p></pre>
<pre>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &nbsp;<a href=3D"http://www.ietf.org/internet-drafts/draft-fineberg-avtext=
-temporal-layer-ext-00.txt" target=3D"_blank">http://www.ietf.org/internet-=
drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a><o:p></o:p></pre>
<pre>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a href=
=3D"http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ex=
t" target=3D"_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-=
temporal-layer-ext</a><o:p></o:p></pre>
<pre>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a href=3D"http://=
tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target=3D"=
_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext=
-00</a><o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Abstract:<o:p></o:p></pre>
<pre>&nbsp;&nbsp; This document defines a mechanism by which packets of Rea=
l-Time<o:p></o:p></pre>
<pre>&nbsp;&nbsp; Tranport Protocol (RTP) video streams encoded with the VP=
8 codec can<o:p></o:p></pre>
<pre>&nbsp;&nbsp; indicate, in an RTP header extension, the temporal layer =
information<o:p></o:p></pre>
<pre>&nbsp;&nbsp; about the frame encoded in the RTP packet.&nbsp; This inf=
ormation can be<o:p></o:p></pre>
<pre>&nbsp;&nbsp; used in a middlebox performing bandwidth management of st=
reams<o:p></o:p></pre>
<pre>&nbsp;&nbsp; without requiring it to decrypt the streams.<o:p></o:p></=
pre>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
_______________________________________________ rtcweb mailing list <a href=
=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb=
" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</blockquote>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
<br>
<o:p></o:p></span></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>avtext mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><a href=3D"https=
://www.ietf.org/mailman/listinfo/avtext" target=3D"_blank">https://www.ietf=
.org/mailman/listinfo/avtext</a><o:p></o:p></pre>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</body>
</html>

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YK,

I would appreciate your collaboration.  Which codecs are you referring 
to when you say "all existing standard scalable video codecs"?

Regards,
Adam

On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:
>
> If the group is to specify something generic, naturally it should be 
> generic enough to cover at least all existing standard scalable video 
> codecs if possible. And I personally think that is possible and not 
> difficult at all. Thus, why limit to only a few scalable video codecs?
>
> I could provide some help here too if needed.
>
> BR, YK
>
> *From:*avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] *On 
> Behalf Of *Adam Fineberg
> *Sent:* Wednesday, July 24, 2013 10:08 AM
> *To:* Bernard Aboba
> *Cc:* avtext@ietf.org; rtcweb@ietf.org; Justin Uberti
> *Subject:* Re: [avtext] [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Bernard,
>
> I apologize if I come across as being difficult here but I stil am not 
> seeing the benefits.  Since the fields are not the same for the 
> codecs, we will be multiplexing the bits and that seems to me to add 
> complexity rather than add clarity.  Also, I can't find an IETF VP9 
> document for the payload format to reference.  If the group thinks 
> generalization is the right approach I would appreciate some 
> collaboration on getting the right bit definitions for the other codecs.
>
> Regards,
> Adam
>
> On 7/23/13 12:07 PM, Bernard Aboba wrote:
>
>     I do not think it is necessary to "support all forms of
>     scalability for all codecs".   In fact, I would make that an
>     explicit "non-goal".  All that was suggested is to try to create a
>     single extension that supports a few common cases.   If it is
>     possible to handle VP8, VP9 and H.264/SVC in a single extension
>     that would be sufficient.  So why not limit it to that?
>
>     ------------------------------------------------------------------------
>
>     Date: Tue, 23 Jul 2013 08:53:45 -0700
>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>     To: stewe@stewe.org <mailto:stewe@stewe.org>
>     CC: juberti@google.com <mailto:juberti@google.com>;
>     bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>;
>     avtext@ietf.org <mailto:avtext@ietf.org>; rtcweb@ietf.org
>     <mailto:rtcweb@ietf.org>
>     Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for
>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>     I've been thinking about this and given the ease at which RFC5285
>     allows for the specification of a header extension and the
>     complexity introduced by trying to generalize the header extension
>     to support all forms of scalability for all codecs that the
>     generalization might not be the best approach.  I'm not sure what
>     we really gain by trying to capture all this in a single header
>     extension rather than one per that can succinctly explain the
>     fields without the complexity of multiplexing the bits.
>
>     Thoughts?
>
>     Regards,
>     Adam
>
>     On 7/19/13 3:44 PM, Stephan Wenger wrote:
>
>         Hi,
>
>         *From: *Adam Fineberg <fineberg@vline.me
>         <mailto:fineberg@vline.me>>
>         *Date: *Friday, 19 July, 2013 15:12
>         *To: *Stephan Wenger <stewe@stewe.org <mailto:stewe@stewe.org>>
>         *Cc: *Justin Uberti <juberti@google.com
>         <mailto:juberti@google.com>>, Bernard Aboba
>         <bernard_aboba@hotmail.com
>         <mailto:bernard_aboba@hotmail.com>>, "avtext@ietf.org
>         <mailto:avtext@ietf.org>" <avtext@ietf.org
>         <mailto:avtext@ietf.org>>, "rtcweb@ietf.org
>         <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org
>         <mailto:rtcweb@ietf.org>>
>         *Subject: *Re: [avtext] [rtcweb] Fwd: New Version Notification
>         for draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>         Stephan,
>
>         Thanks for the info and the reference.  I'm not sure I follow
>         as I'm not at all familiar with H.265.  I'll review the
>         reference and see what I can figure.
>
>         StW: Good luck :-)
>
>         It seems though to me that you are suggesting that except in
>         the simple case, that the data for H.265 would not be well
>         suited to a header extension, am I understanding you
>         correctly? There is no reason the middlebox couldn't get out
>         of band signaling of the VPS as you mention but that would not
>         be within the scope of this header extension.
>
>         StW: well, if you would copy the layer_id into your header
>         extension (just as you need to do for the simple case), a
>         really smart middle box could use this information just as a
>         decoder uses it, assuming that it intercepted the VPS in the
>         first place.  Insofar, I wouldn't rule out the second option
>         on technical grounds.  Whether any of the actual products
>         would bother to do that, ever, is another question.  I think
>         the case ought to be documented, though.  I can help drafting
>         text.
>
>         While we are at it: doing this right could mean that you need
>         multiple specs.  First, a generic header extension mechanism
>         dedicated to side information required for pruning of RTP
>         packet streams---ideally not only for scalable video, although
>         that is the main customer today.  And second, for each
>         "payload" (at present we are talking about H.264/SVC, H.265v1
>         (HEVC), H.265v2 (including scalable and 3D extensions, which
>         are not yet finalized), VP8, and VP9 (I know little about the
>         latter), plus Daala, and whatnot) a mapping of the bits
>         available in the generic header extension to the bits in the
>         payload itself (NAL header and VPS in case of H.265, NALU
>         header in SVC, and the fields you mention for VP8).
>
>         Stephan
>
>
>
>         Any insights are appreciated.
>
>         Regards,
>         Adam
>
>         On 7/19/13 8:33 AM, Stephan Wenger wrote:
>
>             Hi,
>
>             I also believe that 16 bits should be enough.  For H.264
>             and VP8 that has already been demonstrated.  For H.265,
>             some initial thoughts below.  Apologies for the word-count.
>
>             The scalable version of H265 (called SHVC) is currently
>             under development.  The current working draft can be found
>             here:
>             http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.
>              Therein, the options for defining layering structures are
>             considerably more complex.  To start, we have 3 bits for
>             the temporal ID in the NAL unit header of the H.265
>             version 1 (HEVC) base specification (temporal scalability
>             is already nicely supported in version 1).  Just like in
>             SVC.  In the scalable extension, the NAL unit header
>             contains a six bit field that points into a data structure
>             known as "Video Parameter Set" (VPS).  Inside the VPS,
>             those six bits are mapped to to a position in a directed
>             graph (specified through "dimension_id[][]"), which tells
>             you about the reference relationship of the layer in
>             question and its parent layer.  One can recursively follow
>             the graph to determine what used to be called
>             dependency_id, quality_id, view_id, and whatnot.  The six
>             bit pointer field can (or: is to be when possible)
>             organized by the encoder such that it is prudent for a
>             middle box to throw away NAL units (belonging to layers)
>             with higher values of the six bit field first, before
>             throwing away NAL units with lower values.  Relying on
>             this feature, 3+6 bits == 9 bits should be fine for the
>             header extension.
>
>             That said, the ordering by the encoder is just a
>             recommendation, and there may well be cases where
>             different pruning strategies may be advisable.  For
>             example, a layering structure could be constructed that
>             expands into two branches, one using 2D scalable tools
>             only, the other including view_id for multi view coding.
>              By looking at the six bit field alone, a middle box will
>             not be able to meaningfully remove NAL units belonging to
>             one of the branches completely while pruning the other
>             branch.  In order to meaningfully deal with that scenario,
>             there would be two options: one to represent the
>             dimension_id[][] (and associated control info) in the
>             header extension, or require the middle box to have access
>             to the VPS and be able to interpret its content.  The
>             further could take considerably more than 16 bits and we
>             would be talking about a variable length data structure.
>              The latter requires the middle box to have state and a
>             mechanism to intercept the VPS (through signaling---as the
>             encrypted in-band VPS would not be useful under the
>             assumption that the middle box does not have the key to
>             the media---which is the motivation of the draft in the
>             first place).  I personally don't mind at all the second
>             mechanism, as I'm a big fan of out-of-band parameter set
>             transmission and any middle box must be in the signaling
>             path anyway to meaningfully manipulate RTP.  I do not like
>             the first option due to its variable, and possibly
>             substantial, overhead.
>
>             Stephan
>
>             *From: *Justin Uberti <juberti@google.com
>             <mailto:juberti@google.com>>
>             *Date: *Friday, 19 July, 2013 06:32
>             *To: *Bernard Aboba <bernard_aboba@hotmail.com
>             <mailto:bernard_aboba@hotmail.com>>
>             *Cc: *"avtext@ietf.org <mailto:avtext@ietf.org>"
>             <avtext@ietf.org <mailto:avtext@ietf.org>>,
>             "rtcweb@ietf.org <mailto:rtcweb@ietf.org>"
>             <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>             *Subject: *Re: [rtcweb] Fwd: New Version Notification for
>             draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>             Agree those are the right codecs to design for. Since in
>             each case there are fairly low limits on the number of
>             supported layers (i.e. 3 spatial layers for SVC), I think
>             it should be possible to pack the temporal, spatial,
>             quality layer ids into 16 bits.
>
>             On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba
>             <bernard_aboba@hotmail.com
>             <mailto:bernard_aboba@hotmail.com>> wrote:
>
>             If we can support VP8/9 as well as H.264/5 SVC
>
>             that would be a start. It seems doable to me.
>
>
>             On Jul 18, 2013, at 8:34 PM, "Adam Fineberg"
>             <fineberg@vline.me <mailto:fineberg@vline.me>> wrote:
>
>                 Bernard,
>
>                 Are there other codecs you are thinking should be
>                 supported? If it's generalized I would think we want
>                 to be able to cover all known scalable codecs. I'll
>                 look into the H264/SVC fields to see how to encode
>                 them in a generalized header.
>
>                 Regards,
>                 Adam
>
>                 On 7/18/13 7:40 PM, Bernard Aboba wrote:
>
>                     I think it may be possible to generalize this. For
>                     example, for H.264/SVC which can support temporal,
>                     spatial and quality scalability, you would need
>                     the quality_id and dependency_id in addition to
>                     the temporal_id (what you call the temporal layer
>                     index).
>
>                     ------------------------------------------------------------------------
>
>                     Date: Thu, 18 Jul 2013 08:45:38 -0700
>                     From: fineberg@vline.me <mailto:fineberg@vline.me>
>                     To: bernard_aboba@hotmail.com
>                     <mailto:bernard_aboba@hotmail.com>
>                     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>;
>                     avtext@ietf.org <mailto:avtext@ietf.org>
>                     Subject: Re: [rtcweb] Fwd: New Version
>                     Notification for
>                     draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                     Bernard,
>
>                     Good question. I'm not familiar enough with the
>                     parameter requirements of all other scalable
>                     codecs to be able to generalize.  If you'd like to
>                     help specify them, I'd be fine revising the draft
>                     to generalize.
>
>                     Regards,
>                     Adam
>
>                     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>
>                         Since the need is not codec specific (e.g. it
>                         arises with any codec supporting temporal,
>                         spatial and quality scalability), why
>                          a VP8-specific RTP extension?
>
>                         ------------------------------------------------------------------------
>
>                         Date: Wed, 17 Jul 2013 17:09:46 -0700
>                         From: fineberg@vline.me <mailto:fineberg@vline.me>
>                         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>                         Subject: [rtcweb] Fwd: New Version
>                         Notification for
>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                         Hi,
>
>                         I'm working on WebRTC services and have found
>                         that while developing services that forward
>                         VP8 video streams if we want to take advantage
>                         of the VP8 temporal scaling we must get the
>                         temporal layer information from the RTP header
>                         which requires us to decrypt the SRTP packets.
>                         This is undesirable both because the
>                         middle-box needs to have access to the keys as
>                         well as the because of the added overhead of
>                         the decrypt/encrypt cycle. This draft proposes
>                         an RTP header extension that will allow us to
>                         use the VP8 temporal layer information
>                         included in the header extension and therefore
>                         do forwarding without SRTP decryption.
>                         Comments welcome.
>
>                         Regards,
>                         Adam Fineberg
>
>                         fineberg at vline.com
>                         <mailto:fineberg%20at%20vline.com>
>
>                         -------- Original Message --------
>
>                         *Subject:*
>
>                         	
>
>                         New Version Notification for
>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                         *Date:*
>
>                         	
>
>                         Tue, 09 Jul 2013 10:02:05 -0700
>
>                         *From:*
>
>                         	
>
>                         internet-drafts at ietf.org
>                         <mailto:internet-drafts%20at%20ietf.org>
>
>                         *To:*
>
>                         	
>
>                         Adam Fineberg <fineberg at vline.com>
>                         <mailto:fineberg%20at%20vline.com>
>
>
>
>
>                         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                         has been successfully submitted by Adam Fineberg and posted to the
>
>                         IETF repository.
>
>                           
>
>                         Filename:         draft-fineberg-avtext-temporal-layer-ext
>
>                         Revision:         00
>
>                         Title:            A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>
>                         Creation date:    2013-07-08
>
>                         Group:            Individual Submission
>
>                         Number of pages: 6
>
>                         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>
>                         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>
>                         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>
>                           
>
>                           
>
>                         Abstract:
>
>                             This document defines a mechanism by which packets of Real-Time
>
>                             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>
>                             indicate, in an RTP header extension, the temporal layer information
>
>                             about the frame encoded in the RTP packet.  This information can be
>
>                             used in a middlebox performing bandwidth management of streams
>
>                             without requiring it to decrypt the streams.
>
>
>                         _______________________________________________ rtcweb
>                         mailing list rtcweb@ietf.org
>                         <mailto:rtcweb@ietf.org>
>                         https://www.ietf.org/mailman/listinfo/rtcweb
>
>                     -- 
>
>                     Regards,
>
>                     Adam
>
>
>             _______________________________________________
>             rtcweb mailing list
>             rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>             https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>             _______________________________________________
>
>             avtext mailing list
>
>             avtext@ietf.org  <mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinfo/avtext
>
>         -- 
>
>         Regards,
>
>         Adam
>
>     -- 
>
>     Regards,
>
>     Adam
>
>
>
> -- 
> Regards,
> Adam

-- 
Regards,
Adam


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    YK,<br>
    <br>
    I would appreciate your collaboration.&nbsp; Which codecs are you
    referring to when you say "all existing standard scalable video
    codecs"? <br>
    <br>
    Regards,<br>
    Adam<br>
    <br>
    <div class="moz-cite-prefix">On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:<br>
    </div>
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If
            the group is to specify something generic, naturally it
            should be generic enough to cover at least all existing
            standard scalable video codecs if possible. And I personally
            think that is possible and not difficult at all. Thus, why
            limit to only a few scalable video codecs?<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            could provide some help here too if needed.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR,
            YK<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <div>
            <div style="border:none;border-top:solid #B5C4DF
              1.0pt;padding:3.0pt 0in 0in 0in">
              <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                    lang="EN-US">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                  lang="EN-US"> <a class="moz-txt-link-abbreviated" href="mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:avtext-bounces@ietf.org">mailto:avtext-bounces@ietf.org</a>]
                  <b>On Behalf Of </b>Adam Fineberg<br>
                  <b>Sent:</b> Wednesday, July 24, 2013 10:08 AM<br>
                  <b>To:</b> Bernard Aboba<br>
                  <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; Justin
                  Uberti<br>
                  <b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New Version
                  Notification for
                  draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt">Bernard,<br>
            <br>
            I apologize if I come across as being difficult here but I
            stil am not seeing the benefits.&nbsp; Since the fields are not
            the same for the codecs, we will be multiplexing the bits
            and that seems to me to add complexity rather than add
            clarity.&nbsp; Also, I can't find an IETF VP9 document for the
            payload format to reference.&nbsp; If the group thinks
            generalization is the right approach I would appreciate some
            collaboration on getting the right bit definitions for the
            other codecs.<br>
            <br>
            Regards,<br>
            Adam<o:p></o:p></p>
          <div>
            <p class="MsoNormal">On 7/23/13 12:07 PM, Bernard Aboba
              wrote:<o:p></o:p></p>
          </div>
          <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
            <div>
              <p class="MsoNormal" style="margin-bottom:12.0pt">I do not
                think it is necessary to "support all forms of
                scalability for all codecs".&nbsp;&nbsp; In fact, I would make
                that an explicit "non-goal".&nbsp; All that was suggested is
                to try to create a single extension that supports a few
                common cases.&nbsp;&nbsp; If it is possible to handle VP8, VP9 and
                H.264/SVC in a single extension that would be
                sufficient.&nbsp; So why not limit it to that?
                <o:p></o:p></p>
              <div>
                <div class="MsoNormal" style="text-align:center"
                  align="center">
                  <hr id="stopSpelling" align="center" size="2"
                    width="100%">
                </div>
                <p class="MsoNormal" style="margin-bottom:12.0pt">Date:
                  Tue, 23 Jul 2013 08:53:45 -0700<br>
                  From: <a moz-do-not-send="true"
                    href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                  To: <a moz-do-not-send="true"
                    href="mailto:stewe@stewe.org">stewe@stewe.org</a><br>
                  CC: <a moz-do-not-send="true"
                    href="mailto:juberti@google.com">juberti@google.com</a>;
                  <a moz-do-not-send="true"
                    href="mailto:bernard_aboba@hotmail.com">
                    bernard_aboba@hotmail.com</a>; <a
                    moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>;
                  <a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  Subject: Re: [avtext] [rtcweb] Fwd: New Version
                  Notification for
                  draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                  <br>
                  I've been thinking about this and given the ease at
                  which RFC5285 allows for the specification of a header
                  extension and the complexity introduced by trying to
                  generalize the header extension to support all forms
                  of scalability for all codecs that the generalization
                  might not be the best approach.&nbsp; I'm not sure what we
                  really gain by trying to capture all this in a single
                  header extension rather than one per that can
                  succinctly explain the fields without the complexity
                  of multiplexing the bits.<br>
                  <br>
                  Thoughts?<br>
                  <br>
                  Regards,<br>
                  Adam<o:p></o:p></p>
                <div>
                  <p class="MsoNormal">On 7/19/13 3:44 PM, Stephan
                    Wenger wrote:<o:p></o:p></p>
                </div>
                <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:blue">Hi,</span><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div style="border:none;border-top:solid #B5C4DF
                    1.0pt;padding:3.0pt 0in 0in 0in">
                    <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:
                        </span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Adam
                        Fineberg &lt;<a moz-do-not-send="true"
                          href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
                        <b>Date: </b>Friday, 19 July, 2013 15:12 <br>
                        <b>To: </b>Stephan Wenger &lt;<a
                          moz-do-not-send="true"
                          href="mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
                        <b>Cc: </b>Justin Uberti &lt;<a
                          moz-do-not-send="true"
                          href="mailto:juberti@google.com">juberti@google.com</a>&gt;,
                        Bernard Aboba &lt;<a moz-do-not-send="true"
                          href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,
                        "<a moz-do-not-send="true"
                          href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                        &lt;<a moz-do-not-send="true"
                          href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,

                        "<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                        &lt;<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                        <b>Subject: </b>Re: [avtext] [rtcweb] Fwd: New
                        Version Notification for
                        draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div>
                    <div>
                      <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Stephan,<br>
                          <br>
                          Thanks for the info and the reference.&nbsp; I'm
                          not sure I follow as I'm not at all familiar
                          with H.265.&nbsp; I'll review the reference and see
                          what I can figure.&nbsp;<o:p></o:p></span></p>
                    </div>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:blue">StW:
                        Good luck :-) &nbsp;</span><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div>
                    <div>
                      <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">It
                          seems though to me that you are suggesting
                          that except in the simple case, that the data
                          for H.265 would not be well suited to a header
                          extension, am I understanding you correctly?&nbsp;
                          There is no reason the middlebox couldn't get
                          out of band signaling of the VPS as you
                          mention but that would not be within the scope
                          of this header extension.<o:p></o:p></span></p>
                    </div>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:blue">StW:
                        well, if you would copy the layer_id into your
                        header extension (just as you need to do for the
                        simple case), a really smart middle box could
                        use this information just as a decoder uses
                        it,&nbsp;assuming&nbsp;that it intercepted the VPS in the
                        first place. &nbsp;Insofar, I wouldn't rule out the
                        second option on technical grounds. &nbsp;Whether any
                        of the actual products would bother to do that,
                        ever, is another question. &nbsp;I think the case
                        ought to be documented, though. &nbsp;I can help
                        drafting text.</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:blue">While
                        we are at it: doing this right could mean that
                        you need multiple specs. &nbsp;First, a generic
                        header extension mechanism dedicated to side
                        information required for pruning of RTP packet
                        streams&#8212;ideally not only for scalable video,
                        although that is the main customer today. &nbsp;And
                        second, for each "payload" (at present we are
                        talking about H.264/SVC, H.265v1 (HEVC), H.265v2
                        (including scalable and 3D extensions, which are
                        not yet finalized), VP8, and VP9 (I know little
                        about the latter), plus Daala, and whatnot) a
                        mapping of the bits available in the generic
                        header extension to the bits in the payload
                        itself (NAL header and VPS in case of H.265,
                        NALU header in SVC, and the fields you mention
                        for VP8).</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:blue">Stephan</span><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
                  </div>
                  <div>
                    <div>
                      <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
                          <br>
                          Any insights are appreciated.<br>
                          <br>
                          Regards,<br>
                          Adam<o:p></o:p></span></p>
                      <div>
                        <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">On
                            7/19/13 8:33 AM, Stephan Wenger wrote:<o:p></o:p></span></p>
                      </div>
                      <blockquote
                        style="margin-top:5.0pt;margin-bottom:5.0pt">
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Hi,<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">I
                              also believe that 16 bits should be
                              enough. &nbsp;For H.264 and VP8 that has
                              already been demonstrated. &nbsp;For H.265,
                              some initial thoughts below. &nbsp;Apologies
                              for the word-count.<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">The
                              scalable version of H265 (called SHVC) is
                              currently under development. &nbsp;The current
                              working draft can be found here:&nbsp;<a
                                moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip"
                                target="_blank">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
                              &nbsp;Therein, the options for defining
                              layering structures are considerably more
                              complex. &nbsp;To start, we have 3 bits for the
                              temporal ID in the NAL unit header of the
                              H.265 version 1 (HEVC) base specification
                              (temporal scalability is already nicely
                              supported in version 1). &nbsp;Just like in
                              SVC. &nbsp;In the scalable extension, the NAL
                              unit header contains a six bit field that
                              points into a data structure known as
                              "Video Parameter Set" (VPS). &nbsp;Inside the
                              VPS, those six bits are mapped to to a
                              position in a directed graph (specified
                              through "dimension_id[][]"), which tells
                              you about the reference relationship of
                              the layer in question and its parent
                              layer. &nbsp;One can recursively follow the
                              graph to determine what used to be called
                              dependency_id, quality_id, view_id, and
                              whatnot. &nbsp;The six bit pointer field can
                              (or: is to be when possible) organized by
                              the encoder such that it is prudent for a
                              middle box to throw away NAL units
                              (belonging to layers) with higher values
                              of the six bit field first, before
                              throwing away NAL units with lower values.
                              &nbsp;Relying on this feature, 3+6 bits == 9
                              bits should be fine for the header
                              extension.<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">That
                              said, the ordering by the encoder is just
                              a recommendation, and there may well be
                              cases where different pruning strategies
                              may be advisable. &nbsp;For example, a layering
                              structure could be constructed that
                              expands into two branches, one using 2D
                              scalable tools only, the other including
                              view_id for multi view coding. &nbsp;By looking
                              at the six bit field alone, a middle box
                              will not be able to meaningfully remove
                              NAL units belonging to one of the branches
                              completely while pruning the other branch.
                              &nbsp;In order to meaningfully deal with that
                              scenario, there would be two options: one
                              to represent the dimension_id[][] (and
                              associated control info) in the header
                              extension, or require the middle box to
                              have access to the VPS and be able to
                              interpret its content. &nbsp;The further could
                              take considerably more than 16 bits and we
                              would be talking about a variable length
                              data structure. &nbsp;The latter requires the
                              middle box to have state and a mechanism
                              to intercept the VPS (through signaling&#8212;as
                              the encrypted in-band VPS would not be
                              useful under the assumption that the
                              middle box does not have the key to the
                              media&#8212;which is the motivation of the draft
                              in the first place). &nbsp;I personally don't
                              mind at all the second mechanism, as I'm a
                              big fan of out-of-band parameter set
                              transmission and any middle box must be in
                              the signaling path anyway to meaningfully
                              manipulate RTP. &nbsp;I do not like the first
                              option due to its variable, and possibly
                              substantial, overhead.<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Stephan
                              &nbsp;&nbsp;<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div style="border:none;border-top:solid #B5C4DF
                          1.0pt;padding:3.0pt 0in 0in 0in">
                          <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:
                              </span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Justin
                              Uberti &lt;<a moz-do-not-send="true"
                                href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
                              <b>Date: </b>Friday, 19 July, 2013 06:32
                              <br>
                              <b>To: </b>Bernard Aboba &lt;<a
                                moz-do-not-send="true"
                                href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
                              <b>Cc: </b>"<a moz-do-not-send="true"
                                href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                              &lt;<a moz-do-not-send="true"
                                href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                              "<a moz-do-not-send="true"
                                href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                              &lt;<a moz-do-not-send="true"
                                href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                              <b>Subject: </b>Re: [rtcweb] Fwd: New
                              Version Notification for
                              draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                        </div>
                        <div>
                          <div>
                            <div>
                              <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Agree
                                  those are the right codecs to design
                                  for. Since in each case there are
                                  fairly low limits on the number of
                                  supported layers (i.e. 3 spatial
                                  layers for SVC), I think it should be
                                  possible to pack the temporal,
                                  spatial, quality layer ids into 16
                                  bits.
                                  <o:p></o:p></span></p>
                              <div>
                                <p class="MsoNormal"
                                  style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                                <div>
                                  <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">On
                                      Fri, Jul 19, 2013 at 1:56 AM,
                                      Bernard Aboba &lt;<a
                                        moz-do-not-send="true"
                                        href="mailto:bernard_aboba@hotmail.com"
                                        target="_blank">bernard_aboba@hotmail.com</a>&gt;
                                      wrote:<o:p></o:p></span></p>
                                  <div>
                                    <div>
                                      <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">If
                                          we can support VP8/9 as well
                                          as H.264/5 SVC<o:p></o:p></span></p>
                                    </div>
                                    <div>
                                      <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">that
                                          would be a start. It seems
                                          doable to me.<o:p></o:p></span></p>
                                    </div>
                                    <div>
                                      <div>
                                        <div>
                                          <p class="MsoNormal"
                                            style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
                                              On Jul 18, 2013, at 8:34
                                              PM, "Adam Fineberg" &lt;<a
                                                moz-do-not-send="true"
                                                href="mailto:fineberg@vline.me"
                                                target="_blank">fineberg@vline.me</a>&gt;
                                              wrote:<o:p></o:p></span></p>
                                        </div>
                                        <blockquote
                                          style="margin-top:5.0pt;margin-bottom:5.0pt">
                                          <div>
                                            <div>
                                              <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Bernard,<br>
                                                  <br>
                                                  Are there other codecs
                                                  you are thinking
                                                  should be supported?&nbsp;
                                                  If it's generalized I
                                                  would think we want to
                                                  be able to cover all
                                                  known scalable codecs.
                                                  I'll look into the
                                                  H264/SVC fields to see
                                                  how to encode them in
                                                  a generalized header.<br>
                                                  <br>
                                                  Regards,<br>
                                                  Adam<br>
                                                  <br>
                                                  On 7/18/13 7:40 PM,
                                                  Bernard Aboba wrote:<o:p></o:p></span></p>
                                            </div>
                                            <blockquote
                                              style="margin-top:5.0pt;margin-bottom:5.0pt">
                                              <div>
                                                <p class="MsoNormal"
                                                  style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">I
                                                    think it may be
                                                    possible to
                                                    generalize this.&nbsp;
                                                    For example, for
                                                    H.264/SVC which can
                                                    support temporal,
                                                    spatial and quality
                                                    scalability, you
                                                    would need the
                                                    quality_id and
                                                    dependency_id in
                                                    addition to the
                                                    temporal_id (what
                                                    you call the
                                                    temporal layer
                                                    index). &nbsp;&nbsp;
                                                    <o:p></o:p></span></p>
                                                <div>
                                                  <div class="MsoNormal"
style="text-align:center" align="center"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">
                                                      <hr align="center"
                                                        size="2"
                                                        width="100%">
                                                    </span></div>
                                                  <p class="MsoNormal"
                                                    style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Date:
                                                      Thu, 18 Jul 2013
                                                      08:45:38 -0700<br>
                                                      From: <a
                                                        moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                      To: <a
                                                        moz-do-not-send="true"
href="mailto:bernard_aboba@hotmail.com" target="_blank">bernard_aboba@hotmail.com</a><br>
                                                      CC: <a
                                                        moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>; <a
                                                        moz-do-not-send="true"
href="mailto:avtext@ietf.org" target="_blank">
                                                        avtext@ietf.org</a><br>
                                                      Subject: Re:
                                                      [rtcweb] Fwd: New
                                                      Version
                                                      Notification for
                                                      draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                      <br>
                                                      Bernard,<br>
                                                      <br>
                                                      Good question.&nbsp;
                                                      I'm not familiar
                                                      enough with the
                                                      parameter
                                                      requirements of
                                                      all other scalable
                                                      codecs to be able
                                                      to generalize.&nbsp; If
                                                      you'd like to help
                                                      specify them, I'd
                                                      be fine revising
                                                      the draft to
                                                      generalize.<br>
                                                      <br>
                                                      Regards,<br>
                                                      Adam<o:p></o:p></span></p>
                                                  <div>
                                                    <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">On
                                                        7/17/13 8:26 PM,
                                                        Bernard Aboba
                                                        wrote:<o:p></o:p></span></p>
                                                  </div>
                                                  <blockquote
                                                    style="margin-top:5.0pt;margin-bottom:5.0pt">
                                                    <div>
                                                      <p
                                                        class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Since
                                                          the need is
                                                          not codec
                                                          specific (e.g.
                                                          it arises with
                                                          any codec
                                                          supporting
                                                          temporal,
                                                          spatial and
                                                          quality
                                                          scalability),
                                                          why
                                                          <br>
                                                          &nbsp;a
                                                          VP8-specific
                                                          RTP extension?
                                                          <br>
                                                          &nbsp;<o:p></o:p></span></p>
                                                      <div>
                                                        <div
                                                          class="MsoNormal"
style="text-align:center" align="center"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">
                                                          <hr
                                                          align="center"
                                                          size="2"
                                                          width="100%">
                                                          </span></div>
                                                        <p
                                                          class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Date:
                                                          Wed, 17 Jul
                                                          2013 17:09:46
                                                          -0700<br>
                                                          From: <a
                                                          moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                          To: <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                          Subject:
                                                          [rtcweb] Fwd:
                                                          New Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                          <br>
                                                          Hi,<br>
                                                          <br>
                                                          I'm working on
                                                          WebRTC
                                                          services and
                                                          have found
                                                          that while
                                                          developing
                                                          services that
                                                          forward VP8
                                                          video streams
                                                          if we want to
                                                          take advantage
                                                          of the VP8
                                                          temporal
                                                          scaling we
                                                          must get the
                                                          temporal layer
                                                          information
                                                          from the RTP
                                                          header which
                                                          requires us to
                                                          decrypt the
                                                          SRTP packets.
                                                          This is
                                                          undesirable
                                                          both because
                                                          the middle-box
                                                          needs to have
                                                          access to the
                                                          keys as well
                                                          as the because
                                                          of the added
                                                          overhead of
                                                          the
                                                          decrypt/encrypt
                                                          cycle. This
                                                          draft proposes
                                                          an RTP header
                                                          extension that
                                                          will allow us
                                                          to use the VP8
                                                          temporal layer
                                                          information
                                                          included in
                                                          the header
                                                          extension and
                                                          therefore do
                                                          forwarding
                                                          without SRTP
                                                          decryption.
                                                          Comments
                                                          welcome.<br>
                                                          <br>
                                                          Regards,<br>
                                                          Adam Fineberg<o:p></o:p></span></p>
                                                        <div>
                                                          <p
                                                          class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          target="_blank">fineberg
                                                          at vline.com</a><br>
                                                          <br>
                                                          --------
                                                          Original
                                                          Message
                                                          -------- <o:p></o:p></span></p>
                                                          <table
                                                          class="MsoNormalTable"
                                                          border="0"
                                                          cellpadding="0"
cellspacing="0">
                                                          <tbody>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>Subject:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">New
                                                          Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>Date:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">Tue,
                                                          09 Jul 2013
                                                          10:02:05 -0700<o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>From:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal"><a
moz-do-not-send="true" href="mailto:internet-drafts%20at%20ietf.org"
                                                          target="_blank">internet-drafts
                                                          at ietf.org</a><o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>To:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">Adam
                                                          Fineberg&nbsp;<a
                                                          moz-do-not-send="true"
href="mailto:fineberg%20at%20vline.com" target="_blank">&lt;fineberg at
                                                          vline.com&gt;</a><o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          </tbody>
                                                          </table>
                                                          <p
                                                          class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
                                                          <br>
                                                          <br>
                                                          <o:p></o:p></span></p>
                                                          <pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></pre>
                                                          <pre>has been successfully submitted by Adam Fineberg and posted to the<o:p></o:p></pre>
                                                          <pre>IETF repository.<o:p></o:p></pre>
                                                          <pre><o:p>&nbsp;</o:p></pre>
                                                          <pre>Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  draft-fineberg-avtext-temporal-layer-ext<o:p></o:p></pre>
                                                          <pre>Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  00<o:p></o:p></pre>
                                                          <pre>Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information<o:p></o:p></pre>
                                                          <pre>Creation date:&nbsp;&nbsp;  2013-07-08<o:p></o:p></pre>
                                                          <pre>Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  Individual Submission<o:p></o:p></pre>
                                                          <pre>Number of pages: 6<o:p></o:p></pre>
                                                          <pre>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a><o:p></o:p></pre>
                                                          <pre>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a><o:p></o:p></pre>
                                                          <pre>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a><o:p></o:p></pre>
                                                          <pre><o:p>&nbsp;</o:p></pre>
                                                          <pre><o:p>&nbsp;</o:p></pre>
                                                          <pre>Abstract:<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; This document defines a mechanism by which packets of Real-Time<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; Tranport Protocol (RTP) video streams encoded with the VP8 codec can<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; indicate, in an RTP header extension, the temporal layer information<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; about the frame encoded in the RTP packet.&nbsp; This information can be<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; used in a middlebox performing bandwidth management of streams<o:p></o:p></pre>
                                                          <pre>&nbsp;&nbsp; without requiring it to decrypt the streams.<o:p></o:p></pre>
                                                        </div>
                                                        <p
                                                          class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
                                                          _______________________________________________
                                                          rtcweb mailing
                                                          list <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">
rtcweb@ietf.org</a> <a moz-do-not-send="true"
                                                          href="https://www.ietf.org/mailman/listinfo/rtcweb"
target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
                                                      </div>
                                                    </div>
                                                  </blockquote>
                                                  <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                                                  <pre>-- <o:p></o:p></pre>
                                                  <pre>Regards,<o:p></o:p></pre>
                                                  <pre>Adam<o:p></o:p></pre>
                                                </div>
                                              </div>
                                            </blockquote>
                                            <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                                          </div>
                                        </blockquote>
                                      </div>
                                    </div>
                                  </div>
                                  <p class="MsoNormal"
                                    style="margin-bottom:12.0pt"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
_______________________________________________<br>
                                      rtcweb mailing list<br>
                                      <a moz-do-not-send="true"
                                        href="mailto:rtcweb@ietf.org"
                                        target="_blank">rtcweb@ietf.org</a><br>
                                      <a moz-do-not-send="true"
                                        href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                        target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
                                </div>
                                <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                              </div>
                            </div>
                          </div>
                        </div>
                        <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
                            <br>
                            <o:p></o:p></span></p>
                        <pre>_______________________________________________<o:p></o:p></pre>
                        <pre>avtext mailing list<o:p></o:p></pre>
                        <pre><a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/avtext" target="_blank">https://www.ietf.org/mailman/listinfo/avtext</a><o:p></o:p></pre>
                      </blockquote>
                      <p class="MsoNormal"><span
style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
                      <pre>-- <o:p></o:p></pre>
                      <pre>Regards,<o:p></o:p></pre>
                      <pre>Adam<o:p></o:p></pre>
                    </div>
                  </div>
                </blockquote>
                <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
                <pre>-- <o:p></o:p></pre>
                <pre>Regards,<o:p></o:p></pre>
                <pre>Adam<o:p></o:p></pre>
              </div>
            </div>
          </blockquote>
          <p class="MsoNormal"><br>
            <br>
            <o:p></o:p></p>
          <pre>-- <o:p></o:p></pre>
          <pre>Regards,<o:p></o:p></pre>
          <pre>Adam<o:p></o:p></pre>
        </div>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Regards,
Adam</pre>
  </body>
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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Adam Fineberg <fineberg@vline.me>
Thread-Topic: [avtext] [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Adam,

I meant the scalable codecs mentioned so far by everyone, including H.265/H=
EVC and its extensions, and H.264/AVC and its extensions.

BR, YK

From: Adam Fineberg [mailto:fineberg@vline.me]
Sent: Wednesday, July 24, 2013 3:55 PM
To: Wang, Ye-Kui
Cc: Bernard Aboba; avtext@ietf.org; rtcweb@ietf.org; Justin Uberti
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

YK,

I would appreciate your collaboration.  Which codecs are you referring to w=
hen you say "all existing standard scalable video codecs"?

Regards,
Adam
On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:
If the group is to specify something generic, naturally it should be generi=
c enough to cover at least all existing standard scalable video codecs if p=
ossible. And I personally think that is possible and not difficult at all. =
Thus, why limit to only a few scalable video codecs?

I could provide some help here too if needed.

BR, YK

From: avtext-bounces@ietf.org<mailto:avtext-bounces@ietf.org> [mailto:avtex=
t-bounces@ietf.org] On Behalf Of Adam Fineberg
Sent: Wednesday, July 24, 2013 10:08 AM
To: Bernard Aboba
Cc: avtext@ietf.org<mailto:avtext@ietf.org>; rtcweb@ietf.org<mailto:rtcweb@=
ietf.org>; Justin Uberti
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Bernard,

I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.  Since the fields are not the same for the codecs, we will=
 be multiplexing the bits and that seems to me to add complexity rather tha=
n add clarity.  Also, I can't find an IETF VP9 document for the payload for=
mat to reference.  If the group thinks generalization is the right approach=
 I would appreciate some collaboration on getting the right bit definitions=
 for the other codecs.

Regards,
Adam
On 7/23/13 12:07 PM, Bernard Aboba wrote:
I do not think it is necessary to "support all forms of scalability for all=
 codecs".   In fact, I would make that an explicit "non-goal".  All that wa=
s suggested is to try to create a single extension that supports a few comm=
on cases.   If it is possible to handle VP8, VP9 and H.264/SVC in a single =
extension that would be sufficient.  So why not limit it to that?
________________________________
Date: Tue, 23 Jul 2013 08:53:45 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: stewe@stewe.org<mailto:stewe@stewe.org>
CC: juberti@google.com<mailto:juberti@google.com>; bernard_aboba@hotmail.co=
m<mailto:bernard_aboba@hotmail.com>; avtext@ietf.org<mailto:avtext@ietf.org=
>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization might not be the best approach.  =
I'm not sure what we really gain by trying to capture all this in a single =
header extension rather than one per that can succinctly explain the fields=
 without the complexity of multiplexing the bits.

Thoughts?

Regards,
Adam
On 7/19/13 3:44 PM, Stephan Wenger wrote:
Hi,

From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Friday, 19 July, 2013 15:12
To: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Cc: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, Bernard =
Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>>, "avtex=
t@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtext@ietf.org=
>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not at=
 all familiar with H.265.  I'll review the reference and see what I can fig=
ure.

StW: Good luck :-)

It seems though to me that you are suggesting that except in the simple cas=
e, that the data for H.265 would not be well suited to a header extension, =
am I understanding you correctly?  There is no reason the middlebox couldn'=
t get out of band signaling of the VPS as you mention but that would not be=
 within the scope of this header extension.

StW: well, if you would copy the layer_id into your header extension (just =
as you need to do for the simple case), a really smart middle box could use=
 this information just as a decoder uses it, assuming that it intercepted t=
he VPS in the first place.  Insofar, I wouldn't rule out the second option =
on technical grounds.  Whether any of the actual products would bother to d=
o that, ever, is another question.  I think the case ought to be documented=
, though.  I can help drafting text.
While we are at it: doing this right could mean that you need multiple spec=
s.  First, a generic header extension mechanism dedicated to side informati=
on required for pruning of RTP packet streams-ideally not only for scalable=
 video, although that is the main customer today.  And second, for each "pa=
yload" (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 =
(including scalable and 3D extensions, which are not yet finalized), VP8, a=
nd VP9 (I know little about the latter), plus Daala, and whatnot) a mapping=
 of the bits available in the generic header extension to the bits in the p=
ayload itself (NAL header and VPS in case of H.265, NALU header in SVC, and=
 the fields you mention for VP8).

Stephan


Any insights are appreciated.

Regards,
Adam
On 7/19/13 8:33 AM, Stephan Wenger wrote:
Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling-as the encrypted in-band VPS would not be useful under the assump=
tion that the middle box does not have the key to the media-which is the mo=
tivation of the draft in the first place).  I personally don't mind at all =
the second mechanism, as I'm a big fan of out-of-band parameter set transmi=
ssion and any middle box must be in the signaling path anyway to meaningful=
ly manipulate RTP.  I do not like the first option due to its variable, and=
 possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.

On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:
Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).
________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam
On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:

New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.tx=
t

Date:

Tue, 09 Jul 2013 10:02:05 -0700

From:

internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>

To:

Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.com>






A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt

has been successfully submitted by Adam Fineberg and posted to the

IETF repository.



Filename:         draft-fineberg-avtext-temporal-layer-ext

Revision:         00

Title:            A Real-Time Transport Protocol (RTP) Header Extension for=
 VP8 Temporal Layer Information

Creation date:    2013-07-08

Group:            Individual Submission

Number of pages: 6

URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt

Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext

Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00





Abstract:

   This document defines a mechanism by which packets of Real-Time

   Tranport Protocol (RTP) video streams encoded with the VP8 codec can

   indicate, in an RTP header extension, the temporal layer information

   about the frame encoded in the RTP packet.  This information can be

   used in a middlebox performing bandwidth management of streams

   without requiring it to decrypt the streams.

_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--

Regards,

Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb





_______________________________________________

avtext mailing list

avtext@ietf.org<mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinf=
o/avtext


--

Regards,

Adam


--

Regards,

Adam




--

Regards,

Adam



--

Regards,

Adam

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Adam,<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I meant the scalable code=
cs mentioned so far by everyone, including H.265/HEVC and its extensions, a=
nd H.264/AVC and its extensions.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR, YK<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> Adam Fineberg [mail=
to:fineberg@vline.me]
<br>
<b>Sent:</b> Wednesday, July 24, 2013 3:55 PM<br>
<b>To:</b> Wang, Ye-Kui<br>
<b>Cc:</b> Bernard Aboba; avtext@ietf.org; rtcweb@ietf.org; Justin Uberti<b=
r>
<b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">YK,<br>
<br>
I would appreciate your collaboration.&nbsp; Which codecs are you referring=
 to when you say &quot;all existing standard scalable video codecs&quot;?
<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:<o:p></o:p></=
p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">If the group is to specif=
y something generic, naturally it should be generic enough to cover at leas=
t all existing standard scalable video codecs if possible.
 And I personally think that is possible and not difficult at all. Thus, wh=
y limit to only a few scalable video codecs?</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I could provide some help=
 here too if needed.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR, YK</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
<a href=3D"mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a> [<a =
href=3D"mailto:avtext-bounces@ietf.org">mailto:avtext-bounces@ietf.org</a>]
<b>On Behalf Of </b>Adam Fineberg<br>
<b>Sent:</b> Wednesday, July 24, 2013 10:08 AM<br>
<b>To:</b> Bernard Aboba<br>
<b>Cc:</b> <a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>; <a href=
=3D"mailto:rtcweb@ietf.org">
rtcweb@ietf.org</a>; Justin Uberti<br>
<b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Bernard,<br>
<br>
I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.&nbsp; Since the fields are not the same for the codecs, we=
 will be multiplexing the bits and that seems to me to add complexity rathe=
r than add clarity.&nbsp; Also, I can't find
 an IETF VP9 document for the payload format to reference.&nbsp; If the gro=
up thinks generalization is the right approach I would appreciate some coll=
aboration on getting the right bit definitions for the other codecs.<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/23/13 12:07 PM, Bernard Aboba wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I do not think it is =
necessary to &quot;support all forms of scalability for all codecs&quot;.&n=
bsp;&nbsp; In fact, I would make that an explicit &quot;non-goal&quot;.&nbs=
p; All that was suggested is to try to create a single extension that suppo=
rts
 a few common cases.&nbsp;&nbsp; If it is possible to handle VP8, VP9 and H=
.264/SVC in a single extension that would be sufficient.&nbsp; So why not l=
imit it to that?
<o:p></o:p></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Date: Tue, 23 Jul 201=
3 08:53:45 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me">fineberg@vline.me</a><br>
To: <a href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a><br>
CC: <a href=3D"mailto:juberti@google.com">juberti@google.com</a>; <a href=
=3D"mailto:bernard_aboba@hotmail.com">
bernard_aboba@hotmail.com</a>; <a href=3D"mailto:avtext@ietf.org">avtext@ie=
tf.org</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt<br>
<br>
I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization
 might not be the best approach.&nbsp; I'm not sure what we really gain by =
trying to capture all this in a single header extension rather than one per=
 that can succinctly explain the fields without the complexity of multiplex=
ing the bits.<br>
<br>
Thoughts?<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/19/13 3:44 PM, Stephan Wenger wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">Hi,</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;">Adam Fineberg &lt;<a href=3D"mailto:fineberg@vline.=
me">fineberg@vline.me</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 15:12 <br>
<b>To: </b>Stephan Wenger &lt;<a href=3D"mailto:stewe@stewe.org">stewe@stew=
e.org</a>&gt;<br>
<b>Cc: </b>Justin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@=
google.com</a>&gt;, Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotma=
il.com">bernard_aboba@hotmail.com</a>&gt;, &quot;<a href=3D"mailto:avtext@i=
etf.org">avtext@ietf.org</a>&quot; &lt;<a href=3D"mailto:avtext@ietf.org">a=
vtext@ietf.org</a>&gt;,
 &quot;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Stephan,<br>
<br>
Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm n=
ot at all familiar with H.265.&nbsp; I'll review the reference and see what=
 I can figure.&nbsp;</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">StW: Good luck :-) &nbsp;</s=
pan><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">It seems though to me that you are sugg=
esting that except in the simple case, that the data for H.265 would not be=
 well suited to a header extension, am I understanding you
 correctly?&nbsp; There is no reason the middlebox couldn't get out of band=
 signaling of the VPS as you mention but that would not be within the scope=
 of this header extension.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:blue">StW: well, if you would copy the layer_id int=
o your header extension (just as you need to do for the simple case), a rea=
lly smart middle box could use this information just as
 a decoder uses it,&nbsp;assuming&nbsp;that it intercepted the VPS in the f=
irst place. &nbsp;Insofar, I wouldn't rule out the second option on technic=
al grounds. &nbsp;Whether any of the actual products would bother to do tha=
t, ever, is another question. &nbsp;I think the case ought
 to be documented, though. &nbsp;I can help drafting text.</span><o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:blue">While we are at it: doing this right could me=
an that you need multiple specs. &nbsp;First, a generic header extension me=
chanism dedicated to side information required for pruning of
 RTP packet streams&#8212;ideally not only for scalable video, although tha=
t is the main customer today. &nbsp;And second, for each &quot;payload&quot=
; (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 (incl=
uding scalable and 3D extensions, which are not yet finalized),
 VP8, and VP9 (I know little about the latter), plus Daala, and whatnot) a =
mapping of the bits available in the generic header extension to the bits i=
n the payload itself (NAL header and VPS in case of H.265, NALU header in S=
VC, and the fields you mention for
 VP8).</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:blue">Stephan</span><o:p></o:p></p=
>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
<br>
Any insights are appreciated.<br>
<br>
Regards,<br>
Adam</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On 7/19/13 8:33 AM, Stephan Wenger wrot=
e:</span><o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi,</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">I also believe that 16 bits should be e=
nough. &nbsp;For H.264 and VP8 that has already been demonstrated. &nbsp;Fo=
r H.265, some initial thoughts below. &nbsp;Apologies for the word-count.</=
span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">The scalable version of H265 (called SH=
VC) is currently under development. &nbsp;The current working draft can be =
found here:&nbsp;<a href=3D"http://phenix.int-evry.fr/jct/doc_end_user/docu=
ments/13_Incheon/wg11/JCTVC-M1008-v3.zip" target=3D"_blank">http://phenix.i=
nt-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a=
>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">That said, the ordering by the encoder =
is just a recommendation, and there may well be cases where different pruni=
ng strategies may be advisable. &nbsp;For example, a layering
 structure could be constructed that expands into two branches, one using 2=
D scalable tools only, the other including view_id for multi view coding. &=
nbsp;By looking at the six bit field alone, a middle box will not be able t=
o meaningfully remove NAL units belonging
 to one of the branches completely while pruning the other branch. &nbsp;In=
 order to meaningfully deal with that scenario, there would be two options:=
 one to represent the dimension_id[][] (and associated control info) in the=
 header extension, or require the middle
 box to have access to the VPS and be able to interpret its content. &nbsp;=
The further could take considerably more than 16 bits and we would be talki=
ng about a variable length data structure. &nbsp;The latter requires the mi=
ddle box to have state and a mechanism to
 intercept the VPS (through signaling&#8212;as the encrypted in-band VPS wo=
uld not be useful under the assumption that the middle box does not have th=
e key to the media&#8212;which is the motivation of the draft in the first =
place). &nbsp;I personally don't mind at all the
 second mechanism, as I'm a big fan of out-of-band parameter set transmissi=
on and any middle box must be in the signaling path anyway to meaningfully =
manipulate RTP. &nbsp;I do not like the first option due to its variable, a=
nd possibly substantial, overhead.</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Stephan &nbsp;&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;">Justin Uberti &lt;<a href=3D"mailto:juberti@google.=
com">juberti@google.com</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 06:32 <br>
<b>To: </b>Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.com">b=
ernard_aboba@hotmail.com</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quo=
t; &lt;<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, &quot;<a=
 href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"m=
ailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [rtcweb] Fwd: New Version Notification for draft-finebe=
rg-avtext-temporal-layer-ext-00.txt</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Agree those are the right codecs to des=
ign for. Since in each case there are fairly low limits on the number of su=
pported layers (i.e. 3 spatial layers for SVC), I think
 it should be possible to pack the temporal, spatial, quality layer ids int=
o 16 bits.
</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</=
span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On Fri, Jul 19, 2013 at 1:56 AM, Bernar=
d Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">=
bernard_aboba@hotmail.com</a>&gt; wrote:</span><o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If we can support VP8/9 as well as H.26=
4/5 SVC</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">that would be a start. It seems doable =
to me.</span><o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a href=3D"mailt=
o:fineberg@vline.me" target=3D"_blank">fineberg@vline.me</a>&gt; wrote:</sp=
an><o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:</span><o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">I think =
it may be possible to generalize this.&nbsp; For example, for H.264/SVC whi=
ch can support temporal, spatial and quality scalability, you would
 need the quality_id and dependency_id in addition to the temporal_id (what=
 you call the temporal layer index). &nbsp;&nbsp;
</span><o:p></o:p></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Date: Th=
u, 18 Jul 2013 08:45:38 -0700<br>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_=
aboba@hotmail.com</a><br>
CC: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
>; <a href=3D"mailto:avtext@ietf.org" target=3D"_blank">
avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">On 7/17/13 8:26 PM, Bernard Aboba wrote=
:</span><o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Since the need is not codec specific (e=
.g. it arises with any codec supporting temporal, spatial and quality scala=
bility), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;</span><o:p></o:p></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Date: Wed, 17 Jul 2013 17:09:46 -0700<b=
r>
From: <a href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vline=
.me</a><br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
Hi,<br>
<br>
I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt
 the SRTP packets. This is undesirable both because the middle-box needs to=
 have access to the keys as well as the because of the added overhead of th=
e decrypt/encrypt cycle. This draft proposes an RTP header extension that w=
ill allow us to use the VP8 temporal
 layer information included in the header extension and therefore do forwar=
ding without SRTP decryption. Comments welcome.<br>
<br>
Regards,<br>
Adam Fineberg</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><a href=3D"mailto:fineberg%20at%20vline=
.com" target=3D"_blank">fineberg at vline.com</a><br>
<br>
-------- Original Message -------- </span><o:p></o:p></p>
<table class=3D"MsoNormalTable" border=3D"0" cellspacing=3D"0" cellpadding=
=3D"0">
<tbody>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Subjec=
t:</b><o:p></o:p></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Date:<=
/b><o:p></o:p></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">Tue, 09 Jul 2013 10:02:05 -0700<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>From:<=
/b><o:p></o:p></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal"><a href=3D"mailto:internet-drafts%20at%20ietf.org" t=
arget=3D"_blank">internet-drafts at ietf.org</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>To:</b=
><o:p></o:p></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">Adam Fineberg&nbsp;<a href=3D"mailto:fineberg%20at%2=
0vline.com" target=3D"_blank">&lt;fineberg at vline.com&gt;</a><o:p></o:p><=
/p>
</td>
</tr>
</tbody>
</table>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
<br>
<br>
<br>
</span><o:p></o:p></p>
<pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<=
o:p></o:p></pre>
<pre>has been successfully submitted by Adam Fineberg and posted to the<o:p=
></o:p></pre>
<pre>IETF repository.<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; draft-finebe=
rg-avtext-temporal-layer-ext<o:p></o:p></pre>
<pre>Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 00<o:p></o:p=
></pre>
<pre>Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal L=
ayer Information<o:p></o:p></pre>
<pre>Creation date:&nbsp;&nbsp;&nbsp; 2013-07-08<o:p></o:p></pre>
<pre>Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; Individual Submission<o:p></o:p></pre>
<pre>Number of pages: 6<o:p></o:p></pre>
<pre>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &nbsp;<a href=3D"http://www.ietf.org/internet-drafts/draft-fineberg-avtext=
-temporal-layer-ext-00.txt" target=3D"_blank">http://www.ietf.org/internet-=
drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a><o:p></o:p></pre>
<pre>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a href=
=3D"http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ex=
t" target=3D"_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-=
temporal-layer-ext</a><o:p></o:p></pre>
<pre>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a href=3D"http://=
tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target=3D"=
_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext=
-00</a><o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>Abstract:<o:p></o:p></pre>
<pre>&nbsp;&nbsp; This document defines a mechanism by which packets of Rea=
l-Time<o:p></o:p></pre>
<pre>&nbsp;&nbsp; Tranport Protocol (RTP) video streams encoded with the VP=
8 codec can<o:p></o:p></pre>
<pre>&nbsp;&nbsp; indicate, in an RTP header extension, the temporal layer =
information<o:p></o:p></pre>
<pre>&nbsp;&nbsp; about the frame encoded in the RTP packet.&nbsp; This inf=
ormation can be<o:p></o:p></pre>
<pre>&nbsp;&nbsp; used in a middlebox performing bandwidth management of st=
reams<o:p></o:p></pre>
<pre>&nbsp;&nbsp; without requiring it to decrypt the streams.<o:p></o:p></=
pre>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
_______________________________________________ rtcweb mailing list <a href=
=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb=
" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a></span><o:p></o:p></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</blockquote>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a></span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><br>
<br>
<br>
</span><o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>avtext mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><a href=3D"https=
://www.ietf.org/mailman/listinfo/avtext" target=3D"_blank">https://www.ietf=
.org/mailman/listinfo/avtext</a><o:p></o:p></pre>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</body>
</html>

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From stewe@stewe.org  Thu Jul 25 00:45:14 2013
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From: Stephan Wenger <stewe@stewe.org>
To: Adam Fineberg <fineberg@vline.me>, "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
Thread-Topic: [avtext] [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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--_000_CE169EA79FC3Cstewesteweorg_
Content-Type: text/plain; charset="Windows-1252"
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Hi,
I thought about this a bit more.  I think that one sensible document struct=
ure may be as follows:

(1) generic structure of scalability/pruning information to an RTP header e=
xtension (to be worked on in avtext).
(2) mapping of codec-specific features to the generic structure.  As a loca=
tion, I would suggest that (for new codecs) the RTP payload specs are the r=
ight document.  For codecs with existing RTP payload specs, a short doc tha=
t updates the payload spec would be the right place, or a revision of the p=
ayload spec itself if that needs to be done anyway.  This would bundle all =
the codec-specifics to one document.

This document structure, and the philosophy behind it, would allow the desi=
gn of a generic RTP stack that could take advantage of scalability features=
 independently of the scalable codec actually in use.  I believe that such =
an advantage outweighs the possible advantage of quick standardization of a=
 more specific approach as proposed in Adam's draft.  That said, we need to=
 get the design of the generic structure right such that it is indeed usefu=
l, ideally for all present and hopefully for a large subset of future codec=
s.

As for the specs involved:
-H.264 SVC, H.265, and VP8 have payload format RFCs or reasonably stable dr=
afts.
-No one uses H.263 and MPEG-2 scalability, so we can ignore those two.
-As Adam, I'm also not aware of a VP9 payload spec draft.  However, my unde=
rstanding is that VP9 will include a number of scalability features enabled=
 by multiple reference pictures and reference picture resampling.  The latt=
er is a somewhat different approach compared to the traditional scalability=
 implementations.  I think that google planned to freeze the VP9 bitstream =
a few weeks go, but I don't know whether that has happened.  Without a stab=
le spec and/or input from google/webm folks it will indeed be hard to addre=
ss VP9's specific needs, if any.
-There are a bunch of audio codecs that claim to be scalable.  We should ha=
ve a scope discussion on whether those need to be included here.

Stephan



From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Thursday, 25 July, 2013 00:54
To: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com<mailto:yekuiw@qti.qualcomm.com>=
>
Cc: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>, "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avt=
ext@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<=
mailto:rtcweb@ietf.org>>, Justin Uberti <juberti@google.com<mailto:juberti@=
google.com>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

YK,

I would appreciate your collaboration.  Which codecs are you referring to w=
hen you say "all existing standard scalable video codecs"?

Regards,
Adam

On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:
If the group is to specify something generic, naturally it should be generi=
c enough to cover at least all existing standard scalable video codecs if p=
ossible. And I personally think that is possible and not difficult at all. =
Thus, why limit to only a few scalable video codecs?

I could provide some help here too if needed.

BR, YK

From:avtext-bounces@ietf.org<mailto:avtext-bounces@ietf.org> [mailto:avtext=
-bounces@ietf.org] On Behalf Of Adam Fineberg
Sent: Wednesday, July 24, 2013 10:08 AM
To: Bernard Aboba
Cc: avtext@ietf.org<mailto:avtext@ietf.org>; rtcweb@ietf.org<mailto:rtcweb@=
ietf.org>; Justin Uberti
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Bernard,

I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.  Since the fields are not the same for the codecs, we will=
 be multiplexing the bits and that seems to me to add complexity rather tha=
n add clarity.  Also, I can't find an IETF VP9 document for the payload for=
mat to reference.  If the group thinks generalization is the right approach=
 I would appreciate some collaboration on getting the right bit definitions=
 for the other codecs.

Regards,
Adam
On 7/23/13 12:07 PM, Bernard Aboba wrote:
I do not think it is necessary to "support all forms of scalability for all=
 codecs".   In fact, I would make that an explicit "non-goal".  All that wa=
s suggested is to try to create a single extension that supports a few comm=
on cases.   If it is possible to handle VP8, VP9 and H.264/SVC in a single =
extension that would be sufficient.  So why not limit it to that?
________________________________
Date: Tue, 23 Jul 2013 08:53:45 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: stewe@stewe.org<mailto:stewe@stewe.org>
CC: juberti@google.com<mailto:juberti@google.com>; bernard_aboba@hotmail.co=
m<mailto:bernard_aboba@hotmail.com>; avtext@ietf.org<mailto:avtext@ietf.org=
>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization might not be the best approach.  =
I'm not sure what we really gain by trying to capture all this in a single =
header extension rather than one per that can succinctly explain the fields=
 without the complexity of multiplexing the bits.

Thoughts?

Regards,
Adam
On 7/19/13 3:44 PM, Stephan Wenger wrote:
Hi,

From: Adam Fineberg <fineberg@vline.me<mailto:fineberg@vline.me>>
Date: Friday, 19 July, 2013 15:12
To: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Cc: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, Bernard =
Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>>, "avtex=
t@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtext@ietf.org=
>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt

Stephan,

Thanks for the info and the reference.  I'm not sure I follow as I'm not at=
 all familiar with H.265.  I'll review the reference and see what I can fig=
ure.

StW: Good luck :-)

It seems though to me that you are suggesting that except in the simple cas=
e, that the data for H.265 would not be well suited to a header extension, =
am I understanding you correctly?  There is no reason the middlebox couldn'=
t get out of band signaling of the VPS as you mention but that would not be=
 within the scope of this header extension.

StW: well, if you would copy the layer_id into your header extension (just =
as you need to do for the simple case), a really smart middle box could use=
 this information just as a decoder uses it, assuming that it intercepted t=
he VPS in the first place.  Insofar, I wouldn't rule out the second option =
on technical grounds.  Whether any of the actual products would bother to d=
o that, ever, is another question.  I think the case ought to be documented=
, though.  I can help drafting text.
While we are at it: doing this right could mean that you need multiple spec=
s.  First, a generic header extension mechanism dedicated to side informati=
on required for pruning of RTP packet streams=97ideally not only for scalab=
le video, although that is the main customer today.  And second, for each "=
payload" (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v=
2 (including scalable and 3D extensions, which are not yet finalized), VP8,=
 and VP9 (I know little about the latter), plus Daala, and whatnot) a mappi=
ng of the bits available in the generic header extension to the bits in the=
 payload itself (NAL header and VPS in case of H.265, NALU header in SVC, a=
nd the fields you mention for VP8).

Stephan


Any insights are appreciated.

Regards,
Adam
On 7/19/13 8:33 AM, Stephan Wenger wrote:
Hi,
I also believe that 16 bits should be enough.  For H.264 and VP8 that has a=
lready been demonstrated.  For H.265, some initial thoughts below.  Apologi=
es for the word-count.

The scalable version of H265 (called SHVC) is currently under development. =
 The current working draft can be found here: http://phenix.int-evry.fr/jct=
/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.  Therein, the o=
ptions for defining layering structures are considerably more complex.  To =
start, we have 3 bits for the temporal ID in the NAL unit header of the H.2=
65 version 1 (HEVC) base specification (temporal scalability is already nic=
ely supported in version 1).  Just like in SVC.  In the scalable extension,=
 the NAL unit header contains a six bit field that points into a data struc=
ture known as "Video Parameter Set" (VPS).  Inside the VPS, those six bits =
are mapped to to a position in a directed graph (specified through "dimensi=
on_id[][]"), which tells you about the reference relationship of the layer =
in question and its parent layer.  One can recursively follow the graph to =
determine what used to be called dependency_id, quality_id, view_id, and wh=
atnot.  The six bit pointer field can (or: is to be when possible) organize=
d by the encoder such that it is prudent for a middle box to throw away NAL=
 units (belonging to layers) with higher values of the six bit field first,=
 before throwing away NAL units with lower values.  Relying on this feature=
, 3+6 bits =3D=3D 9 bits should be fine for the header extension.

That said, the ordering by the encoder is just a recommendation, and there =
may well be cases where different pruning strategies may be advisable.  For=
 example, a layering structure could be constructed that expands into two b=
ranches, one using 2D scalable tools only, the other including view_id for =
multi view coding.  By looking at the six bit field alone, a middle box wil=
l not be able to meaningfully remove NAL units belonging to one of the bran=
ches completely while pruning the other branch.  In order to meaningfully d=
eal with that scenario, there would be two options: one to represent the di=
mension_id[][] (and associated control info) in the header extension, or re=
quire the middle box to have access to the VPS and be able to interpret its=
 content.  The further could take considerably more than 16 bits and we wou=
ld be talking about a variable length data structure.  The latter requires =
the middle box to have state and a mechanism to intercept the VPS (through =
signaling=97as the encrypted in-band VPS would not be useful under the assu=
mption that the middle box does not have the key to the media=97which is th=
e motivation of the draft in the first place).  I personally don't mind at =
all the second mechanism, as I'm a big fan of out-of-band parameter set tra=
nsmission and any middle box must be in the signaling path anyway to meanin=
gfully manipulate RTP.  I do not like the first option due to its variable,=
 and possibly substantial, overhead.

Stephan


From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, 19 July, 2013 06:32
To: Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.c=
om>>
Cc: "avtext@ietf.org<mailto:avtext@ietf.org>" <avtext@ietf.org<mailto:avtex=
t@ietf.org>>, "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<ma=
ilto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Agree those are the right codecs to design for. Since in each case there ar=
e fairly low limits on the number of supported layers (i.e. 3 spatial layer=
s for SVC), I think it should be possible to pack the temporal, spatial, qu=
ality layer ids into 16 bits.

On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba <bernard_aboba@hotmail.com<m=
ailto:bernard_aboba@hotmail.com>> wrote:
If we can support VP8/9 as well as H.264/5 SVC
that would be a start. It seems doable to me.

On Jul 18, 2013, at 8:34 PM, "Adam Fineberg" <fineberg@vline.me<mailto:fine=
berg@vline.me>> wrote:
Bernard,

Are there other codecs you are thinking should be supported?  If it's gener=
alized I would think we want to be able to cover all known scalable codecs.=
 I'll look into the H264/SVC fields to see how to encode them in a generali=
zed header.

Regards,
Adam

On 7/18/13 7:40 PM, Bernard Aboba wrote:
I think it may be possible to generalize this.  For example, for H.264/SVC =
which can support temporal, spatial and quality scalability, you would need=
 the quality_id and dependency_id in addition to the temporal_id (what you =
call the temporal layer index).
________________________________
Date: Thu, 18 Jul 2013 08:45:38 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: bernard_aboba@hotmail.com<mailto:bernard_aboba@hotmail.com>
CC: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; avtext@ietf.org<mailto:avtext@=
ietf.org>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt

Bernard,

Good question.  I'm not familiar enough with the parameter requirements of =
all other scalable codecs to be able to generalize.  If you'd like to help =
specify them, I'd be fine revising the draft to generalize.

Regards,
Adam
On 7/17/13 8:26 PM, Bernard Aboba wrote:
Since the need is not codec specific (e.g. it arises with any codec support=
ing temporal, spatial and quality scalability), why
 a VP8-specific RTP extension?

________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me<mailto:fineberg@vline.me>
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com<mailto:fineberg%20at%20vline.com>

-------- Original Message --------
Subject:

New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.tx=
t

Date:

Tue, 09 Jul 2013 10:02:05 -0700

From:

internet-drafts at ietf.org<mailto:internet-drafts%20at%20ietf.org>

To:

Adam Fineberg <fineberg at vline.com><mailto:fineberg%20at%20vline.com>





A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt

has been successfully submitted by Adam Fineberg and posted to the

IETF repository.



Filename:         draft-fineberg-avtext-temporal-layer-ext

Revision:         00

Title:            A Real-Time Transport Protocol (RTP) Header Extension for=
 VP8 Temporal Layer Information

Creation date:    2013-07-08

Group:            Individual Submission

Number of pages: 6

URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt

Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext

Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00





Abstract:

   This document defines a mechanism by which packets of Real-Time

   Tranport Protocol (RTP) video streams encoded with the VP8 codec can

   indicate, in an RTP header extension, the temporal layer information

   about the frame encoded in the RTP packet.  This information can be

   used in a middlebox performing bandwidth management of streams

   without requiring it to decrypt the streams.

_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb


--

Regards,

Adam


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb




_______________________________________________

avtext mailing list

avtext@ietf.org<mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinf=
o/avtext


--

Regards,

Adam


--

Regards,

Adam



--

Regards,

Adam


--
Regards,
Adam

--_000_CE169EA79FC3Cstewesteweorg_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
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</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>Hi,</div>
<div>I thought about this a bit more. &nbsp;I think that one sensible docum=
ent structure may be as follows:</div>
<div><br>
</div>
<div>(1) generic structure of scalability/pruning information to an RTP hea=
der extension (to be worked on in avtext).</div>
<div>(2) mapping of codec-specific features to the generic structure. &nbsp=
;As a location, I would suggest that (for new codecs) the RTP payload specs=
 are the right document. &nbsp;For codecs with existing RTP payload specs, =
a short doc that updates the payload spec
 would be the right place, or a revision of the payload spec itself if that=
 needs to be done anyway. &nbsp;This would bundle all the codec-specifics t=
o one document.</div>
<div><br>
</div>
<div>This document structure, and the philosophy behind it, would allow the=
 design of a generic RTP stack that could take advantage of scalability fea=
tures independently of the scalable codec actually in use. &nbsp;I believe =
that such an advantage outweighs the
 possible advantage of quick standardization of a more specific approach as=
 proposed in Adam's draft. &nbsp;That said, we need to get the design of th=
e generic structure right such that it is indeed useful, ideally for all pr=
esent and hopefully for a large subset
 of future codecs.</div>
<div><br>
</div>
<div>As for the specs involved:</div>
<div>-H.264 SVC, H.265, and VP8 have payload format RFCs or reasonably stab=
le drafts.</div>
<div>-No one uses H.263 and MPEG-2 scalability, so we can ignore those two.=
</div>
<div>-As Adam, I'm also not aware of a VP9 payload spec draft. &nbsp;Howeve=
r, my understanding is that VP9 will include a number of scalability featur=
es enabled by multiple reference pictures and reference picture resampling.=
 &nbsp;The latter is a somewhat different
 approach compared to the traditional scalability implementations. &nbsp;I =
think that google planned to freeze the VP9 bitstream a few weeks go, but I=
 don't know whether that has happened. &nbsp;Without a stable spec and/or i=
nput from google/webm folks it will indeed
 be hard to address VP9's specific needs, if any.</div>
<div>-There are a bunch of audio codecs that claim to be scalable. &nbsp;We=
 should have a scope discussion on whether those need to be included here.<=
/div>
<div><br>
</div>
<div>Stephan</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Adam Fineberg &lt;<a href=3D"=
mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, 25 July, 2013 00:54=
 <br>
<span style=3D"font-weight:bold">To: </span>&quot;Wang, Ye-Kui&quot; &lt;<a=
 href=3D"mailto:yekuiw@qti.qualcomm.com">yekuiw@qti.qualcomm.com</a>&gt;<br=
>
<span style=3D"font-weight:bold">Cc: </span>Bernard Aboba &lt;<a href=3D"ma=
ilto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;, &quot;<a=
 href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a>&quot; &lt;<a href=3D"m=
ailto:avtext@ietf.org">avtext@ietf.org</a>&gt;, &quot;<a href=3D"mailto:rtc=
web@ietf.org">rtcweb@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;, Justin Ube=
rti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt;<br=
>
<span style=3D"font-weight:bold">Subject: </span>Re: [avtext] [rtcweb] Fwd:=
 New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.t=
xt<br>
</div>
<div><br>
</div>
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">YK,<br>
<br>
I would appreciate your collaboration.&nbsp; Which codecs are you referring=
 to when you say &quot;all existing standard scalable video codecs&quot;?
<br>
<br>
Regards,<br>
Adam<br>
<br>
<div class=3D"moz-cite-prefix">On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:<br>
</div>
<blockquote cite=3D"mid:8BA7D4CEACFFE04BA2D902BF11719A83384A0A5B@nasanexd02=
f.na.qualcomm.com" type=3D"cite">
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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); ">If the group is to specify somethi=
ng generic, naturally it should be generic enough to cover at least all exi=
sting standard scalable video codecs
 if possible. And I personally think that is possible and not difficult at =
all. Thus, why limit to only a few scalable video codecs?<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); ">I could provide some help here too=
 if needed.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); ">BR, YK<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF
              1.0pt;padding:3.0pt 0in 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size: 10pt; font-family: Taho=
ma, sans-serif; color: windowtext; " lang=3D"EN-US">From:</span></b><span s=
tyle=3D"font-size: 10pt; font-family: Tahoma, sans-serif; color: windowtext=
; " lang=3D"EN-US"><a class=3D"moz-txt-link-abbreviated" href=3D"mailto:avt=
ext-bounces@ietf.org">avtext-bounces@ietf.org</a>
 [<a class=3D"moz-txt-link-freetext" href=3D"mailto:avtext-bounces@ietf.org=
">mailto:avtext-bounces@ietf.org</a>]
<b>On Behalf Of </b>Adam Fineberg<br>
<b>Sent:</b> Wednesday, July 24, 2013 10:08 AM<br>
<b>To:</b> Bernard Aboba<br>
<b>Cc:</b> <a class=3D"moz-txt-link-abbreviated" href=3D"mailto:avtext@ietf=
.org">avtext@ietf.org</a>;
<a class=3D"moz-txt-link-abbreviated" href=3D"mailto:rtcweb@ietf.org">rtcwe=
b@ietf.org</a>; Justin Uberti<br>
<b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Bernard,<br>
<br>
I apologize if I come across as being difficult here but I stil am not seei=
ng the benefits.&nbsp; Since the fields are not the same for the codecs, we=
 will be multiplexing the bits and that seems to me to add complexity rathe=
r than add clarity.&nbsp; Also, I can't find
 an IETF VP9 document for the payload format to reference.&nbsp; If the gro=
up thinks generalization is the right approach I would appreciate some coll=
aboration on getting the right bit definitions for the other codecs.<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/23/13 12:07 PM, Bernard Aboba wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I do not think it is =
necessary to &quot;support all forms of scalability for all codecs&quot;.&n=
bsp;&nbsp; In fact, I would make that an explicit &quot;non-goal&quot;.&nbs=
p; All that was suggested is to try to create a single extension that suppo=
rts
 a few common cases.&nbsp;&nbsp; If it is possible to handle VP8, VP9 and H=
.264/SVC in a single extension that would be sufficient.&nbsp; So why not l=
imit it to that?
<o:p></o:p></p>
<div>
<div class=3D"MsoNormal" style=3D"text-align:center" align=3D"center">
<hr id=3D"stopSpelling" align=3D"center" size=3D"2" width=3D"100%">
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Date: Tue, 23 Jul 201=
3 08:53:45 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me">fineber=
g@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:stewe@stewe.org">stewe@stewe=
.org</a><br>
CC: <a moz-do-not-send=3D"true" href=3D"mailto:juberti@google.com">juberti@=
google.com</a>;
<a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com">berna=
rd_aboba@hotmail.com</a>;
<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@ietf.org=
</a>; <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">
rtcweb@ietf.org</a><br>
Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for draft-fine=
berg-avtext-temporal-layer-ext-00.txt<br>
<br>
I've been thinking about this and given the ease at which RFC5285 allows fo=
r the specification of a header extension and the complexity introduced by =
trying to generalize the header extension to support all forms of scalabili=
ty for all codecs that the generalization
 might not be the best approach.&nbsp; I'm not sure what we really gain by =
trying to capture all this in a single header extension rather than one per=
 that can succinctly explain the fields without the complexity of multiplex=
ing the bits.<br>
<br>
Thoughts?<br>
<br>
Regards,<br>
Adam<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 7/19/13 3:44 PM, Stephan Wenger wrote:<o:p></o:p>=
</p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; color: blue; ">Hi,</span><span style=3D"font-size: 10.5pt; =
font-family: Calibri, sans-serif; "><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF
                    1.0pt;padding:3.0pt 0in 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size: 11pt; font-family: Cali=
bri, sans-serif; ">From:
</span></b><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif=
; ">Adam Fineberg &lt;<a moz-do-not-send=3D"true" href=3D"mailto:fineberg@v=
line.me">fineberg@vline.me</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 15:12 <br>
<b>To: </b>Stephan Wenger &lt;<a moz-do-not-send=3D"true" href=3D"mailto:st=
ewe@stewe.org">stewe@stewe.org</a>&gt;<br>
<b>Cc: </b>Justin Uberti &lt;<a moz-do-not-send=3D"true" href=3D"mailto:jub=
erti@google.com">juberti@google.com</a>&gt;, Bernard Aboba &lt;<a moz-do-no=
t-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hot=
mail.com</a>&gt;, &quot;<a moz-do-not-send=3D"true" href=3D"mailto:avtext@i=
etf.org">avtext@ietf.org</a>&quot;
 &lt;<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@iet=
f.org</a>&gt;, &quot;<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf=
.org">rtcweb@ietf.org</a>&quot; &lt;<a moz-do-not-send=3D"true" href=3D"mai=
lto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [avtext] [rtcweb] Fwd: New Version Notification for dra=
ft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Stephan,<br>
<br>
Thanks for the info and the reference.&nbsp; I'm not sure I follow as I'm n=
ot at all familiar with H.265.&nbsp; I'll review the reference and see what=
 I can figure.&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; color: blue; ">StW: Good luck :-) &nbsp;</span><span style=
=3D"font-size: 10.5pt; font-family: Calibri, sans-serif; "><o:p></o:p></spa=
n></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">It seems though to me that you are suggesting that except=
 in the simple case, that the data for H.265 would not be well suited to a =
header extension, am I understanding
 you correctly?&nbsp; There is no reason the middlebox couldn't get out of =
band signaling of the VPS as you mention but that would not be within the s=
cope of this header extension.<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family: Calibri, sans-serif; col=
or: blue; ">StW: well, if you would copy the layer_id into your header exte=
nsion (just as you need to do for the simple case), a really smart middle b=
ox could use this information just as
 a decoder uses it,&nbsp;assuming&nbsp;that it intercepted the VPS in the f=
irst place. &nbsp;Insofar, I wouldn't rule out the second option on technic=
al grounds. &nbsp;Whether any of the actual products would bother to do tha=
t, ever, is another question. &nbsp;I think the case ought
 to be documented, though. &nbsp;I can help drafting text.</span><o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family: Calibri, sans-serif; col=
or: blue; ">While we are at it: doing this right could mean that you need m=
ultiple specs. &nbsp;First, a generic header extension mechanism dedicated =
to side information required for pruning
 of RTP packet streams=97ideally not only for scalable video, although that=
 is the main customer today. &nbsp;And second, for each &quot;payload&quot;=
 (at present we are talking about H.264/SVC, H.265v1 (HEVC), H.265v2 (inclu=
ding scalable and 3D extensions, which are not yet
 finalized), VP8, and VP9 (I know little about the latter), plus Daala, and=
 whatnot) a mapping of the bits available in the generic header extension t=
o the bits in the payload itself (NAL header and VPS in case of H.265, NALU=
 header in SVC, and the fields you
 mention for VP8).</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; color: blue; ">Stephan</span><span style=3D"font-size: 10.5=
pt; font-family: Calibri, sans-serif; "><o:p></o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; "><br>
<br>
Any insights are appreciated.<br>
<br>
Regards,<br>
Adam<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">On 7/19/13 8:33 AM, Stephan Wenger wrote:<o:p></o:p></spa=
n></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Hi,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">I also believe that 16 bits should be enough. &nbsp;For H=
.264 and VP8 that has already been demonstrated. &nbsp;For H.265, some init=
ial thoughts below. &nbsp;Apologies for the word-count.<o:p></o:p></span></=
p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">The scalable version of H265 (called SHVC) is currently u=
nder development. &nbsp;The current working draft can be found here:&nbsp;<=
a moz-do-not-send=3D"true" href=3D"http://phenix.int-evry.fr/jct/doc_end_us=
er/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip" target=3D"_blank">http://p=
henix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3=
.zip</a>.
 &nbsp;Therein, the options for defining layering structures are considerab=
ly more complex. &nbsp;To start, we have 3 bits for the temporal ID in the =
NAL unit header of the H.265 version 1 (HEVC) base specification (temporal =
scalability is already nicely supported in
 version 1). &nbsp;Just like in SVC. &nbsp;In the scalable extension, the N=
AL unit header contains a six bit field that points into a data structure k=
nown as &quot;Video Parameter Set&quot; (VPS). &nbsp;Inside the VPS, those =
six bits are mapped to to a position in a directed graph
 (specified through &quot;dimension_id[][]&quot;), which tells you about th=
e reference relationship of the layer in question and its parent layer. &nb=
sp;One can recursively follow the graph to determine what used to be called=
 dependency_id, quality_id, view_id, and whatnot.
 &nbsp;The six bit pointer field can (or: is to be when possible) organized=
 by the encoder such that it is prudent for a middle box to throw away NAL =
units (belonging to layers) with higher values of the six bit field first, =
before throwing away NAL units with lower
 values. &nbsp;Relying on this feature, 3&#43;6 bits =3D=3D 9 bits should b=
e fine for the header extension.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">That said, the ordering by the encoder is just a recommen=
dation, and there may well be cases where different pruning strategies may =
be advisable. &nbsp;For example, a layering
 structure could be constructed that expands into two branches, one using 2=
D scalable tools only, the other including view_id for multi view coding. &=
nbsp;By looking at the six bit field alone, a middle box will not be able t=
o meaningfully remove NAL units belonging
 to one of the branches completely while pruning the other branch. &nbsp;In=
 order to meaningfully deal with that scenario, there would be two options:=
 one to represent the dimension_id[][] (and associated control info) in the=
 header extension, or require the middle
 box to have access to the VPS and be able to interpret its content. &nbsp;=
The further could take considerably more than 16 bits and we would be talki=
ng about a variable length data structure. &nbsp;The latter requires the mi=
ddle box to have state and a mechanism to
 intercept the VPS (through signaling=97as the encrypted in-band VPS would =
not be useful under the assumption that the middle box does not have the ke=
y to the media=97which is the motivation of the draft in the first place). =
&nbsp;I personally don't mind at all the
 second mechanism, as I'm a big fan of out-of-band parameter set transmissi=
on and any middle box must be in the signaling path anyway to meaningfully =
manipulate RTP. &nbsp;I do not like the first option due to its variable, a=
nd possibly substantial, overhead.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Stephan &nbsp;&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div style=3D"border:none;border-top:solid #B5C4DF
                          1.0pt;padding:3.0pt 0in 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size: 11pt; font-family: Cali=
bri, sans-serif; ">From:
</span></b><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif=
; ">Justin Uberti &lt;<a moz-do-not-send=3D"true" href=3D"mailto:juberti@go=
ogle.com">juberti@google.com</a>&gt;<br>
<b>Date: </b>Friday, 19 July, 2013 06:32 <br>
<b>To: </b>Bernard Aboba &lt;<a moz-do-not-send=3D"true" href=3D"mailto:ber=
nard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
<b>Cc: </b>&quot;<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org=
">avtext@ietf.org</a>&quot; &lt;<a moz-do-not-send=3D"true" href=3D"mailto:=
avtext@ietf.org">avtext@ietf.org</a>&gt;, &quot;<a moz-do-not-send=3D"true"=
 href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a moz-do-no=
t-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [rtcweb] Fwd: New Version Notification for draft-finebe=
rg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Agree those are the right codecs to design for. Since in =
each case there are fairly low limits on the number of supported layers (i.=
e. 3 spatial layers for SVC), I think
 it should be possible to pack the temporal, spatial, quality layer ids int=
o 16 bits.
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; "><o:p>&nbsp;</o:p></span></=
p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba &lt;<a moz=
-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_=
blank">bernard_aboba@hotmail.com</a>&gt; wrote:<o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">If we can support VP8/9 as well as H.264/5 SVC<o:p></o:p>=
</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">that would be a start. It seems doable to me.<o:p></o:p><=
/span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; "><br>
On Jul 18, 2013, at 8:34 PM, &quot;Adam Fineberg&quot; &lt;<a moz-do-not-se=
nd=3D"true" href=3D"mailto:fineberg@vline.me" target=3D"_blank">fineberg@vl=
ine.me</a>&gt; wrote:<o:p></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Bernard,<br>
<br>
Are there other codecs you are thinking should be supported?&nbsp; If it's =
generalized I would think we want to be able to cover all known scalable co=
decs. I'll look into the H264/SVC fields to see how to encode them in a gen=
eralized header.<br>
<br>
Regards,<br>
Adam<br>
<br>
On 7/18/13 7:40 PM, Bernard Aboba wrote:<o:p></o:p></span></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; ">I think it may be possible=
 to generalize this.&nbsp; For example, for H.264/SVC which can support tem=
poral, spatial and quality scalability, you
 would need the quality_id and dependency_id in addition to the temporal_id=
 (what you call the temporal layer index). &nbsp;&nbsp;
<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" style=3D"text-align:center" align=3D"center"><span=
 style=3D"font-size: 10.5pt; font-family: Calibri, sans-serif; ">
<hr align=3D"center" size=3D"2" width=3D"100%">
</span></div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; ">Date: Thu, 18 Jul 2013 08:=
45:38 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">
bernard_aboba@hotmail.com</a><br>
CC: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a>;
<a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org" target=3D"_blan=
k">avtext@ietf.org</a><br>
Subject: Re: [rtcweb] Fwd: New Version Notification for draft-fineberg-avte=
xt-temporal-layer-ext-00.txt<br>
<br>
Bernard,<br>
<br>
Good question.&nbsp; I'm not familiar enough with the parameter requirement=
s of all other scalable codecs to be able to generalize.&nbsp; If you'd lik=
e to help specify them, I'd be fine revising the draft to generalize.<br>
<br>
Regards,<br>
Adam<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">On 7/17/13 8:26 PM, Bernard Aboba wrote:<o:p></o:p></span=
></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Since the need is not codec specific (e.g. it arises with=
 any codec supporting temporal, spatial and quality scalability), why
<br>
&nbsp;a VP8-specific RTP extension? <br>
&nbsp;<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" style=3D"text-align:center" align=3D"center"><span=
 style=3D"font-size: 10.5pt; font-family: Calibri, sans-serif; ">
<hr align=3D"center" size=3D"2" width=3D"100%">
</span></div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; ">Date: Wed, 17 Jul 2013 17:09:46 -0700<br>
From: <a moz-do-not-send=3D"true" href=3D"mailto:fineberg@vline.me" target=
=3D"_blank">fineberg@vline.me</a><br>
To: <a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_=
blank">rtcweb@ietf.org</a><br>
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<br>
<br>
Hi,<br>
<br>
I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt
 the SRTP packets. This is undesirable both because the middle-box needs to=
 have access to the keys as well as the because of the added overhead of th=
e decrypt/encrypt cycle. This draft proposes an RTP header extension that w=
ill allow us to use the VP8 temporal
 layer information included in the header extension and therefore do forwar=
ding without SRTP decryption. Comments welcome.<br>
<br>
Regards,<br>
Adam Fineberg<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><a moz-do-not-send=3D"true" href=3D"mailto:fineberg%20at%=
20vline.com" target=3D"_blank">fineberg at vline.com</a><br>
<br>
-------- Original Message -------- <o:p></o:p></span></p>
<table class=3D"MsoNormalTable" border=3D"0" cellpadding=3D"0" cellspacing=
=3D"0">
<tbody>
<tr>
<td style=3D"padding:0in
                                                          0in 0in 0in" nowr=
ap=3D"nowrap" valign=3D"top">
<p class=3D"MsoNormal" style=3D"text-align:right" align=3D"right"><b>Subjec=
t:<o:p></o:p></b></p>
</td>
<td style=3D"padding:0in
                                                          0in 0in 0in">
<p class=3D"MsoNormal">New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt<o:p></o:p></p>
</td>
</tr>
<tr>
<td style=3D"padding:0in
                                                          0in 0in 0in" nowr=
ap=3D"nowrap" valign=3D"top">
<p class=3D"MsoNormal" style=3D"text-align:right" align=3D"right"><b>Date:<=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in
                                                          0in 0in 0in">
<p class=3D"MsoNormal">Tue, 09 Jul 2013 10:02:05 -0700<o:p></o:p></p>
</td>
</tr>
<tr>
<td style=3D"padding:0in
                                                          0in 0in 0in" nowr=
ap=3D"nowrap" valign=3D"top">
<p class=3D"MsoNormal" style=3D"text-align:right" align=3D"right"><b>From:<=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in
                                                          0in 0in 0in">
<p class=3D"MsoNormal"><a moz-do-not-send=3D"true" href=3D"mailto:internet-=
drafts%20at%20ietf.org" target=3D"_blank">internet-drafts at ietf.org</a><o=
:p></o:p></p>
</td>
</tr>
<tr>
<td style=3D"padding:0in
                                                          0in 0in 0in" nowr=
ap=3D"nowrap" valign=3D"top">
<p class=3D"MsoNormal" style=3D"text-align:right" align=3D"right"><b>To:<o:=
p></o:p></b></p>
</td>
<td style=3D"padding:0in
                                                          0in 0in 0in">
<p class=3D"MsoNormal">Adam Fineberg&nbsp;<a moz-do-not-send=3D"true" href=
=3D"mailto:fineberg%20at%20vline.com" target=3D"_blank">&lt;fineberg at vli=
ne.com&gt;</a><o:p></o:p></p>
</td>
</tr>
</tbody>
</table>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><br>
<br>
<br>
<o:p></o:p></span></p>
<pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<=
o:p></o:p></pre>
<pre>has been successfully submitted by Adam Fineberg and posted to the<o:p=
></o:p></pre>
<pre>IETF repository.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  draft-fineberg-av=
text-temporal-layer-ext<o:p></o:p></pre>
<pre>Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  00<o:p></o:p></pr=
e>
<pre>Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  A =
Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer =
Information<o:p></o:p></pre>
<pre>Creation date:&nbsp;&nbsp;  2013-07-08<o:p></o:p></pre>
<pre>Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  In=
dividual Submission<o:p></o:p></pre>
<pre>Number of pages: 6<o:p></o:p></pre>
<pre>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &nbsp;<a moz-do-not-send=3D"true" href=3D"http://www.ietf.org/internet-dra=
fts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target=3D"_blank">http=
://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00=
.txt</a><o:p></o:p></pre>
<pre>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a moz-d=
o-not-send=3D"true" href=3D"http://datatracker.ietf.org/doc/draft-fineberg-=
avtext-temporal-layer-ext" target=3D"_blank">http://datatracker.ietf.org/do=
c/draft-fineberg-avtext-temporal-layer-ext</a><o:p></o:p></pre>
<pre>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;<a moz-do-not-send=
=3D"true" href=3D"http://tools.ietf.org/html/draft-fineberg-avtext-temporal=
-layer-ext-00" target=3D"_blank">http://tools.ietf.org/html/draft-fineberg-=
avtext-temporal-layer-ext-00</a><o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Abstract:<o:p></o:p></pre>
<pre>&nbsp;&nbsp; This document defines a mechanism by which packets of Rea=
l-Time<o:p></o:p></pre>
<pre>&nbsp;&nbsp; Tranport Protocol (RTP) video streams encoded with the VP=
8 codec can<o:p></o:p></pre>
<pre>&nbsp;&nbsp; indicate, in an RTP header extension, the temporal layer =
information<o:p></o:p></pre>
<pre>&nbsp;&nbsp; about the frame encoded in the RTP packet.&nbsp; This inf=
ormation can be<o:p></o:p></pre>
<pre>&nbsp;&nbsp; used in a middlebox performing bandwidth management of st=
reams<o:p></o:p></pre>
<pre>&nbsp;&nbsp; without requiring it to decrypt the streams.<o:p></o:p></=
pre>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><br>
_______________________________________________ rtcweb mailing list <a moz-=
do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">
rtcweb@ietf.org</a> <a moz-do-not-send=3D"true" href=3D"https://www.ietf.or=
g/mailman/listinfo/rtcweb" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
</blockquote>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize: 10.5pt; font-family: Calibri, sans-serif; "><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org" target=3D"_blan=
k">rtcweb@ietf.org</a><br>
<a moz-do-not-send=3D"true" href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o=
:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><br>
<br>
<o:p></o:p></span></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>avtext mailing list<o:p></o:p></pre>
<pre><a moz-do-not-send=3D"true" href=3D"mailto:avtext@ietf.org">avtext@iet=
f.org</a><a moz-do-not-send=3D"true" href=3D"https://www.ietf.org/mailman/l=
istinfo/avtext" target=3D"_blank">https://www.ietf.org/mailman/listinfo/avt=
ext</a><o:p></o:p></pre>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size: 10.5pt; font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Regards,<o:p></o:p></pre>
<pre>Adam<o:p></o:p></pre>
</div>
</div>
</blockquote>
<br>
<pre class=3D"moz-signature" cols=3D"72">--=20
Regards,
Adam</pre>
</div>
</div>
</span>
</body>
</html>

--_000_CE169EA79FC3Cstewesteweorg_--

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Subject: [rtcweb] Call for scribes for IETF87
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If you are able to take minutes at IETF 87, please contact the chairs; we
would ideally like two minute-takers per session.  We will also be looking
for folks to channel the jabber session to the room.

thanks,

Ted

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If you are able to take minutes at IETF 87, please contact the chairs; we w=
ould ideally like two minute-takers per session.=A0 We will also be looking=
 for folks to channel the jabber session to the room.<br><br>thanks,<br><br=
>
Ted<br>

--047d7ba97972096bac04e2576009--

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From: Jonathan Lennox <jonathan@vidyo.com>
To: Adam Fineberg <fineberg@vline.me>, "avtext@ietf.org" <avtext@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 25 Jul 2013 12:49:25 -0400
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--_000_C3759687E4991243A1A0BD44EAC823034E10AF5568BE235maillan_--

From martin.thomson@gmail.com  Thu Jul 25 11:33:05 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [rtcweb] Security Architecture -07 review
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Just a short list of niggles and nits.  I only reviewed the diff.

The abstract has been truncated.

Section 4.1, last paragraph on p11, s/[RFC6455]./[RFC6455].,/

Section 5.1
   [...]  Implementations MUST either
   choose to terminate the call or display a warning at that point.

I don't recall the discussion that lead to this conclusion.  I'm still
in the "stop sending" camp on this.  I understand that browsers
already require machinery that notifies them of when a page
transitions to becoming mixed content, because the lock icon tends to
disappear on gmail all the time.  So the only technical concern I can
think of is the fact that this will appear as though there was a
sudden massive packet loss event.  I believe that's workable, as long
as RTCP and connectivity checks continue.

Section 5.2:
   [...]  Note that
   it is unlikely that browsers would have an X.509 certificate, but
   servers might.

This is a little over-simplified.  1.  Servers will have an X.509, and
that will need to be trusted somehow.  2.  Clients will have X.509
certificates or else DTLS won't work, it's just that it's unlikely to
be trusted for authentication of a domain name.

I hope that we intend to talk about screen sharing at some point.
That seems to be a bit of a big hole here.  Even if all the mechanisms
described are implemented, I don't believe that to be sufficient for
this to be deployed.   I tend to think that there needs to be a way to
either opt-out (or preferably opt-in) of screen sharing.  This
probably crosses into W3C territory a bit too.

Section 5.4:
   [...]  Either such
   enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or
   the JS [...]

The TURN only option at the browser level might be possible, but this
is pretty unrealistic.

Section 5.5:
   Unless the user specifically configures an
   external key pair, different key pairs MUST be used for each
   origin. (This avoids creating a super-cookie.)

The "unless" here is interesting.  Why do you believe that caveat to
be necessary?

On a related note, we had some offline discussions about what it means
to use the same key pair over time for the same site.  Early Firefox
implementations used a new key pair for every RTCPeerConnection,
which, aside from being a little CPU-intensive, tends to make it
impossible to audit calls after the fact.  That is, it is useful to be
able to ask the person you thought you were talking to for their
certificate fingerprint to ensure that you really were talking to that
person.

However, the cookie issue is still a problem.  Removing the
certificate and key pair when cookies are cleared is necessary.

   [...]  However, if Null ciphers are used, that MUST be presented to
   the user at the top-level UI.

At what point did we decide that the Null cipher was acceptable?  I
thought that TLS pretty much prohibited the negotiation of the null
cipher.  I certainly wouldn't want to imply that this is an acceptable
thing to do.

From martin.thomson@gmail.com  Thu Jul 25 11:59:53 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [rtcweb] Security -05 review
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I should have read this before security-arch, because it mentions the
need to remove persistent identifiers.  It probably needs to be a
little more specific than it is though.

I also note a typo: s/Gmail/email/  (let the RFC editor add hyphens or
capitalization or whatever, I can never work out how to type that one)

Otherwise, the changes look OK to me.

From martin.thomson@gmail.com  Thu Jul 25 13:47:36 2013
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On 25 July 2013 11:32, Martin Thomson <martin.thomson@gmail.com> wrote:
> Section 5.2:
>    [...]  Note that
>    it is unlikely that browsers would have an X.509 certificate, but
>    servers might.
>
> This is a little over-simplified.  1.  Servers will have an X.509, and
> that will need to be trusted somehow.  2.  Clients will have X.509
> certificates or else DTLS won't work, it's just that it's unlikely to
> be trusted for authentication of a domain name.

Expanding on this a little, with a text proposal.  Sadly, this didn't
turn out to be easy.

OLD:
   In addition, they SHOULD support requests for access that promise
   that media from this grant will be sent to a single communicating
   peer (obviously there could be other requests for other peers).
   E.g., "Call customerservice@ford.com".
NEW:
   In addition, browsers SHOULD ...
(Editorial only)

OLD:
   The semantics of this request
   are that the media stream from the camera and microphone will only be
   routed through a connection which has been cryptographically verified
   (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP
   handshake) as being associated with the stated identity.
NEW:
   An application that requests access that is limited in this way cannot
   access or modify the acquired media or send media to any destination
   other than the specified identity.  The browser will prevent the media
   from being transmitted over a connection which has a cryptographically
   verified identity (see Section !).
(I don't think that it's necessary to do anything more than cite the
appropriate section.  I would have liked to identify a
peer-authentication section here, but Section 5.6 deals only with the
IdP stuff.  If this were my document, I'd promote 5.6 to a new
top-level section and add a new 5.6 that deals with peer
authentication of both forms.  The existing text regarding domain
name-based authentication is light-to-non-existent.

e.g.
5.6. Peer Authentication

   <First paragraph of the current 5.6 is pretty good>

   Establishing the identity of a communication peer requires that DTLS-SRTP
   be used with X.509 certificates being presented by both peers.  Two
mechanisms
   are then possible:

   Identity provider: Most browsers do not customarily have domain names and
      X.509 certificates for those domains.  [The new] Section 6
describes a mechanism
      whereby a cryptographic assertion binding a self-signed X.509
certificate to
      an identity specific to an identity prover can be created and verified.

   Domain name: An X.509 certificate that contains a subjectAltName containing
      a domain name can be verified using the procedures defined in RFC 6125.
      This option is, however, only likely to be feasible for organizations that
      operate servers, since it depends on use of a domain name.

   <existing paragraph>
   Whatever the underlying technology, the general principle is that the
   party which is being authenticated is NOT the signaling site [...]

Feel free to completely rewrite, or ignore this, I'm not across all
the sensitivities here.  I do think that the IdP stuff is big enough
to get its own section.  I actually still think that it's a big enough
concept that a dedicated document would have been a better choice,
still....

Another possibility is to deal with this a little better in Section
3.1, which I note could use a better term than "Other users" to refer
to media/real-time peers.)

REMOVE:
   Note that
   it is unlikely that browsers would have an X.509 certificate, but
   servers might.
KEEP?:
   Browsers servicing such requests SHOULD clearly
   indicate that identity to the user when asking for permission.  The
   idea behind this type of permissions is that a user might have a
   fairly narrow list of peers he is willing to communicate with, e.g.,
   "my mother" rather than "anyone on Facebook".  Narrow permissions
   grants allow the browser to do that enforcement.
(I'm sure about the 2119 SHOULD here.  I think that there are several
ways to think about this.  For the most part,  I believe that there's
no need to make any sort of normative statement.  Absence of an
indication isn't going to disadvantage users, they just lose the
opportunity to learn about restrictions.  Non-normative text that
suggests the UI is a different matter: applications will appreciate
any feature that increases the chance of a user hitting "allow".)

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On Jul 25, 2013, at 11:32 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> Just a short list of niggles and nits.  I only reviewed the diff.
>=20
> The abstract has been truncated.
>=20
> Section 4.1, last paragraph on p11, s/[RFC6455]./[RFC6455].,/
>=20
> Section 5.1
>   [...]  Implementations MUST either
>   choose to terminate the call or display a warning at that point.
>=20
> I don't recall the discussion that lead to this conclusion.  I'm still
> in the "stop sending" camp on this.  I understand that browsers
> already require machinery that notifies them of when a page
> transitions to becoming mixed content, because the lock icon tends to
> disappear on gmail all the time.  So the only technical concern I can
> think of is the fact that this will appear as though there was a
> sudden massive packet loss event.  I believe that's workable, as long
> as RTCP and connectivity checks continue.
>=20
> Section 5.2:
>   [...]  Note that
>   it is unlikely that browsers would have an X.509 certificate, but
>   servers might.
>=20
> This is a little over-simplified.  1.  Servers will have an X.509, and
> that will need to be trusted somehow.  2.  Clients will have X.509
> certificates or else DTLS won't work, it's just that it's unlikely to
> be trusted for authentication of a domain name.
>=20
> I hope that we intend to talk about screen sharing at some point.
> That seems to be a bit of a big hole here.  Even if all the mechanisms
> described are implemented, I don't believe that to be sufficient for
> this to be deployed.   I tend to think that there needs to be a way to
> either opt-out (or preferably opt-in) of screen sharing.  This
> probably crosses into W3C territory a bit too.
>=20
> Section 5.4:
>   [...]  Either such
>   enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or
>   the JS [...]
>=20
> The TURN only option at the browser level might be possible, but this
> is pretty unrealistic.
>=20
> Section 5.5:
>   Unless the user specifically configures an
>   external key pair, different key pairs MUST be used for each
>   origin. (This avoids creating a super-cookie.)
>=20
> The "unless" here is interesting.  Why do you believe that caveat to
> be necessary?
>=20
> On a related note, we had some offline discussions about what it means
> to use the same key pair over time for the same site.  Early Firefox
> implementations used a new key pair for every RTCPeerConnection,
> which, aside from being a little CPU-intensive, tends to make it
> impossible to audit calls after the fact.  That is, it is useful to be
> able to ask the person you thought you were talking to for their
> certificate fingerprint to ensure that you really were talking to that
> person.
>=20
> However, the cookie issue is still a problem.  Removing the
> certificate and key pair when cookies are cleared is necessary.

Wrapping the DTLS handshake inside a DH exchange would achieve both =
goals (preventing a super-cookie from being passively observed, as well =
as giving key continuity).

-d



>=20
>   [...]  However, if Null ciphers are used, that MUST be presented to
>   the user at the top-level UI.
>=20
> At what point did we decide that the Null cipher was acceptable?  I
> thought that TLS pretty much prohibited the negotiation of the null
> cipher.  I certainly wouldn't want to imply that this is an acceptable
> thing to do.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 25 July 2013 14:54, Dan Wing <dwing@cisco.com> wrote:
>> However, the cookie issue is still a problem.  Removing the
>> certificate and key pair when cookies are cleared is necessary.
>
> Wrapping the DTLS handshake inside a DH exchange would achieve both goals (preventing a super-cookie from being passively observed, as well as giving key continuity).

For passive observers, yes, but I'm not that concerned about them.
After all, they are seeing the flow, so can make all sorts of
inferences based on the flow addressing.

The concern is the first party, who gets the certificate.  Using DH
isn't going to fix that.

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On Jul 25, 2013, at 4:12 PM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 25 July 2013 14:54, Dan Wing <dwing@cisco.com> wrote:
>>> However, the cookie issue is still a problem.  Removing the
>>> certificate and key pair when cookies are cleared is necessary.
>>=20
>> Wrapping the DTLS handshake inside a DH exchange would achieve both =
goals (preventing a super-cookie from being passively observed, as well =
as giving key continuity).
>=20
> For passive observers, yes, but I'm not that concerned about them.
> After all, they are seeing the flow, so can make all sorts of
> inferences based on the flow addressing.

But that doesn't mean we should purposefully leak even more information =
at another layer.

> The concern is the first party, who gets the certificate.  Using DH
> isn't going to fix that.

Then you're arguing for "endpoint MUST NOT change its certificate".

-d


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On 25 July 2013 17:00, Dan Wing <dwing@cisco.com> wrote:
>> The concern is the first party, who gets the certificate.  Using DH
>> isn't going to fix that.
>
> Then you're arguing for "endpoint MUST NOT change its certificate".

Nope.  There are reasons to maintain a stable certificate (being able
to audit old calls is helped by this, though a log of fingerprints
used for calls over time might do the same, albeit a little less
transparently; performance is another reason), just as there are
reasons to change the certificate (privacy, mainly an ability to keep
peers from linking calls).

And yes, generally DH > not DH, but I'm not sure the benefits are
huge, depending on what your threat model is.  I was just pointing out
that it doesn't actually help for the cases that I was really
concerned about.  The women's shelter case in particular.

From dwing@cisco.com  Thu Jul 25 17:34:33 2013
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On Jul 25, 2013, at 5:12 PM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

No comment on leaking information at other layers?  We just improved the =
RTCP CNAME to remove its leakage (draft-ietf-avtcore-6222bis), as an =
example of reducing -- rather than expanding -- personally-identifyable =
super-cookies.  The DTLS handshake is over the same media path improved =
by 6222bis.

> On 25 July 2013 17:00, Dan Wing <dwing@cisco.com> wrote:
>>> The concern is the first party, who gets the certificate.  Using DH
>>> isn't going to fix that.
>>=20
>> Then you're arguing for "endpoint MUST NOT change its certificate".
>=20
> Nope.  There are reasons to maintain a stable certificate (being able
> to audit old calls is helped by this, though a log of fingerprints
> used for calls over time might do the same, albeit a little less
> transparently; performance is another reason), just as there are
> reasons to change the certificate (privacy, mainly an ability to keep
> peers from linking calls).
>=20
> And yes, generally DH > not DH, but I'm not sure the benefits are
> huge, depending on what your threat model is.  I was just pointing out
> that it doesn't actually help for the cases that I was really
> concerned about.  The women's shelter case in particular.

Thanks for the explanation.  I concur the public/private key needs a way =
to be destroyed and a new one generated.=20

We could envision a more sophisticated mechanism, based on lots of =
different criteria, where different public/private key pairs are used =
and where some are destroyed immediately after certain calls.  For =
example, I might want to use a public/private key pair for all of my =
work calls and leave that static (as I want that identity to stick with =
me) but an ephemeral, one-time-only public/private key pair when not =
engaging in phone calls related to my work, such as calling my =
psychologist.

-d


From fineberg@vline.me  Thu Jul 25 20:35:17 2013
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Jonathan,

Thanks for the feedback.  You are of course correct, I'm not sure why I 
forgot about the authentication of the RTP header.  The intent was not 
to have to perform the crypto on the payload of course but it would 
still require header changes which therefore needs the header 
re-authenticated.

Regards,
Adam

On 7/25/13 9:49 AM, Jonathan Lennox wrote:
>
> Hi, Adam â€“
>
> (Speaking as an individual.)
>
> I think thereâ€™s a problem with the use case as youâ€™ve described here, 
> which I think calls into question the usefulness of the whole mechanism.
>
> As I understand you, youâ€™re planning to send a single VP8 stream 
> containing all the layers, and then have a media-aware middlebox 
> selectively drop some of the streamâ€™s packets to do temporal scaling 
> without needing the middlebox to be in the crypto context.
>
> Unfortunately, RTP doesnâ€™t really let you do this.  The problem is 
> that by just dropping packets without rewriting RTP sequence numbers 
> or RTCP sender reports, this is going to look like enormous packet 
> loss to RTP (which, in a sense, it is).  A middlebox like this â€“ which 
> is acting as a media-modifying RTP translator, in RTP terminology â€“ 
> needs to rewrite RTP headers and RTCP sender reports in order to keep 
> the stream and its metadata consistent and valid.
>
> Unfortunately, the translator needs to have the crypto keys in order 
> to make these modifications, so Iâ€™m not sure this header extension 
> adds much benefit over just validating the authentication, decrypting 
> (say) the initial block of the packet (wherever the temporal layer 
> information is known to be), modifying the headers, and then 
> re-authenticating the packet.
>
> Is there something Iâ€™m missing here?
>
> *From:*Adam Fineberg [mailto:fineberg@vline.me]
> *Sent:* Wednesday, July 17, 2013 6:37 PM
> *To:* avtext@ietf.org
> *Subject:* [avtext] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> Hi,
>
> I'm working on WebRTC services and have found that while developing 
> services that forward VP8 video streams if we want to take advantage 
> of the VP8 temporal scaling we must get the temporal layer information 
> from the RTP header which requires us to decrypt the SRTP packets.ï¿½Âï¿½ 
> This is undesirable both because the middlebox needs to have acesss to 
> the keys as well as the because of the added overhead of the 
> decrypt/encrypt cycle.ï¿½Âï¿½ This draft proposes an RTP header extension 
> that will allow us to use the VP8 temporal layer information included 
> in the header extension and therefore do forwarding without SRTP 
> decryption.ï¿½Âï¿½ Comments welcome.
>
> Regards,
> Adam Fineberg
>
> fineberg@vline.com <mailto:fineberg@vline.com>
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> *Date: *
>
> 	
>
> Tue, 09 Jul 2013 10:02:05 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> Adam Fineberg <fineberg@vline.com> <mailto:fineberg@vline.com>
>
> A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
> has been successfully submitted by Adam Fineberg and posted to the
> IETF repository.
>   
> Filename:       draft-fineberg-avtext-temporal-layer-ext
> Revision:       00
> Title:          A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
> Creation date:  2013-07-08
> Group:          Individual Submission
> Number of pages: 6
> URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
> Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
> Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>   
>   
> Abstract:
>     This document defines a mechanism by which packets of Real-Time
>     Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>     indicate, in an RTP header extension, the temporal layer information
>     about the frame encoded in the RTP packet.  This information can be
>     used in a middlebox performing bandwidth management of streams
>     without requiring it to decrypt the streams.
>   
>                                                                                    
>   
>   
> The IETF Secretariat
>   
>


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    <div class="moz-cite-prefix">Jonathan,<br>
      <br>
      Thanks for the feedback.Â  You are of course correct, I'm not sure
      why I forgot about the authentication of the RTP header.Â  The
      intent was not to have to perform the crypto on the payload of
      course but it would still require header changes which therefore
      needs the header re-authenticated.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/25/13 9:49 AM, Jonathan Lennox wrote:<br>
    </div>
    <blockquote
      cite="mid:C3759687E4991243A1A0BD44EAC823034E10AF5568@BE235.mail.lan"
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,
            Adam â€“<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">(Speaking
            as an individual.)<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            think thereâ€™s a problem with the use case as youâ€™ve
            described here, which I think calls into question the
            usefulness of the whole mechanism.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">As
            I understand you, youâ€™re planning to send a single VP8
            stream containing all the layers, and then have a
            media-aware middlebox selectively drop some of the streamâ€™s
            packets to do temporal scaling without needing the middlebox
            to be in the crypto context. <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Unfortunately,
            RTP doesnâ€™t really let you do this.Â  The problem is that by
            just dropping packets without rewriting RTP sequence numbers
            or RTCP sender reports, this is going to look like enormous
            packet loss to RTP (which, in a sense, it is).Â  A middlebox
            like this â€“ which is acting as a media-modifying RTP
            translator, in RTP terminology â€“ needs to rewrite RTP
            headers and RTCP sender reports in order to keep the stream
            and its metadata consistent and valid.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Unfortunately,
            the translator needs to have the crypto keys in order to
            make these modifications, so Iâ€™m not sure this header
            extension adds much benefit over just validating the
            authentication, decrypting (say) the initial block of the
            packet (wherever the temporal layer information is known to
            be), modifying the headers, and then re-authenticating the
            packet.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Is
            there something Iâ€™m missing here?<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â 
            <o:p></o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0in 0in 0in">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                Adam Fineberg [<a class="moz-txt-link-freetext" href="mailto:fineberg@vline.me">mailto:fineberg@vline.me</a>] <br>
                <b>Sent:</b> Wednesday, July 17, 2013 6:37 PM<br>
                <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:avtext@ietf.org">avtext@ietf.org</a><br>
                <b>Subject:</b> [avtext] Fwd: New Version Notification
                for draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>Â </o:p></p>
        <p class="MsoNormal">Hi,<br>
          <br>
          I'm working on WebRTC services and have found that while
          developing services that forward VP8 video streams if we want
          to take advantage of the VP8 temporal scaling we must get the
          temporal layer information from the RTP header which requires
          us to decrypt the SRTP packets.<span
            style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>Â<span
style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>
          This is undesirable both because the middlebox needs to have
          acesss to the keys as well as the because of the added
          overhead of the decrypt/encrypt cycle.<span
            style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>Â<span
style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>
          This draft proposes an RTP header extension that will allow us
          to use the VP8 temporal layer information included in the
          header extension and therefore do forwarding without SRTP
          decryption.<span
            style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>Â<span
style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">ï¿½</span>
          Comments welcome.<br>
          <br>
          Regards,<br>
          Adam Fineberg<o:p></o:p></p>
        <div>
          <p class="MsoNormal"><a moz-do-not-send="true"
              href="mailto:fineberg@vline.com">fineberg@vline.com</a><br>
            <br>
            -------- Original Message -------- <o:p></o:p></p>
          <table class="MsoNormalTable" border="0" cellpadding="0"
            cellspacing="0">
            <tbody>
              <tr>
                <td style="padding:0in 0in 0in 0in" nowrap="nowrap"
                  valign="top">
                  <p class="MsoNormal" style="text-align:right"
                    align="right"><b>Subject: <o:p></o:p></b></p>
                </td>
                <td style="padding:0in 0in 0in 0in">
                  <p class="MsoNormal">New Version Notification for
                    draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></p>
                </td>
              </tr>
              <tr>
                <td style="padding:0in 0in 0in 0in" nowrap="nowrap"
                  valign="top">
                  <p class="MsoNormal" style="text-align:right"
                    align="right"><b>Date: <o:p></o:p></b></p>
                </td>
                <td style="padding:0in 0in 0in 0in">
                  <p class="MsoNormal">Tue, 09 Jul 2013 10:02:05 -0700<o:p></o:p></p>
                </td>
              </tr>
              <tr>
                <td style="padding:0in 0in 0in 0in" nowrap="nowrap"
                  valign="top">
                  <p class="MsoNormal" style="text-align:right"
                    align="right"><b>From: <o:p></o:p></b></p>
                </td>
                <td style="padding:0in 0in 0in 0in">
                  <p class="MsoNormal"><a moz-do-not-send="true"
                      href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                </td>
              </tr>
              <tr>
                <td style="padding:0in 0in 0in 0in" nowrap="nowrap"
                  valign="top">
                  <p class="MsoNormal" style="text-align:right"
                    align="right"><b>To: <o:p></o:p></b></p>
                </td>
                <td style="padding:0in 0in 0in 0in">
                  <p class="MsoNormal">Adam Fineberg <a
                      moz-do-not-send="true"
                      href="mailto:fineberg@vline.com">&lt;fineberg@vline.com&gt;</a><o:p></o:p></p>
                </td>
              </tr>
            </tbody>
          </table>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p>Â </o:p></p>
          <pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></pre>
          <pre>has been successfully submitted by Adam Fineberg and posted to the<o:p></o:p></pre>
          <pre>IETF repository.<o:p></o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre>Filename:Â Â Â Â Â   draft-fineberg-avtext-temporal-layer-ext<o:p></o:p></pre>
          <pre>Revision:Â Â Â Â Â   00<o:p></o:p></pre>
          <pre>Title:Â Â Â Â Â Â Â Â   A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information<o:p></o:p></pre>
          <pre>Creation date:  2013-07-08<o:p></o:p></pre>
          <pre>Group:Â Â Â Â Â Â Â Â   Individual Submission<o:p></o:p></pre>
          <pre>Number of pages: 6<o:p></o:p></pre>
          <pre>URL:Â Â Â Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a><o:p></o:p></pre>
          <pre>Status:Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a><o:p></o:p></pre>
          <pre>Htmlized:Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a><o:p></o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre>Abstract:<o:p></o:p></pre>
          <pre>Â Â  This document defines a mechanism by which packets of Real-Time<o:p></o:p></pre>
          <pre>Â Â  Tranport Protocol (RTP) video streams encoded with the VP8 codec can<o:p></o:p></pre>
          <pre>Â Â  indicate, in an RTP header extension, the temporal layer information<o:p></o:p></pre>
          <pre>Â Â  about the frame encoded in the RTP packet.Â  This information can be<o:p></o:p></pre>
          <pre>Â Â  used in a middlebox performing bandwidth management of streams<o:p></o:p></pre>
          <pre>Â Â  without requiring it to decrypt the streams.<o:p></o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre> Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â <o:p></o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre><o:p>Â </o:p></pre>
          <pre>The IETF Secretariat<o:p></o:p></pre>
          <pre><o:p>Â </o:p></pre>
        </div>
        <p class="MsoNormal"><o:p>Â </o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
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From fineberg@vline.me  Fri Jul 26 06:14:32 2013
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Stephan,

This seems like a reasonable approach.  I like your point about 
designing the RTP stack to generically handle scalability features.  My 
concern is complexity and overhead for what should be a very lightweight 
process however generalizing seems to be the consensus so that should be 
the direction forward.

Regards,
Adam

On 7/25/13 12:44 AM, Stephan Wenger wrote:
> Hi,
> I thought about this a bit more.  I think that one sensible document 
> structure may be as follows:
>
> (1) generic structure of scalability/pruning information to an RTP 
> header extension (to be worked on in avtext).
> (2) mapping of codec-specific features to the generic structure.  As a 
> location, I would suggest that (for new codecs) the RTP payload specs 
> are the right document.  For codecs with existing RTP payload specs, a 
> short doc that updates the payload spec would be the right place, or a 
> revision of the payload spec itself if that needs to be done anyway. 
>  This would bundle all the codec-specifics to one document.
>
> This document structure, and the philosophy behind it, would allow the 
> design of a generic RTP stack that could take advantage of scalability 
> features independently of the scalable codec actually in use.  I 
> believe that such an advantage outweighs the possible advantage of 
> quick standardization of a more specific approach as proposed in 
> Adam's draft.  That said, we need to get the design of the generic 
> structure right such that it is indeed useful, ideally for all present 
> and hopefully for a large subset of future codecs.
>
> As for the specs involved:
> -H.264 SVC, H.265, and VP8 have payload format RFCs or reasonably 
> stable drafts.
> -No one uses H.263 and MPEG-2 scalability, so we can ignore those two.
> -As Adam, I'm also not aware of a VP9 payload spec draft.  However, my 
> understanding is that VP9 will include a number of scalability 
> features enabled by multiple reference pictures and reference picture 
> resampling.  The latter is a somewhat different approach compared to 
> the traditional scalability implementations.  I think that google 
> planned to freeze the VP9 bitstream a few weeks go, but I don't know 
> whether that has happened.  Without a stable spec and/or input from 
> google/webm folks it will indeed be hard to address VP9's specific 
> needs, if any.
> -There are a bunch of audio codecs that claim to be scalable.  We 
> should have a scope discussion on whether those need to be included here.
>
> Stephan
>
>
>
> From: Adam Fineberg <fineberg@vline.me <mailto:fineberg@vline.me>>
> Date: Thursday, 25 July, 2013 00:54
> To: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com 
> <mailto:yekuiw@qti.qualcomm.com>>
> Cc: Bernard Aboba <bernard_aboba@hotmail.com 
> <mailto:bernard_aboba@hotmail.com>>, "avtext@ietf.org 
> <mailto:avtext@ietf.org>" <avtext@ietf.org <mailto:avtext@ietf.org>>, 
> "rtcweb@ietf.org <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org 
> <mailto:rtcweb@ietf.org>>, Justin Uberti <juberti@google.com 
> <mailto:juberti@google.com>>
> Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for 
> draft-fineberg-avtext-temporal-layer-ext-00.txt
>
> YK,
>
> I would appreciate your collaboration.  Which codecs are you referring 
> to when you say "all existing standard scalable video codecs"?
>
> Regards,
> Adam
>
> On 7/24/13 3:32 PM, Wang, Ye-Kui wrote:
>>
>> If the group is to specify something generic, naturally it should be 
>> generic enough to cover at least all existing standard scalable video 
>> codecs if possible. And I personally think that is possible and not 
>> difficult at all. Thus, why limit to only a few scalable video codecs?
>>
>> I could provide some help here too if needed.
>>
>> BR, YK
>>
>> *From:*avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] *On 
>> Behalf Of *Adam Fineberg
>> *Sent:* Wednesday, July 24, 2013 10:08 AM
>> *To:* Bernard Aboba
>> *Cc:* avtext@ietf.org; rtcweb@ietf.org; Justin Uberti
>> *Subject:* Re: [avtext] [rtcweb] Fwd: New Version Notification for 
>> draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>> Bernard,
>>
>> I apologize if I come across as being difficult here but I stil am 
>> not seeing the benefits.  Since the fields are not the same for the 
>> codecs, we will be multiplexing the bits and that seems to me to add 
>> complexity rather than add clarity.  Also, I can't find an IETF VP9 
>> document for the payload format to reference.  If the group thinks 
>> generalization is the right approach I would appreciate some 
>> collaboration on getting the right bit definitions for the other codecs.
>>
>> Regards,
>> Adam
>>
>> On 7/23/13 12:07 PM, Bernard Aboba wrote:
>>
>>     I do not think it is necessary to "support all forms of
>>     scalability for all codecs".   In fact, I would make that an
>>     explicit "non-goal".  All that was suggested is to try to create
>>     a single extension that supports a few common cases.   If it is
>>     possible to handle VP8, VP9 and H.264/SVC in a single extension
>>     that would be sufficient. So why not limit it to that?
>>
>>     ------------------------------------------------------------------------
>>
>>     Date: Tue, 23 Jul 2013 08:53:45 -0700
>>     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>     To: stewe@stewe.org <mailto:stewe@stewe.org>
>>     CC: juberti@google.com <mailto:juberti@google.com>;
>>     bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>;
>>     avtext@ietf.org <mailto:avtext@ietf.org>; rtcweb@ietf.org
>>     <mailto:rtcweb@ietf.org>
>>     Subject: Re: [avtext] [rtcweb] Fwd: New Version Notification for
>>     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>     I've been thinking about this and given the ease at which RFC5285
>>     allows for the specification of a header extension and the
>>     complexity introduced by trying to generalize the header
>>     extension to support all forms of scalability for all codecs that
>>     the generalization might not be the best approach.  I'm not sure
>>     what we really gain by trying to capture all this in a single
>>     header extension rather than one per that can succinctly explain
>>     the fields without the complexity of multiplexing the bits.
>>
>>     Thoughts?
>>
>>     Regards,
>>     Adam
>>
>>     On 7/19/13 3:44 PM, Stephan Wenger wrote:
>>
>>         Hi,
>>
>>         *From: *Adam Fineberg <fineberg@vline.me
>>         <mailto:fineberg@vline.me>>
>>         *Date: *Friday, 19 July, 2013 15:12
>>         *To: *Stephan Wenger <stewe@stewe.org <mailto:stewe@stewe.org>>
>>         *Cc: *Justin Uberti <juberti@google.com
>>         <mailto:juberti@google.com>>, Bernard Aboba
>>         <bernard_aboba@hotmail.com
>>         <mailto:bernard_aboba@hotmail.com>>, "avtext@ietf.org
>>         <mailto:avtext@ietf.org>" <avtext@ietf.org
>>         <mailto:avtext@ietf.org>>, "rtcweb@ietf.org
>>         <mailto:rtcweb@ietf.org>" <rtcweb@ietf.org
>>         <mailto:rtcweb@ietf.org>>
>>         *Subject: *Re: [avtext] [rtcweb] Fwd: New Version
>>         Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>         Stephan,
>>
>>         Thanks for the info and the reference.  I'm not sure I follow
>>         as I'm not at all familiar with H.265. I'll review the
>>         reference and see what I can figure.
>>
>>         StW: Good luck :-)
>>
>>         It seems though to me that you are suggesting that except in
>>         the simple case, that the data for H.265 would not be well
>>         suited to a header extension, am I understanding you
>>         correctly?  There is no reason the middlebox couldn't get out
>>         of band signaling of the VPS as you mention but that would
>>         not be within the scope of this header extension.
>>
>>         StW: well, if you would copy the layer_id into your header
>>         extension (just as you need to do for the simple case), a
>>         really smart middle box could use this information just as a
>>         decoder uses it, assuming that it intercepted the VPS in the
>>         first place.  Insofar, I wouldn't rule out the second option
>>         on technical grounds.  Whether any of the actual products
>>         would bother to do that, ever, is another question.  I think
>>         the case ought to be documented, though.  I can help drafting
>>         text.
>>
>>         While we are at it: doing this right could mean that you need
>>         multiple specs.  First, a generic header extension mechanism
>>         dedicated to side information required for pruning of RTP
>>         packet streams—ideally not only for scalable video, although
>>         that is the main customer today.  And second, for each
>>         "payload" (at present we are talking about H.264/SVC, H.265v1
>>         (HEVC), H.265v2 (including scalable and 3D extensions, which
>>         are not yet finalized), VP8, and VP9 (I know little about the
>>         latter), plus Daala, and whatnot) a mapping of the bits
>>         available in the generic header extension to the bits in the
>>         payload itself (NAL header and VPS in case of H.265, NALU
>>         header in SVC, and the fields you mention for VP8).
>>
>>         Stephan
>>
>>
>>
>>         Any insights are appreciated.
>>
>>         Regards,
>>         Adam
>>
>>         On 7/19/13 8:33 AM, Stephan Wenger wrote:
>>
>>             Hi,
>>
>>             I also believe that 16 bits should be enough.  For H.264
>>             and VP8 that has already been demonstrated.  For H.265,
>>             some initial thoughts below.  Apologies for the word-count.
>>
>>             The scalable version of H265 (called SHVC) is currently
>>             under development.  The current working draft can be
>>             found here:
>>             http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip.
>>              Therein, the options for defining layering structures
>>             are considerably more complex.  To start, we have 3 bits
>>             for the temporal ID in the NAL unit header of the H.265
>>             version 1 (HEVC) base specification (temporal scalability
>>             is already nicely supported in version 1).  Just like in
>>             SVC.  In the scalable extension, the NAL unit header
>>             contains a six bit field that points into a data
>>             structure known as "Video Parameter Set" (VPS).  Inside
>>             the VPS, those six bits are mapped to to a position in a
>>             directed graph (specified through "dimension_id[][]"),
>>             which tells you about the reference relationship of the
>>             layer in question and its parent layer.  One can
>>             recursively follow the graph to determine what used to be
>>             called dependency_id, quality_id, view_id, and whatnot.
>>              The six bit pointer field can (or: is to be when
>>             possible) organized by the encoder such that it is
>>             prudent for a middle box to throw away NAL units
>>             (belonging to layers) with higher values of the six bit
>>             field first, before throwing away NAL units with lower
>>             values.  Relying on this feature, 3+6 bits == 9 bits
>>             should be fine for the header extension.
>>
>>             That said, the ordering by the encoder is just a
>>             recommendation, and there may well be cases where
>>             different pruning strategies may be advisable.  For
>>             example, a layering structure could be constructed that
>>             expands into two branches, one using 2D scalable tools
>>             only, the other including view_id for multi view coding.
>>              By looking at the six bit field alone, a middle box will
>>             not be able to meaningfully remove NAL units belonging to
>>             one of the branches completely while pruning the other
>>             branch.  In order to meaningfully deal with that
>>             scenario, there would be two options: one to represent
>>             the dimension_id[][] (and associated control info) in the
>>             header extension, or require the middle box to have
>>             access to the VPS and be able to interpret its content.
>>              The further could take considerably more than 16 bits
>>             and we would be talking about a variable length data
>>             structure.  The latter requires the middle box to have
>>             state and a mechanism to intercept the VPS (through
>>             signaling—as the encrypted in-band VPS would not be
>>             useful under the assumption that the middle box does not
>>             have the key to the media—which is the motivation of the
>>             draft in the first place).  I personally don't mind at
>>             all the second mechanism, as I'm a big fan of out-of-band
>>             parameter set transmission and any middle box must be in
>>             the signaling path anyway to meaningfully manipulate RTP.
>>              I do not like the first option due to its variable, and
>>             possibly substantial, overhead.
>>
>>             Stephan
>>
>>             *From: *Justin Uberti <juberti@google.com
>>             <mailto:juberti@google.com>>
>>             *Date: *Friday, 19 July, 2013 06:32
>>             *To: *Bernard Aboba <bernard_aboba@hotmail.com
>>             <mailto:bernard_aboba@hotmail.com>>
>>             *Cc: *"avtext@ietf.org <mailto:avtext@ietf.org>"
>>             <avtext@ietf.org <mailto:avtext@ietf.org>>,
>>             "rtcweb@ietf.org <mailto:rtcweb@ietf.org>"
>>             <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>>             *Subject: *Re: [rtcweb] Fwd: New Version Notification for
>>             draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>             Agree those are the right codecs to design for. Since in
>>             each case there are fairly low limits on the number of
>>             supported layers (i.e. 3 spatial layers for SVC), I think
>>             it should be possible to pack the temporal, spatial,
>>             quality layer ids into 16 bits.
>>
>>             On Fri, Jul 19, 2013 at 1:56 AM, Bernard Aboba
>>             <bernard_aboba@hotmail.com
>>             <mailto:bernard_aboba@hotmail.com>> wrote:
>>
>>             If we can support VP8/9 as well as H.264/5 SVC
>>
>>             that would be a start. It seems doable to me.
>>
>>
>>             On Jul 18, 2013, at 8:34 PM, "Adam Fineberg"
>>             <fineberg@vline.me <mailto:fineberg@vline.me>> wrote:
>>
>>                 Bernard,
>>
>>                 Are there other codecs you are thinking should be
>>                 supported?  If it's generalized I would think we want
>>                 to be able to cover all known scalable codecs. I'll
>>                 look into the H264/SVC fields to see how to encode
>>                 them in a generalized header.
>>
>>                 Regards,
>>                 Adam
>>
>>                 On 7/18/13 7:40 PM, Bernard Aboba wrote:
>>
>>                     I think it may be possible to generalize this. 
>>                     For example, for H.264/SVC which can support
>>                     temporal, spatial and quality scalability, you
>>                     would need the quality_id and dependency_id in
>>                     addition to the temporal_id (what you call the
>>                     temporal layer index).
>>
>>                     ------------------------------------------------------------------------
>>
>>                     Date: Thu, 18 Jul 2013 08:45:38 -0700
>>                     From: fineberg@vline.me <mailto:fineberg@vline.me>
>>                     To: bernard_aboba@hotmail.com
>>                     <mailto:bernard_aboba@hotmail.com>
>>                     CC: rtcweb@ietf.org <mailto:rtcweb@ietf.org>;
>>                     avtext@ietf.org <mailto:avtext@ietf.org>
>>                     Subject: Re: [rtcweb] Fwd: New Version
>>                     Notification for
>>                     draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>                     Bernard,
>>
>>                     Good question.  I'm not familiar enough with the
>>                     parameter requirements of all other scalable
>>                     codecs to be able to generalize. If you'd like to
>>                     help specify them, I'd be fine revising the draft
>>                     to generalize.
>>
>>                     Regards,
>>                     Adam
>>
>>                     On 7/17/13 8:26 PM, Bernard Aboba wrote:
>>
>>                         Since the need is not codec specific (e.g. it
>>                         arises with any codec supporting temporal,
>>                         spatial and quality scalability), why
>>                          a VP8-specific RTP extension?
>>
>>                         ------------------------------------------------------------------------
>>
>>                         Date: Wed, 17 Jul 2013 17:09:46 -0700
>>                         From: fineberg@vline.me
>>                         <mailto:fineberg@vline.me>
>>                         To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>                         Subject: [rtcweb] Fwd: New Version
>>                         Notification for
>>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>                         Hi,
>>
>>                         I'm working on WebRTC services and have found
>>                         that while developing services that forward
>>                         VP8 video streams if we want to take
>>                         advantage of the VP8 temporal scaling we must
>>                         get the temporal layer information from the
>>                         RTP header which requires us to decrypt the
>>                         SRTP packets. This is undesirable both
>>                         because the middle-box needs to have access
>>                         to the keys as well as the because of the
>>                         added overhead of the decrypt/encrypt cycle.
>>                         This draft proposes an RTP header extension
>>                         that will allow us to use the VP8 temporal
>>                         layer information included in the header
>>                         extension and therefore do forwarding without
>>                         SRTP decryption. Comments welcome.
>>
>>                         Regards,
>>                         Adam Fineberg
>>
>>                         fineberg at vline.com
>>                         <mailto:fineberg%20at%20vline.com>
>>
>>                         -------- Original Message --------
>>
>>                         *Subject:*
>>
>>                         	
>>
>>                         New Version Notification for
>>                         draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>                         *Date:*
>>
>>                         	
>>
>>                         Tue, 09 Jul 2013 10:02:05 -0700
>>
>>                         *From:*
>>
>>                         	
>>
>>                         internet-drafts at ietf.org
>>                         <mailto:internet-drafts%20at%20ietf.org>
>>
>>                         *To:*
>>
>>                         	
>>
>>                         Adam Fineberg <fineberg at vline.com>
>>                         <mailto:fineberg%20at%20vline.com>
>>
>>
>>
>>
>>                         A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>                         has been successfully submitted by Adam Fineberg and posted to the
>>
>>                         IETF repository.
>>
>>                           
>>
>>                         Filename:         draft-fineberg-avtext-temporal-layer-ext
>>
>>                         Revision:         00
>>
>>                         Title:            A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information
>>
>>                         Creation date:    2013-07-08
>>
>>                         Group:            Individual Submission
>>
>>                         Number of pages: 6
>>
>>                         URL:http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt
>>
>>                         Status:http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext
>>
>>                         Htmlized:http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00
>>
>>                           
>>
>>                           
>>
>>                         Abstract:
>>
>>                             This document defines a mechanism by which packets of Real-Time
>>
>>                             Tranport Protocol (RTP) video streams encoded with the VP8 codec can
>>
>>                             indicate, in an RTP header extension, the temporal layer information
>>
>>                             about the frame encoded in the RTP packet.  This information can be
>>
>>                             used in a middlebox performing bandwidth management of streams
>>
>>                             without requiring it to decrypt the streams.
>>
>>
>>                         _______________________________________________
>>                         rtcweb mailing list rtcweb@ietf.org
>>                         <mailto:rtcweb@ietf.org>
>>                         https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>                     -- 
>>
>>                     Regards,
>>
>>                     Adam
>>
>>
>>             _______________________________________________
>>             rtcweb mailing list
>>             rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>             https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>>             _______________________________________________
>>
>>             avtext mailing list
>>
>>             avtext@ietf.org  <mailto:avtext@ietf.org>https://www.ietf.org/mailman/listinfo/avtext
>>
>>


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    <div class="moz-cite-prefix">Stephan,<br>
      <br>
      This seems like a reasonable approach.  I like your point about
      designing the RTP stack to generically handle scalability
      features.  My concern is complexity and overhead for what should
      be a very lightweight process however generalizing seems to be the
      consensus so that should be the direction forward.<br>
      <br>
      Regards,<br>
      Adam<br>
      <br>
      On 7/25/13 12:44 AM, Stephan Wenger wrote:<br>
    </div>
    <blockquote cite="mid:CE169EA7.9FC3C%25stewe@stewe.org" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <div>Hi,</div>
      <div>I thought about this a bit more.  I think that one sensible
        document structure may be as follows:</div>
      <div><br>
      </div>
      <div>(1) generic structure of scalability/pruning information to
        an RTP header extension (to be worked on in avtext).</div>
      <div>(2) mapping of codec-specific features to the generic
        structure.  As a location, I would suggest that (for new codecs)
        the RTP payload specs are the right document.  For codecs with
        existing RTP payload specs, a short doc that updates the payload
        spec would be the right place, or a revision of the payload spec
        itself if that needs to be done anyway.  This would bundle all
        the codec-specifics to one document.</div>
      <div><br>
      </div>
      <div>This document structure, and the philosophy behind it, would
        allow the design of a generic RTP stack that could take
        advantage of scalability features independently of the scalable
        codec actually in use.  I believe that such an advantage
        outweighs the possible advantage of quick standardization of a
        more specific approach as proposed in Adam's draft.  That said,
        we need to get the design of the generic structure right such
        that it is indeed useful, ideally for all present and hopefully
        for a large subset of future codecs.</div>
      <div><br>
      </div>
      <div>As for the specs involved:</div>
      <div>-H.264 SVC, H.265, and VP8 have payload format RFCs or
        reasonably stable drafts.</div>
      <div>-No one uses H.263 and MPEG-2 scalability, so we can ignore
        those two.</div>
      <div>-As Adam, I'm also not aware of a VP9 payload spec draft.
         However, my understanding is that VP9 will include a number of
        scalability features enabled by multiple reference pictures and
        reference picture resampling.  The latter is a somewhat
        different approach compared to the traditional scalability
        implementations.  I think that google planned to freeze the VP9
        bitstream a few weeks go, but I don't know whether that has
        happened.  Without a stable spec and/or input from google/webm
        folks it will indeed be hard to address VP9's specific needs, if
        any.</div>
      <div>-There are a bunch of audio codecs that claim to be scalable.
         We should have a scope discussion on whether those need to be
        included here.</div>
      <div><br>
      </div>
      <div>Stephan</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
      </div>
      <span id="OLK_SRC_BODY_SECTION">
        <div style="font-family:Calibri; font-size:11pt;
          text-align:left; color:black; BORDER-BOTTOM: medium none;
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          BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
          <span style="font-weight:bold">From: </span>Adam Fineberg
          &lt;<a moz-do-not-send="true" href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
          <span style="font-weight:bold">Date: </span>Thursday, 25
          July, 2013 00:54 <br>
          <span style="font-weight:bold">To: </span>"Wang, Ye-Kui" &lt;<a
            moz-do-not-send="true" href="mailto:yekuiw@qti.qualcomm.com">yekuiw@qti.qualcomm.com</a>&gt;<br>
          <span style="font-weight:bold">Cc: </span>Bernard Aboba &lt;<a
            moz-do-not-send="true"
            href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
          "<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
          &lt;<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;,
          Justin Uberti &lt;<a moz-do-not-send="true"
            href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
          <span style="font-weight:bold">Subject: </span>Re: [avtext]
          [rtcweb] Fwd: New Version Notification for
          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
        </div>
        <div><br>
        </div>
        <div>
          <div bgcolor="#FFFFFF" text="#000000">YK,<br>
            <br>
            I would appreciate your collaboration.  Which codecs are you
            referring to when you say "all existing standard scalable
            video codecs"?
            <br>
            <br>
            Regards,<br>
            Adam<br>
            <br>
            <div class="moz-cite-prefix">On 7/24/13 3:32 PM, Wang,
              Ye-Kui wrote:<br>
            </div>
            <blockquote
cite="mid:8BA7D4CEACFFE04BA2D902BF11719A83384A0A5B@nasanexd02f.na.qualcomm.com"
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                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); ">If the group is to specify something
                    generic, naturally it should be generic enough to
                    cover at least all existing standard scalable video
                    codecs if possible. And I personally think that is
                    possible and not difficult at all. Thus, why limit
                    to only a few scalable video codecs?<o:p></o:p></span></p>
                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); "><o:p> </o:p></span></p>
                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); ">I could provide some help here too if
                    needed.<o:p></o:p></span></p>
                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); "><o:p> </o:p></span></p>
                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); ">BR, YK<o:p></o:p></span></p>
                <p class="MsoNormal"><span style="font-size: 11pt;
                    font-family: Calibri, sans-serif; color: rgb(31, 73,
                    125); "><o:p> </o:p></span></p>
                <div style="border:none;border-left:solid blue
                  1.5pt;padding:0in 0in 0in 4.0pt">
                  <div>
                    <div style="border:none;border-top:solid #B5C4DF
                      1.0pt;padding:3.0pt 0in 0in 0in">
                      <p class="MsoNormal"><b><span style="font-size:
                            10pt; font-family: Tahoma, sans-serif;
                            color: windowtext; " lang="EN-US">From:</span></b><span
                          style="font-size: 10pt; font-family: Tahoma,
                          sans-serif; color: windowtext; " lang="EN-US"><a
                            moz-do-not-send="true"
                            class="moz-txt-link-abbreviated"
                            href="mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a>
                          [<a moz-do-not-send="true"
                            class="moz-txt-link-freetext"
                            href="mailto:avtext-bounces@ietf.org">mailto:avtext-bounces@ietf.org</a>]
                          <b>On Behalf Of </b>Adam Fineberg<br>
                          <b>Sent:</b> Wednesday, July 24, 2013 10:08 AM<br>
                          <b>To:</b> Bernard Aboba<br>
                          <b>Cc:</b> <a moz-do-not-send="true"
                            class="moz-txt-link-abbreviated"
                            href="mailto:avtext@ietf.org">avtext@ietf.org</a>;
                          <a moz-do-not-send="true"
                            class="moz-txt-link-abbreviated"
                            href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>;
                          Justin Uberti<br>
                          <b>Subject:</b> Re: [avtext] [rtcweb] Fwd: New
                          Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
                    </div>
                  </div>
                  <p class="MsoNormal"><o:p> </o:p></p>
                  <p class="MsoNormal" style="margin-bottom:12.0pt">Bernard,<br>
                    <br>
                    I apologize if I come across as being difficult here
                    but I stil am not seeing the benefits.  Since the
                    fields are not the same for the codecs, we will be
                    multiplexing the bits and that seems to me to add
                    complexity rather than add clarity.  Also, I can't
                    find an IETF VP9 document for the payload format to
                    reference.  If the group thinks generalization is
                    the right approach I would appreciate some
                    collaboration on getting the right bit definitions
                    for the other codecs.<br>
                    <br>
                    Regards,<br>
                    Adam<o:p></o:p></p>
                  <div>
                    <p class="MsoNormal">On 7/23/13 12:07 PM, Bernard
                      Aboba wrote:<o:p></o:p></p>
                  </div>
                  <blockquote
                    style="margin-top:5.0pt;margin-bottom:5.0pt">
                    <div>
                      <p class="MsoNormal" style="margin-bottom:12.0pt">I
                        do not think it is necessary to "support all
                        forms of scalability for all codecs".   In fact,
                        I would make that an explicit "non-goal".  All
                        that was suggested is to try to create a single
                        extension that supports a few common cases.   If
                        it is possible to handle VP8, VP9 and H.264/SVC
                        in a single extension that would be sufficient. 
                        So why not limit it to that?
                        <o:p></o:p></p>
                      <div>
                        <div class="MsoNormal" style="text-align:center"
                          align="center">
                          <hr id="stopSpelling" align="center" size="2"
                            width="100%">
                        </div>
                        <p class="MsoNormal"
                          style="margin-bottom:12.0pt">Date: Tue, 23 Jul
                          2013 08:53:45 -0700<br>
                          From: <a moz-do-not-send="true"
                            href="mailto:fineberg@vline.me">fineberg@vline.me</a><br>
                          To: <a moz-do-not-send="true"
                            href="mailto:stewe@stewe.org">stewe@stewe.org</a><br>
                          CC: <a moz-do-not-send="true"
                            href="mailto:juberti@google.com">juberti@google.com</a>;
                          <a moz-do-not-send="true"
                            href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>;
                          <a moz-do-not-send="true"
                            href="mailto:avtext@ietf.org">avtext@ietf.org</a>;
                          <a moz-do-not-send="true"
                            href="mailto:rtcweb@ietf.org">
                            rtcweb@ietf.org</a><br>
                          Subject: Re: [avtext] [rtcweb] Fwd: New
                          Version Notification for
                          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                          <br>
                          I've been thinking about this and given the
                          ease at which RFC5285 allows for the
                          specification of a header extension and the
                          complexity introduced by trying to generalize
                          the header extension to support all forms of
                          scalability for all codecs that the
                          generalization might not be the best
                          approach.  I'm not sure what we really gain by
                          trying to capture all this in a single header
                          extension rather than one per that can
                          succinctly explain the fields without the
                          complexity of multiplexing the bits.<br>
                          <br>
                          Thoughts?<br>
                          <br>
                          Regards,<br>
                          Adam<o:p></o:p></p>
                        <div>
                          <p class="MsoNormal">On 7/19/13 3:44 PM,
                            Stephan Wenger wrote:<o:p></o:p></p>
                        </div>
                        <blockquote
                          style="margin-top:5.0pt;margin-bottom:5.0pt">
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; color: blue; ">Hi,</span><span
                                style="font-size: 10.5pt; font-family:
                                Calibri, sans-serif; "><o:p></o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div style="border:none;border-top:solid
                            #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
                            <p class="MsoNormal"><b><span
                                  style="font-size: 11pt; font-family:
                                  Calibri, sans-serif; ">From:
                                </span></b><span style="font-size: 11pt;
                                font-family: Calibri, sans-serif; ">Adam
                                Fineberg &lt;<a moz-do-not-send="true"
                                  href="mailto:fineberg@vline.me">fineberg@vline.me</a>&gt;<br>
                                <b>Date: </b>Friday, 19 July, 2013
                                15:12 <br>
                                <b>To: </b>Stephan Wenger &lt;<a
                                  moz-do-not-send="true"
                                  href="mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
                                <b>Cc: </b>Justin Uberti &lt;<a
                                  moz-do-not-send="true"
                                  href="mailto:juberti@google.com">juberti@google.com</a>&gt;,
                                Bernard Aboba &lt;<a
                                  moz-do-not-send="true"
                                  href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;,
                                "<a moz-do-not-send="true"
                                  href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                                &lt;<a moz-do-not-send="true"
                                  href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                                "<a moz-do-not-send="true"
                                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                                &lt;<a moz-do-not-send="true"
                                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                                <b>Subject: </b>Re: [avtext] [rtcweb]
                                Fwd: New Version Notification for
                                draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div>
                            <div>
                              <p class="MsoNormal"><span
                                  style="font-size: 10.5pt; font-family:
                                  Calibri, sans-serif; ">Stephan,<br>
                                  <br>
                                  Thanks for the info and the
                                  reference.  I'm not sure I follow as
                                  I'm not at all familiar with H.265. 
                                  I'll review the reference and see what
                                  I can figure. <o:p></o:p></span></p>
                            </div>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; color: blue; ">StW: Good
                                luck :-)  </span><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p></o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div>
                            <div>
                              <p class="MsoNormal"><span
                                  style="font-size: 10.5pt; font-family:
                                  Calibri, sans-serif; ">It seems though
                                  to me that you are suggesting that
                                  except in the simple case, that the
                                  data for H.265 would not be well
                                  suited to a header extension, am I
                                  understanding you correctly?  There is
                                  no reason the middlebox couldn't get
                                  out of band signaling of the VPS as
                                  you mention but that would not be
                                  within the scope of this header
                                  extension.<o:p></o:p></span></p>
                            </div>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span
                                style="font-family: Calibri, sans-serif;
                                color: blue; ">StW: well, if you would
                                copy the layer_id into your header
                                extension (just as you need to do for
                                the simple case), a really smart middle
                                box could use this information just as a
                                decoder uses it, assuming that it
                                intercepted the VPS in the first place.
                                 Insofar, I wouldn't rule out the second
                                option on technical grounds.  Whether
                                any of the actual products would bother
                                to do that, ever, is another question.
                                 I think the case ought to be
                                documented, though.  I can help drafting
                                text.</span><o:p></o:p></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span
                                style="font-family: Calibri, sans-serif;
                                color: blue; ">While we are at it: doing
                                this right could mean that you need
                                multiple specs.  First, a generic header
                                extension mechanism dedicated to side
                                information required for pruning of RTP
                                packet streams—ideally not only for
                                scalable video, although that is the
                                main customer today.  And second, for
                                each "payload" (at present we are
                                talking about H.264/SVC, H.265v1 (HEVC),
                                H.265v2 (including scalable and 3D
                                extensions, which are not yet
                                finalized), VP8, and VP9 (I know little
                                about the latter), plus Daala, and
                                whatnot) a mapping of the bits available
                                in the generic header extension to the
                                bits in the payload itself (NAL header
                                and VPS in case of H.265, NALU header in
                                SVC, and the fields you mention for
                                VP8).</span><o:p></o:p></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; "><o:p> </o:p></span></p>
                          </div>
                          <div>
                            <p class="MsoNormal"><span style="font-size:
                                10.5pt; font-family: Calibri,
                                sans-serif; color: blue; ">Stephan</span><span
                                style="font-size: 10.5pt; font-family:
                                Calibri, sans-serif; "><o:p></o:p></span></p>
                          </div>
                          <div>
                            <div>
                              <p class="MsoNormal"
                                style="margin-bottom:12.0pt"><span
                                  style="font-size: 10.5pt; font-family:
                                  Calibri, sans-serif; "><br>
                                  <br>
                                  Any insights are appreciated.<br>
                                  <br>
                                  Regards,<br>
                                  Adam<o:p></o:p></span></p>
                              <div>
                                <p class="MsoNormal"><span
                                    style="font-size: 10.5pt;
                                    font-family: Calibri, sans-serif; ">On
                                    7/19/13 8:33 AM, Stephan Wenger
                                    wrote:<o:p></o:p></span></p>
                              </div>
                              <blockquote
                                style="margin-top:5.0pt;margin-bottom:5.0pt">
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      ">Hi,<o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      ">I also believe that 16 bits
                                      should be enough.  For H.264 and
                                      VP8 that has already been
                                      demonstrated.  For H.265, some
                                      initial thoughts below.  Apologies
                                      for the word-count.<o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      ">The scalable version of H265
                                      (called SHVC) is currently under
                                      development.  The current working
                                      draft can be found here: <a
                                        moz-do-not-send="true"
href="http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip"
                                        target="_blank">http://phenix.int-evry.fr/jct/doc_end_user/documents/13_Incheon/wg11/JCTVC-M1008-v3.zip</a>.
                                       Therein, the options for defining
                                      layering structures are
                                      considerably more complex.  To
                                      start, we have 3 bits for the
                                      temporal ID in the NAL unit header
                                      of the H.265 version 1 (HEVC) base
                                      specification (temporal
                                      scalability is already nicely
                                      supported in version 1).  Just
                                      like in SVC.  In the scalable
                                      extension, the NAL unit header
                                      contains a six bit field that
                                      points into a data structure known
                                      as "Video Parameter Set" (VPS).
                                       Inside the VPS, those six bits
                                      are mapped to to a position in a
                                      directed graph (specified through
                                      "dimension_id[][]"), which tells
                                      you about the reference
                                      relationship of the layer in
                                      question and its parent layer.
                                       One can recursively follow the
                                      graph to determine what used to be
                                      called dependency_id, quality_id,
                                      view_id, and whatnot.  The six bit
                                      pointer field can (or: is to be
                                      when possible) organized by the
                                      encoder such that it is prudent
                                      for a middle box to throw away NAL
                                      units (belonging to layers) with
                                      higher values of the six bit field
                                      first, before throwing away NAL
                                      units with lower values.  Relying
                                      on this feature, 3+6 bits == 9
                                      bits should be fine for the header
                                      extension.<o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      ">That said, the ordering by the
                                      encoder is just a recommendation,
                                      and there may well be cases where
                                      different pruning strategies may
                                      be advisable.  For example, a
                                      layering structure could be
                                      constructed that expands into two
                                      branches, one using 2D scalable
                                      tools only, the other including
                                      view_id for multi view coding.  By
                                      looking at the six bit field
                                      alone, a middle box will not be
                                      able to meaningfully remove NAL
                                      units belonging to one of the
                                      branches completely while pruning
                                      the other branch.  In order to
                                      meaningfully deal with that
                                      scenario, there would be two
                                      options: one to represent the
                                      dimension_id[][] (and associated
                                      control info) in the header
                                      extension, or require the middle
                                      box to have access to the VPS and
                                      be able to interpret its content.
                                       The further could take
                                      considerably more than 16 bits and
                                      we would be talking about a
                                      variable length data structure.
                                       The latter requires the middle
                                      box to have state and a mechanism
                                      to intercept the VPS (through
                                      signaling—as the encrypted in-band
                                      VPS would not be useful under the
                                      assumption that the middle box
                                      does not have the key to the
                                      media—which is the motivation of
                                      the draft in the first place).  I
                                      personally don't mind at all the
                                      second mechanism, as I'm a big fan
                                      of out-of-band parameter set
                                      transmission and any middle box
                                      must be in the signaling path
                                      anyway to meaningfully manipulate
                                      RTP.  I do not like the first
                                      option due to its variable, and
                                      possibly substantial, overhead.<o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      ">Stephan   <o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div style="border:none;border-top:solid
                                  #B5C4DF 1.0pt;padding:3.0pt 0in 0in
                                  0in">
                                  <p class="MsoNormal"><b><span
                                        style="font-size: 11pt;
                                        font-family: Calibri,
                                        sans-serif; ">From:
                                      </span></b><span style="font-size:
                                      11pt; font-family: Calibri,
                                      sans-serif; ">Justin Uberti &lt;<a
                                        moz-do-not-send="true"
                                        href="mailto:juberti@google.com">juberti@google.com</a>&gt;<br>
                                      <b>Date: </b>Friday, 19 July,
                                      2013 06:32 <br>
                                      <b>To: </b>Bernard Aboba &lt;<a
                                        moz-do-not-send="true"
                                        href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;<br>
                                      <b>Cc: </b>"<a
                                        moz-do-not-send="true"
                                        href="mailto:avtext@ietf.org">avtext@ietf.org</a>"
                                      &lt;<a moz-do-not-send="true"
                                        href="mailto:avtext@ietf.org">avtext@ietf.org</a>&gt;,
                                      "<a moz-do-not-send="true"
                                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>"
                                      &lt;<a moz-do-not-send="true"
                                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
                                      <b>Subject: </b>Re: [rtcweb] Fwd:
                                      New Version Notification for
                                      draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></span></p>
                                </div>
                                <div>
                                  <p class="MsoNormal"><span
                                      style="font-size: 10.5pt;
                                      font-family: Calibri, sans-serif;
                                      "><o:p> </o:p></span></p>
                                </div>
                                <div>
                                  <div>
                                    <div>
                                      <p class="MsoNormal"><span
                                          style="font-size: 10.5pt;
                                          font-family: Calibri,
                                          sans-serif; ">Agree those are
                                          the right codecs to design
                                          for. Since in each case there
                                          are fairly low limits on the
                                          number of supported layers
                                          (i.e. 3 spatial layers for
                                          SVC), I think it should be
                                          possible to pack the temporal,
                                          spatial, quality layer ids
                                          into 16 bits.
                                          <o:p></o:p></span></p>
                                      <div>
                                        <p class="MsoNormal"
                                          style="margin-bottom:12.0pt"><span
                                            style="font-size: 10.5pt;
                                            font-family: Calibri,
                                            sans-serif; "><o:p> </o:p></span></p>
                                        <div>
                                          <p class="MsoNormal"><span
                                              style="font-size: 10.5pt;
                                              font-family: Calibri,
                                              sans-serif; ">On Fri, Jul
                                              19, 2013 at 1:56 AM,
                                              Bernard Aboba &lt;<a
                                                moz-do-not-send="true"
                                                href="mailto:bernard_aboba@hotmail.com"
                                                target="_blank">bernard_aboba@hotmail.com</a>&gt;
                                              wrote:<o:p></o:p></span></p>
                                          <div>
                                            <div>
                                              <p class="MsoNormal"><span
                                                  style="font-size:
                                                  10.5pt; font-family:
                                                  Calibri, sans-serif; ">If
                                                  we can support VP8/9
                                                  as well as H.264/5 SVC<o:p></o:p></span></p>
                                            </div>
                                            <div>
                                              <p class="MsoNormal"><span
                                                  style="font-size:
                                                  10.5pt; font-family:
                                                  Calibri, sans-serif; ">that
                                                  would be a start. It
                                                  seems doable to me.<o:p></o:p></span></p>
                                            </div>
                                            <div>
                                              <div>
                                                <div>
                                                  <p class="MsoNormal"
                                                    style="margin-bottom:12.0pt"><span
                                                      style="font-size:
                                                      10.5pt;
                                                      font-family:
                                                      Calibri,
                                                      sans-serif; "><br>
                                                      On Jul 18, 2013,
                                                      at 8:34 PM, "Adam
                                                      Fineberg" &lt;<a
                                                        moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a>&gt;
                                                      wrote:<o:p></o:p></span></p>
                                                </div>
                                                <blockquote
                                                  style="margin-top:5.0pt;margin-bottom:5.0pt">
                                                  <div>
                                                    <div>
                                                      <p
                                                        class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">Bernard,<br>
                                                          <br>
                                                          Are there
                                                          other codecs
                                                          you are
                                                          thinking
                                                          should be
                                                          supported?  If
                                                          it's
                                                          generalized I
                                                          would think we
                                                          want to be
                                                          able to cover
                                                          all known
                                                          scalable
                                                          codecs. I'll
                                                          look into the
                                                          H264/SVC
                                                          fields to see
                                                          how to encode
                                                          them in a
                                                          generalized
                                                          header.<br>
                                                          <br>
                                                          Regards,<br>
                                                          Adam<br>
                                                          <br>
                                                          On 7/18/13
                                                          7:40 PM,
                                                          Bernard Aboba
                                                          wrote:<o:p></o:p></span></p>
                                                    </div>
                                                    <blockquote
                                                      style="margin-top:5.0pt;margin-bottom:5.0pt">
                                                      <div>
                                                        <p
                                                          class="MsoNormal"
style="margin-bottom:12.0pt"><span style="font-size: 10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">I
                                                          think it may
                                                          be possible to
                                                          generalize
                                                          this.  For
                                                          example, for
                                                          H.264/SVC
                                                          which can
                                                          support
                                                          temporal,
                                                          spatial and
                                                          quality
                                                          scalability,
                                                          you would need
                                                          the quality_id
                                                          and
                                                          dependency_id
                                                          in addition to
                                                          the
                                                          temporal_id
                                                          (what you call
                                                          the temporal
                                                          layer index).
                                                            
                                                          <o:p></o:p></span></p>
                                                        <div>
                                                          <div
                                                          class="MsoNormal"
style="text-align:center" align="center"><span style="font-size: 10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">
                                                          <hr
                                                          align="center"
                                                          size="2"
                                                          width="100%">
                                                          </span></div>
                                                          <p
                                                          class="MsoNormal"
style="margin-bottom:12.0pt"><span style="font-size: 10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">Date:
                                                          Thu, 18 Jul
                                                          2013 08:45:38
                                                          -0700<br>
                                                          From: <a
                                                          moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                          To: <a
                                                          moz-do-not-send="true"
href="mailto:bernard_aboba@hotmail.com" target="_blank">
bernard_aboba@hotmail.com</a><br>
                                                          CC: <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>;
                                                          <a
                                                          moz-do-not-send="true"
href="mailto:avtext@ietf.org" target="_blank">avtext@ietf.org</a><br>
                                                          Subject: Re:
                                                          [rtcweb] Fwd:
                                                          New Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                          <br>
                                                          Bernard,<br>
                                                          <br>
                                                          Good
                                                          question.  I'm
                                                          not familiar
                                                          enough with
                                                          the parameter
                                                          requirements
                                                          of all other
                                                          scalable
                                                          codecs to be
                                                          able to
                                                          generalize. 
                                                          If you'd like
                                                          to help
                                                          specify them,
                                                          I'd be fine
                                                          revising the
                                                          draft to
                                                          generalize.<br>
                                                          <br>
                                                          Regards,<br>
                                                          Adam<o:p></o:p></span></p>
                                                          <div>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">On
                                                          7/17/13 8:26
                                                          PM, Bernard
                                                          Aboba wrote:<o:p></o:p></span></p>
                                                          </div>
                                                          <blockquote
                                                          style="margin-top:5.0pt;margin-bottom:5.0pt">
                                                          <div>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">Since
                                                          the need is
                                                          not codec
                                                          specific (e.g.
                                                          it arises with
                                                          any codec
                                                          supporting
                                                          temporal,
                                                          spatial and
                                                          quality
                                                          scalability),
                                                          why
                                                          <br>
                                                           a
                                                          VP8-specific
                                                          RTP extension?
                                                          <br>
                                                           <o:p></o:p></span></p>
                                                          <div>
                                                          <div
                                                          class="MsoNormal"
style="text-align:center" align="center"><span style="font-size: 10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">
                                                          <hr
                                                          align="center"
                                                          size="2"
                                                          width="100%">
                                                          </span></div>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; ">Date:
                                                          Wed, 17 Jul
                                                          2013 17:09:46
                                                          -0700<br>
                                                          From: <a
                                                          moz-do-not-send="true"
href="mailto:fineberg@vline.me" target="_blank">fineberg@vline.me</a><br>
                                                          To: <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                          Subject:
                                                          [rtcweb] Fwd:
                                                          New Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<br>
                                                          <br>
                                                          Hi,<br>
                                                          <br>
                                                          I'm working on
                                                          WebRTC
                                                          services and
                                                          have found
                                                          that while
                                                          developing
                                                          services that
                                                          forward VP8
                                                          video streams
                                                          if we want to
                                                          take advantage
                                                          of the VP8
                                                          temporal
                                                          scaling we
                                                          must get the
                                                          temporal layer
                                                          information
                                                          from the RTP
                                                          header which
                                                          requires us to
                                                          decrypt the
                                                          SRTP packets.
                                                          This is
                                                          undesirable
                                                          both because
                                                          the middle-box
                                                          needs to have
                                                          access to the
                                                          keys as well
                                                          as the because
                                                          of the added
                                                          overhead of
                                                          the
                                                          decrypt/encrypt
                                                          cycle. This
                                                          draft proposes
                                                          an RTP header
                                                          extension that
                                                          will allow us
                                                          to use the VP8
                                                          temporal layer
                                                          information
                                                          included in
                                                          the header
                                                          extension and
                                                          therefore do
                                                          forwarding
                                                          without SRTP
                                                          decryption.
                                                          Comments
                                                          welcome.<br>
                                                          <br>
                                                          Regards,<br>
                                                          Adam Fineberg<o:p></o:p></span></p>
                                                          <div>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; "><a
moz-do-not-send="true" href="mailto:fineberg%20at%20vline.com"
                                                          target="_blank">fineberg
                                                          at vline.com</a><br>
                                                          <br>
                                                          --------
                                                          Original
                                                          Message
                                                          -------- <o:p></o:p></span></p>
                                                          <table
                                                          class="MsoNormalTable"
                                                          border="0"
                                                          cellpadding="0"
cellspacing="0">
                                                          <tbody>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>Subject:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">New
                                                          Version
                                                          Notification
                                                          for
                                                          draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>Date:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">Tue,
                                                          09 Jul 2013
                                                          10:02:05 -0700<o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>From:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal"><a
moz-do-not-send="true" href="mailto:internet-drafts%20at%20ietf.org"
                                                          target="_blank">internet-drafts
                                                          at ietf.org</a><o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          <tr>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in"
                                                          nowrap="nowrap"
                                                          valign="top">
                                                          <p
                                                          class="MsoNormal"
style="text-align:right" align="right"><b>To:<o:p></o:p></b></p>
                                                          </td>
                                                          <td
                                                          style="padding:0in
                                                          0in 0in 0in">
                                                          <p
                                                          class="MsoNormal">Adam
                                                          Fineberg <a
                                                          moz-do-not-send="true"
href="mailto:fineberg%20at%20vline.com" target="_blank">&lt;fineberg at
                                                          vline.com&gt;</a><o:p></o:p></p>
                                                          </td>
                                                          </tr>
                                                          </tbody>
                                                          </table>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; "><br>
                                                          <br>
                                                          <br>
                                                          <o:p></o:p></span></p>
                                                          <pre>A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt<o:p></o:p></pre>
                                                          <pre>has been successfully submitted by Adam Fineberg and posted to the<o:p></o:p></pre>
                                                          <pre>IETF repository.<o:p></o:p></pre>
                                                          <pre><o:p> </o:p></pre>
                                                          <pre>Filename:         draft-fineberg-avtext-temporal-layer-ext<o:p></o:p></pre>
                                                          <pre>Revision:         00<o:p></o:p></pre>
                                                          <pre>Title:            A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temporal Layer Information<o:p></o:p></pre>
                                                          <pre>Creation date:    2013-07-08<o:p></o:p></pre>
                                                          <pre>Group:            Individual Submission<o:p></o:p></pre>
                                                          <pre>Number of pages: 6<o:p></o:p></pre>
                                                          <pre>URL:             <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt" target="_blank">http://www.ietf.org/internet-drafts/draft-fineberg-avtext-temporal-layer-ext-00.txt</a><o:p></o:p></pre>
                                                          <pre>Status:          <a moz-do-not-send="true" href="http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext" target="_blank">http://datatracker.ietf.org/doc/draft-fineberg-avtext-temporal-layer-ext</a><o:p></o:p></pre>
                                                          <pre>Htmlized:        <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00" target="_blank">http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext-00</a><o:p></o:p></pre>
                                                          <pre><o:p> </o:p></pre>
                                                          <pre><o:p> </o:p></pre>
                                                          <pre>Abstract:<o:p></o:p></pre>
                                                          <pre>   This document defines a mechanism by which packets of Real-Time<o:p></o:p></pre>
                                                          <pre>   Tranport Protocol (RTP) video streams encoded with the VP8 codec can<o:p></o:p></pre>
                                                          <pre>   indicate, in an RTP header extension, the temporal layer information<o:p></o:p></pre>
                                                          <pre>   about the frame encoded in the RTP packet.  This information can be<o:p></o:p></pre>
                                                          <pre>   used in a middlebox performing bandwidth management of streams<o:p></o:p></pre>
                                                          <pre>   without requiring it to decrypt the streams.<o:p></o:p></pre>
                                                          </div>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; "><br>
                                                          _______________________________________________
                                                          rtcweb mailing
                                                          list <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">
rtcweb@ietf.org</a> <a moz-do-not-send="true"
                                                          href="https://www.ietf.org/mailman/listinfo/rtcweb"
target="_blank">
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
                                                          </div>
                                                          </div>
                                                          </blockquote>
                                                          <p
                                                          class="MsoNormal"><span
                                                          style="font-size:
                                                          10.5pt;
                                                          font-family:
                                                          Calibri,
                                                          sans-serif; "><o:p> </o:p></span></p>
                                                          <pre>-- <o:p></o:p></pre>
                                                          <pre>Regards,<o:p></o:p></pre>
                                                          <pre>Adam<o:p></o:p></pre>
                                                        </div>
                                                      </div>
                                                    </blockquote>
                                                    <p class="MsoNormal"><span
                                                        style="font-size:
                                                        10.5pt;
                                                        font-family:
                                                        Calibri,
                                                        sans-serif; "><o:p> </o:p></span></p>
                                                  </div>
                                                </blockquote>
                                              </div>
                                            </div>
                                          </div>
                                          <p class="MsoNormal"
                                            style="margin-bottom:12.0pt"><span
                                              style="font-size: 10.5pt;
                                              font-family: Calibri,
                                              sans-serif; "><br>
_______________________________________________<br>
                                              rtcweb mailing list<br>
                                              <a moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                              <a moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
                                        </div>
                                        <p class="MsoNormal"><span
                                            style="font-size: 10.5pt;
                                            font-family: Calibri,
                                            sans-serif; "><o:p> </o:p></span></p>
                                      </div>
                                    </div>
                                  </div>
                                </div>
                                <p class="MsoNormal"><span
                                    style="font-size: 10.5pt;
                                    font-family: Calibri, sans-serif; "><br>
                                    <br>
                                    <o:p></o:p></span></p>
                                <pre>_______________________________________________<o:p></o:p></pre>
                                <pre>avtext mailing list<o:p></o:p></pre>
                                <pre><a moz-do-not-send="true" href="mailto:avtext@ietf.org">avtext@ietf.org</a><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/avtext" target="_blank">https://www.ietf.org/mailman/listinfo/avtext</a><o:p></o:p></pre>
                              </blockquote>
                              <br>
                            </div>
                          </div>
                        </blockquote>
                      </div>
                    </div>
                  </blockquote>
                </div>
              </div>
            </blockquote>
          </div>
        </div>
      </span></blockquote>
    <br>
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Date: Fri, 26 Jul 2013 09:42:56 -0400
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From: Kaiduan Xie <kaiduanx@gmail.com>
To: Oleg Moskalenko <mom040267@gmail.com>
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Subject: Re: [rtcweb] [BEHAVE] Fwd: New Version Notification for draft-uberti-behave-turn-rest-00.txt
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--001a11c3850000c0fa04e26a52cb
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Justin,

I recommend you to put a note into the draft to state the following two
points,

1) The user name and password are generated by the web server instead of
the TURN server.

2) There is no communication channel required between web server and TURN
server.

>From the title (TURN Server REST API) and the text, it is easy to
misunderstand that TURN server processes the HTTP POST request.

Thanks,

/Kaiduan

On Wed, Jul 24, 2013 at 10:06 AM, Oleg Moskalenko <mom040267@gmail.com>wrote:

> Thank you for the new link.
>
> I checked the new version of the draft and I personally see no problem in
> the text. There is no proprietary software requirements in the draft. It
> simply defines the logic how the TURN server and web server can organize
> the temporary password generation, without imposing any proprietary
> requirements and specs on the software. It is mentioning a possible
> communication channel between web server and TURN server without defining
> any specs and as it is written that channel is not required. As it is
> written, I do not think that it has to be separated into two pieces - it is
> a single solid logical functionality definition.
>
> Thanks
> Oleg
>
>
> On Wed, Jul 24, 2013 at 1:46 AM, Rajmohan Banavi <rajmohanbanavi@gmail.com
> > wrote:
>
>> This is the draft (BEHAVE WG) I am referring to -
>> http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
>>
>>
>>> This is not the case. It is not the TURN server who generates the
>>> credentials. The web server must generate the temporary password, and to be
>>> able to do that the web server must have the shared secret - the same as
>>> TURN server has. How they share the same shared secret I'd leave outside
>>> the proposed specs.
>>>
>>> OK fine.
>>
>>
>>> It is rather clear - the web server takes the shared secret and it
>>> generates the temporary password for long-term TURN credentials. The TURN
>>> server can reproduce that generation process and obtain the same temporary
>>> password - because the TURN server knows the same shared secret as the web
>>> server.
>>>
>>
>> OK fine.
>>
>> Thanks,
>> Rajmohan
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c3850000c0fa04e26a52cb
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Justin,</div><div><br></div>I recommend you to put a =
note into the draft to state the following two points,<div><br></div><div>1=
) The user name and password are generated by the web server instead of the=
 TURN server.</div>
<div><br></div><div>2) There is no communication channel required between w=
eb server and TURN server.</div><div><br></div><div>From the title (TURN Se=
rver REST API) and the text, it is easy to misunderstand that TURN server p=
rocesses the HTTP POST request.</div>
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Thanks,</di=
v><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">/Kaiduan<=
br><br><div class=3D"gmail_quote">On Wed, Jul 24, 2013 at 10:06 AM, Oleg Mo=
skalenko <span dir=3D"ltr">&lt;<a href=3D"mailto:mom040267@gmail.com" targe=
t=3D"_blank">mom040267@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>Thank you for the=
 new link. <br><br>I checked the new version of the draft and I personally =
see no problem in the text. There is no proprietary software requirements i=
n the draft. It simply defines the logic how the TURN server and web server=
 can organize the temporary password generation, without imposing any propr=
ietary requirements and specs on the software. It is mentioning a possible =
communication channel between web server and TURN server without defining a=
ny specs and as it is written that channel is not required. As it is writte=
n, I do not think that it has to be separated into two pieces - it is a sin=
gle solid logical functionality definition.<br>

<br></div>Thanks<span class=3D"HOEnZb"><font color=3D"#888888"><br></font><=
/span></div><span class=3D"HOEnZb"><font color=3D"#888888">Oleg <br></font>=
</span><div><div class=3D"h5"><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">
On Wed, Jul 24, 2013 at 1:46 AM, Rajmohan Banavi <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:rajmohanbanavi@gmail.com" target=3D"_blank">rajmohanbanavi@gm=
ail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span style=3D"font-family:=
arial,sans-serif;font-size:13px">This is the draft (BEHAVE WG) I am referri=
ng to -=A0</span><a href=3D"http://tools.ietf.org/html/draft-uberti-behave-=
turn-rest-00" style=3D"font-family:arial,sans-serif;font-size:13px" target=
=3D"_blank">http://tools.ietf.org/html/draft-uberti-behave-turn-rest-00</a>=
<br style=3D"font-family:arial,sans-serif;font-size:13px">


<div class=3D"gmail_extra" style=3D"font-family:arial,sans-serif;font-size:=
13px"><br><div class=3D"gmail_quote"><div><div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><br></div><div>This is not the case. It is not the TURN server who generat=
es the credentials. The web server must generate the temporary password, an=
d to be able to do that the web server must have the shared secret - the sa=
me as TURN server has. How they share the same shared secret I&#39;d leave =
outside the proposed specs.<br>


</div><div><br></div></div></div></div></blockquote></div></div><div>OK fin=
e.</div><div><div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex">


<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>It is rather clear - the web server takes the shared secret and it generat=
es the temporary password for long-term TURN credentials. The TURN server c=
an reproduce that generation process and obtain the same temporary password=
 - because the TURN server knows the same shared secret as the web server.<=
br>


</div><div><div></div></div></div></div></div></blockquote><div><br></div><=
/div></div><div>OK fine.</div></div></div><div class=3D"gmail_extra"><br>Th=
anks,</div><div class=3D"gmail_extra">Rajmohan</div></div>
</blockquote></div><br></div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a11c3850000c0fa04e26a52cb--

From fluffy@iii.ca  Fri Jul 26 11:10:06 2013
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From: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Question about the status of various drafts
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Yah, I get your point. I will try and find a way to better represent =
this on the slides.=20


On Jul 5, 2013, at 3:25 PM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

> I think you need to take dependencies into account.   There are =
several reasons for this:=20
>=20
>    a. Until you've gone through the dependency graph for the items, =
the full list of documents that need to be tracked isn't known.=20
>    b. Estimates of completion are dependent on the progress of =
normative dependencies (e.g. they can't be published until the =
dependencies are resolved).=20
>=20
> As an example,  draft-ietf-rtcweb-overview (which is listed as 80 =
percent complete) has normative references to the following drafts:=20
>=20
>    [I-D.ietf-mmusic-sctp-sdp] 70
>    [I-D.ietf-rtcweb-audio] 50
>    [I-D.ietf-rtcweb-data-channel] 70
>    [I-D.ietf-rtcweb-jsep] 50
>    [I-D.ietf-rtcweb-rtp-usage] 40
>    [I-D.ietf-rtcweb-security] 80
>    [I-D.ietf-rtcweb-security-arch] 80
>    [I-D.nandakumar-rtcweb-stun-uri] 80
>    [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70
>            =20
> Since draft-ietf-rtcweb-overview is dependent on a document (rtp =
usage) which only has an estimate of 40 percent done) I'd say that the =
estimate is probably closer to 40 than 80.=20
>=20
>=20
> > From: fluffy@iii.ca
> > Date: Thu, 4 Jul 2013 17:47:20 -0700
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Question about the status of various drafts
> >=20
> >=20
> > As part of putting together some chair slides for Berlin, I'm trying =
to get a very rough idea of where things are. I'd like to get folks =
input on what percent done various drafts are where Done is a published =
RFC.=20
> >=20
> > This is nearly impossible to do at any level of accuracy but I do =
want to get a vague idea. I took a very rough stab below just to have a =
starting point to get the conversation going. I'm sure my numbers are =
mostly wrong but I have tried to ask a few people and sort of take the =
central cluster of the guess. If you think I am way out, please provide =
some feedback of what you think the number actually is.
> >=20
> > I'm happy to get feedback on specific drafts or feedback of the form =
they are all too low or too high. I do encourage people to realistically =
look at the data tracker to see how long other drafts have taken for =
each stage to get a baseline instead of just going with their gut feel.=20=

> >=20
> > Thanks, Cullen
> >=20
> > PS - if you are an author and look at the number next to your draft =
and think it is way off, please please, say something.=20
> >=20
> >=20
> >=20
> > draft-ietf-rtcweb-overview	 80
> > draft-ietf-rtcweb-use-cases-and-requirements	70
> > http://dev.w3.org/2011/webrtc/editor/getusermedia.html	70
> > draft-burnett-rtcweb-constraints-registry	 80
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html	50
> > draft-nandakumar-rtcweb-stun-uri	 80
> > draft-petithuguenin-behave-turn-uris	 80
> > draft-ietf-rtcweb-security	 80
> > draft-ietf-rtcweb-security-arch	 80
> > draft-muthu-behave-consent-freshness-02 50
> > draft-ietf-rtcweb-data-channel	 70
> > draft-ietf-mmusic-sctp-sdp	 70
> > draft-jesup-rtcweb-data-protocol 50
> > draft-ietf-tsvwg-sctp-dtls-encaps	 70
> > draft-ietf-rtcweb-rtp-usage	 40
> > draft-ietf-avtcore-6222bis	 80
> > draft-ietf-avtcore-avp-codecs	 80
> > draft-ietf-avtcore-srtp-encrypted-header-ext	80
> > draft-ietf-avtext-multiple-clock-rates	 70
> > draft-lennox-rtcweb-rtp-media-type-mux 80
> > draft-ietf-avtcore-rtp-circuit-breakers	 70
> > draft-ietf-avtcore-multi-media-rtp-session 70
> > draft-lennox-avtcore-rtp-multi-stream	 70
> > draft-ietf-rtcweb-jsep	 50
> > draft-ietf-mmusic-msid	 70
> > draft-rescorla-mmusic-ice-trickle	 25
> > draft-ietf-mmusic-sdp-bundle-negotiation	50
> > draft-nandakumar-mmusic-sdp-mux-attributes	25
> > draft-dhesikan-tsvwg-rtcweb-qos	 10 or 90
> > draft-ietf-rtcweb-audio	 50
> > draft-ietf-payload-rtp-opus	 90
> > draft-ietf-payload-vp8-08	 95
> >=20
> >=20
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Fri Jul 26 11:56:45 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>, Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Question about the status of various drafts
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Hi Cullen,

I may missed something, but it seems a little strange to me when individual=
 drafts, that haven't even been adopted yet, are indicated as 70-80% comple=
te...

Some drafts have also expired.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Cullen Jennings [fluffy@iii.ca]
To: Bernard Aboba [bernard_aboba@hotmail.com]
CC: rtcweb@ietf.org [rtcweb@ietf.org]
Subject: Re: [rtcweb] Question about the status of various drafts

Yah, I get your point. I will try and find a way to better represent this o=
n the slides.


On Jul 5, 2013, at 3:25 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrote=
:

> I think you need to take dependencies into account.   There are several r=
easons for this:
>
>    a. Until you've gone through the dependency graph for the items, the f=
ull list of documents that need to be tracked isn't known.
>    b. Estimates of completion are dependent on the progress of normative =
dependencies (e.g. they can't be published until the dependencies are resol=
ved).
>
> As an example,  draft-ietf-rtcweb-overview (which is listed as 80 percent=
 complete) has normative references to the following drafts:
>
>    [I-D.ietf-mmusic-sctp-sdp] 70
>    [I-D.ietf-rtcweb-audio] 50
>    [I-D.ietf-rtcweb-data-channel] 70
>    [I-D.ietf-rtcweb-jsep] 50
>    [I-D.ietf-rtcweb-rtp-usage] 40
>    [I-D.ietf-rtcweb-security] 80
>    [I-D.ietf-rtcweb-security-arch] 80
>    [I-D.nandakumar-rtcweb-stun-uri] 80
>    [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70
>
> Since draft-ietf-rtcweb-overview is dependent on a document (rtp usage) w=
hich only has an estimate of 40 percent done) I'd say that the estimate is =
probably closer to 40 than 80.
>
>
> > From: fluffy@iii.ca
> > Date: Thu, 4 Jul 2013 17:47:20 -0700
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Question about the status of various drafts
> >
> >
> > As part of putting together some chair slides for Berlin, I'm trying to=
 get a very rough idea of where things are. I'd like to get folks input on =
what percent done various drafts are where Done is a published RFC.
> >
> > This is nearly impossible to do at any level of accuracy but I do want =
to get a vague idea. I took a very rough stab below just to have a starting=
 point to get the conversation going. I'm sure my numbers are mostly wrong =
but I have tried to ask a few people and sort of take the central cluster o=
f the guess. If you think I am way out, please provide some feedback of wha=
t you think the number actually is.
> >
> > I'm happy to get feedback on specific drafts or feedback of the form th=
ey are all too low or too high. I do encourage people to realistically look=
 at the data tracker to see how long other drafts have taken for each stage=
 to get a baseline instead of just going with their gut feel.
> >
> > Thanks, Cullen
> >
> > PS - if you are an author and look at the number next to your draft and=
 think it is way off, please please, say something.
> >
> >
> >
> > draft-ietf-rtcweb-overview   80
> > draft-ietf-rtcweb-use-cases-and-requirements        70
> > http://dev.w3.org/2011/webrtc/editor/getusermedia.html       70
> > draft-burnett-rtcweb-constraints-registry    80
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html     50
> > draft-nandakumar-rtcweb-stun-uri     80
> > draft-petithuguenin-behave-turn-uris         80
> > draft-ietf-rtcweb-security   80
> > draft-ietf-rtcweb-security-arch      80
> > draft-muthu-behave-consent-freshness-02 50
> > draft-ietf-rtcweb-data-channel       70
> > draft-ietf-mmusic-sctp-sdp   70
> > draft-jesup-rtcweb-data-protocol 50
> > draft-ietf-tsvwg-sctp-dtls-encaps    70
> > draft-ietf-rtcweb-rtp-usage  40
> > draft-ietf-avtcore-6222bis   80
> > draft-ietf-avtcore-avp-codecs        80
> > draft-ietf-avtcore-srtp-encrypted-header-ext        80
> > draft-ietf-avtext-multiple-clock-rates       70
> > draft-lennox-rtcweb-rtp-media-type-mux 80
> > draft-ietf-avtcore-rtp-circuit-breakers      70
> > draft-ietf-avtcore-multi-media-rtp-session 70
> > draft-lennox-avtcore-rtp-multi-stream        70
> > draft-ietf-rtcweb-jsep       50
> > draft-ietf-mmusic-msid       70
> > draft-rescorla-mmusic-ice-trickle    25
> > draft-ietf-mmusic-sdp-bundle-negotiation    50
> > draft-nandakumar-mmusic-sdp-mux-attributes  25
> > draft-dhesikan-tsvwg-rtcweb-qos      10 or 90
> > draft-ietf-rtcweb-audio      50
> > draft-ietf-payload-rtp-opus  90
> > draft-ietf-payload-vp8-08    95
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Cullen,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I may missed something, but it seems a little s=
trange to me when individual drafts, that haven't even been adopted yet, ar=
e indicated as 70-80% complete...</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Some drafts have also expired.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Cullen Jennings [fluffy@iii.ca]<br>
<b>To:</b> Bernard Aboba [bernard_aboba@hotmail.com]<br>
<b>CC:</b> rtcweb@ietf.org [rtcweb@ietf.org]<br>
<b>Subject:</b> Re: [rtcweb] Question about the status of various drafts</s=
pan></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
Yah, I get your point. I will try and find a way to better represent this o=
n the slides.
<br>
<br>
<br>
On Jul 5, 2013, at 3:25 PM, Bernard Aboba &lt;bernard_aboba@hotmail.com&gt;=
 wrote:<br>
<br>
&gt; I think you need to take dependencies into account.&nbsp;&nbsp; There =
are several reasons for this:
<br>
&gt; <br>
&gt;&nbsp;&nbsp;&nbsp; a. Until you've gone through the dependency graph fo=
r the items, the full list of documents that need to be tracked isn't known=
.
<br>
&gt;&nbsp;&nbsp;&nbsp; b. Estimates of completion are dependent on the prog=
ress of normative dependencies (e.g. they can't be published until the depe=
ndencies are resolved).
<br>
&gt; <br>
&gt; As an example,&nbsp; draft-ietf-rtcweb-overview (which is listed as 80=
 percent complete) has normative references to the following drafts:
<br>
&gt; <br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-mmusic-sctp-sdp] 70<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-audio] 50<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-data-channel] 70<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-jsep] 50<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-rtp-usage] 40<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-security] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-security-arch] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.nandakumar-rtcweb-stun-uri] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; <br>
&gt; Since draft-ietf-rtcweb-overview is dependent on a document (rtp usage=
) which only has an estimate of 40 percent done) I'd say that the estimate =
is probably closer to 40 than 80.
<br>
&gt; <br>
&gt; <br>
&gt; &gt; From: fluffy@iii.ca<br>
&gt; &gt; Date: Thu, 4 Jul 2013 17:47:20 -0700<br>
&gt; &gt; To: rtcweb@ietf.org<br>
&gt; &gt; Subject: [rtcweb] Question about the status of various drafts<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; As part of putting together some chair slides for Berlin, I'm try=
ing to get a very rough idea of where things are. I'd like to get folks inp=
ut on what percent done various drafts are where Done is a published RFC.
<br>
&gt; &gt; <br>
&gt; &gt; This is nearly impossible to do at any level of accuracy but I do=
 want to get a vague idea. I took a very rough stab below just to have a st=
arting point to get the conversation going. I'm sure my numbers are mostly =
wrong but I have tried to ask a few people
 and sort of take the central cluster of the guess. If you think I am way o=
ut, please provide some feedback of what you think the number actually is.<=
br>
&gt; &gt; <br>
&gt; &gt; I'm happy to get feedback on specific drafts or feedback of the f=
orm they are all too low or too high. I do encourage people to realisticall=
y look at the data tracker to see how long other drafts have taken for each=
 stage to get a baseline instead of just
 going with their gut feel. <br>
&gt; &gt; <br>
&gt; &gt; Thanks, Cullen<br>
&gt; &gt; <br>
&gt; &gt; PS - if you are an author and look at the number next to your dra=
ft and think it is way off, please please, say something.
<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; draft-ietf-rtcweb-overview&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-use-cases-and-requirements&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; <a href=3D"http://dev.w3.org/2011/webrtc/editor/getusermedia.html=
">http://dev.w3.org/2011/webrtc/editor/getusermedia.html</a>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-burnett-rtcweb-constraints-registry&nbsp;&nbsp;&nbsp; 80<br=
>
&gt; &gt; <a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html">http=
://dev.w3.org/2011/webrtc/editor/webrtc.html</a>&nbsp;&nbsp;&nbsp;&nbsp; 50=
<br>
&gt; &gt; draft-nandakumar-rtcweb-stun-uri&nbsp;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-petithuguenin-behave-turn-uris&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-security&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-security-arch&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 80<=
br>
&gt; &gt; draft-muthu-behave-consent-freshness-02 50<br>
&gt; &gt; draft-ietf-rtcweb-data-channel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; 70<br>
&gt; &gt; draft-ietf-mmusic-sctp-sdp&nbsp;&nbsp; 70<br>
&gt; &gt; draft-jesup-rtcweb-data-protocol 50<br>
&gt; &gt; draft-ietf-tsvwg-sctp-dtls-encaps&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-ietf-rtcweb-rtp-usage&nbsp; 40<br>
&gt; &gt; draft-ietf-avtcore-6222bis&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-avtcore-avp-codecs&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; 80<br>
&gt; &gt; draft-ietf-avtcore-srtp-encrypted-header-ext&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-avtext-multiple-clock-rates&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; 70<br>
&gt; &gt; draft-lennox-rtcweb-rtp-media-type-mux 80<br>
&gt; &gt; draft-ietf-avtcore-rtp-circuit-breakers&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; 70<br>
&gt; &gt; draft-ietf-avtcore-multi-media-rtp-session 70<br>
&gt; &gt; draft-lennox-avtcore-rtp-multi-stream&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-ietf-rtcweb-jsep&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-ietf-mmusic-msid&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-rescorla-mmusic-ice-trickle&nbsp;&nbsp;&nbsp; 25<br>
&gt; &gt; draft-ietf-mmusic-sdp-bundle-negotiation&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-nandakumar-mmusic-sdp-mux-attributes&nbsp; 25<br>
&gt; &gt; draft-dhesikan-tsvwg-rtcweb-qos&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10 =
or 90<br>
&gt; &gt; draft-ietf-rtcweb-audio&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-ietf-payload-rtp-opus&nbsp; 90<br>
&gt; &gt; draft-ietf-payload-vp8-08&nbsp;&nbsp;&nbsp; 95<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; rtcweb@ietf.org<br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
rtcweb@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><br>
</div>
</span></font>
</body>
</html>

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Cullen Jennings <fluffy@iii.ca>, Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Question about the status of various drafts
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I'd missed that.

By definition an author draft has 0% completion because no working group ha=
s reached consensus on any of the text.

Looks like you (or other RTCWEB chairs) need to start talking to other WG c=
hairs about initiating adoption of drafts as WG items. I don't at the momen=
t see anything in the list that should avoid going the WG route.

Regards

Keith

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: 26 July 2013 19:57
To: Cullen Jennings; Bernard Aboba
Cc: rtcweb_ietf.org
Subject: Re: [rtcweb] Question about the status of various drafts

Hi Cullen,

I may missed something, but it seems a little strange to me when individual=
 drafts, that haven't even been adopted yet, are indicated as 70-80% comple=
te...

Some drafts have also expired.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Cullen Jennings [fluffy@iii.ca]
To: Bernard Aboba [bernard_aboba@hotmail.com]
CC: rtcweb@ietf.org [rtcweb@ietf.org]
Subject: Re: [rtcweb] Question about the status of various drafts

Yah, I get your point. I will try and find a way to better represent this o=
n the slides.


On Jul 5, 2013, at 3:25 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrote=
:

> I think you need to take dependencies into account.   There are several r=
easons for this:
>
>    a. Until you've gone through the dependency graph for the items, the f=
ull list of documents that need to be tracked isn't known.
>    b. Estimates of completion are dependent on the progress of normative =
dependencies (e.g. they can't be published until the dependencies are resol=
ved).
>
> As an example,  draft-ietf-rtcweb-overview (which is listed as 80 percent=
 complete) has normative references to the following drafts:
>
>    [I-D.ietf-mmusic-sctp-sdp] 70
>    [I-D.ietf-rtcweb-audio] 50
>    [I-D.ietf-rtcweb-data-channel] 70
>    [I-D.ietf-rtcweb-jsep] 50
>    [I-D.ietf-rtcweb-rtp-usage] 40
>    [I-D.ietf-rtcweb-security] 80
>    [I-D.ietf-rtcweb-security-arch] 80
>    [I-D.nandakumar-rtcweb-stun-uri] 80
>    [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70
>
> Since draft-ietf-rtcweb-overview is dependent on a document (rtp usage) w=
hich only has an estimate of 40 percent done) I'd say that the estimate is =
probably closer to 40 than 80.
>
>
> > From: fluffy@iii.ca
> > Date: Thu, 4 Jul 2013 17:47:20 -0700
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Question about the status of various drafts
> >
> >
> > As part of putting together some chair slides for Berlin, I'm trying to=
 get a very rough idea of where things are. I'd like to get folks input on =
what percent done various drafts are where Done is a published RFC.
> >
> > This is nearly impossible to do at any level of accuracy but I do want =
to get a vague idea. I took a very rough stab below just to have a starting=
 point to get the conversation going. I'm sure my numbers are mostly wrong =
but I have tried to ask a few people and sort of take the central cluster o=
f the guess. If you think I am way out, please provide some feedback of wha=
t you think the number actually is.
> >
> > I'm happy to get feedback on specific drafts or feedback of the form th=
ey are all too low or too high. I do encourage people to realistically look=
 at the data tracker to see how long other drafts have taken for each stage=
 to get a baseline instead of just going with their gut feel.
> >
> > Thanks, Cullen
> >
> > PS - if you are an author and look at the number next to your draft and=
 think it is way off, please please, say something.
> >
> >
> >
> > draft-ietf-rtcweb-overview   80
> > draft-ietf-rtcweb-use-cases-and-requirements        70
> > http://dev.w3.org/2011/webrtc/editor/getusermedia.html       70
> > draft-burnett-rtcweb-constraints-registry    80
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html     50
> > draft-nandakumar-rtcweb-stun-uri     80
> > draft-petithuguenin-behave-turn-uris         80
> > draft-ietf-rtcweb-security   80
> > draft-ietf-rtcweb-security-arch      80
> > draft-muthu-behave-consent-freshness-02 50
> > draft-ietf-rtcweb-data-channel       70
> > draft-ietf-mmusic-sctp-sdp   70
> > draft-jesup-rtcweb-data-protocol 50
> > draft-ietf-tsvwg-sctp-dtls-encaps    70
> > draft-ietf-rtcweb-rtp-usage  40
> > draft-ietf-avtcore-6222bis   80
> > draft-ietf-avtcore-avp-codecs        80
> > draft-ietf-avtcore-srtp-encrypted-header-ext        80
> > draft-ietf-avtext-multiple-clock-rates       70
> > draft-lennox-rtcweb-rtp-media-type-mux 80
> > draft-ietf-avtcore-rtp-circuit-breakers      70
> > draft-ietf-avtcore-multi-media-rtp-session 70
> > draft-lennox-avtcore-rtp-multi-stream        70
> > draft-ietf-rtcweb-jsep       50
> > draft-ietf-mmusic-msid       70
> > draft-rescorla-mmusic-ice-trickle    25
> > draft-ietf-mmusic-sdp-bundle-negotiation    50
> > draft-nandakumar-mmusic-sdp-mux-attributes  25
> > draft-dhesikan-tsvwg-rtcweb-qos      10 or 90
> > draft-ietf-rtcweb-audio      50
> > draft-ietf-payload-rtp-opus  90
> > draft-ietf-payload-vp8-08    95
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">I&#8217;d missed that.<o:p></o:p></spa=
n></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">By definition an author draft has 0% c=
ompletion because no working group has reached consensus on any of the text=
.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Looks like you (or other RTCWEB chairs=
) need to start talking to other WG chairs about initiating adoption of dra=
fts as WG items. I don&#8217;t
 at the moment see anything in the list that should avoid going the WG rout=
e.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Regards<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Keith<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span lang=3D"EN-US" style=3D"font-siz=
e:12.0pt">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN=
-US" style=3D"font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</=
span></font></b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;font-family:Tahoma"> rtcweb-bounces@ietf.org
 [mailto:rtcweb-bounces@ietf.org] <b><span style=3D"font-weight:bold">On Be=
half Of </span>
</b>Christer Holmberg<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> 26 July 2013 19:57<br>
<b><span style=3D"font-weight:bold">To:</span></b> Cullen Jennings; Bernard=
 Aboba<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> rtcweb_ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [rtcweb] Questi=
on about the status of various drafts</span></font><span lang=3D"EN-US"><o:=
p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">Hi Cullen,<o=
:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">&nbsp;<o:p><=
/o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">I may missed=
 something, but it seems a little strange to me when individual drafts, tha=
t haven't even been adopted yet, are indicated
 as 70-80% complete...<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">&nbsp;<o:p><=
/o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">Some drafts =
have also expired.<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">&nbsp;<o:p><=
/o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">Regards,<o:p=
></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">&nbsp;<o:p><=
/o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">Christer<o:p=
></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black">&nbsp;<o:p><=
/o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black"><br>
<br>
Sent from <b><i><span style=3D"font-weight:bold;font-style:italic">Windows<=
/span></i></b> using
</span></font><b><font size=3D"2" color=3D"blue" face=3D"Calibri"><span sty=
le=3D"font-size:11.0pt;font-family:Calibri;color:blue;font-weight:bold">Tou=
chDown</span></font></b><font size=3D"2" color=3D"black" face=3D"Calibri"><=
span style=3D"font-size:11.0pt;font-family:Calibri;
color:black">
 (www.nitrodesk.com)<o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:Calibri;color:black"><br>
-----Original Message-----<br>
<b><span style=3D"font-weight:bold">From:</span></b> Cullen Jennings [fluff=
y@iii.ca]<br>
<b><span style=3D"font-weight:bold">To:</span></b> Bernard Aboba [bernard_a=
boba@hotmail.com]<br>
<b><span style=3D"font-weight:bold">CC:</span></b> rtcweb@ietf.org [rtcweb@=
ietf.org]<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [rtcweb] Questi=
on about the status of various drafts<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Times New Roman"><span styl=
e=3D"font-size:
10.0pt"><br>
Yah, I get your point. I will try and find a way to better represent this o=
n the slides.
<br>
<br>
<br>
On Jul 5, 2013, at 3:25 PM, Bernard Aboba &lt;bernard_aboba@hotmail.com&gt;=
 wrote:<br>
<br>
&gt; I think you need to take dependencies into account.&nbsp;&nbsp; There =
are several reasons for this:
<br>
&gt; <br>
&gt;&nbsp;&nbsp;&nbsp; a. Until you've gone through the dependency graph fo=
r the items, the full list of documents that need to be tracked isn't known=
.
<br>
&gt;&nbsp;&nbsp;&nbsp; b. Estimates of completion are dependent on the prog=
ress of normative dependencies (e.g. they can't be published until the depe=
ndencies are resolved).
<br>
&gt; <br>
&gt; As an example,&nbsp; draft-ietf-rtcweb-overview (which is listed as 80=
 percent complete) has normative references to the following drafts:
<br>
&gt; <br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-mmusic-sctp-sdp] 70<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-audio] 50<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-data-channel] 70<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-jsep] 50<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-rtp-usage] 40<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-security] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.ietf-rtcweb-security-arch] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.nandakumar-rtcweb-stun-uri] 80<br>
&gt;&nbsp;&nbsp;&nbsp; [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; <br>
&gt; Since draft-ietf-rtcweb-overview is dependent on a document (rtp usage=
) which only has an estimate of 40 percent done) I'd say that the estimate =
is probably closer to 40 than 80.
<br>
&gt; <br>
&gt; <br>
&gt; &gt; From: fluffy@iii.ca<br>
&gt; &gt; Date: Thu, 4 Jul 2013 17:47:20 -0700<br>
&gt; &gt; To: rtcweb@ietf.org<br>
&gt; &gt; Subject: [rtcweb] Question about the status of various drafts<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; As part of putting together some chair slides for <st1:State w:st=
=3D"on"><st1:place w:st=3D"on">Berlin</st1:place></st1:State>, I'm trying t=
o get a very rough idea of where things are. I'd like to get folks input on=
 what percent done various drafts are where
 Done is a published RFC. <br>
&gt; &gt; <br>
&gt; &gt; This is nearly impossible to do at any level of accuracy but I do=
 want to get a vague idea. I took a very rough stab below just to have a st=
arting point to get the conversation going. I'm sure my numbers are mostly =
wrong but I have tried to ask a few people
 and sort of take the central cluster of the guess. If you think I am way o=
ut, please provide some feedback of what you think the number actually is.<=
br>
&gt; &gt; <br>
&gt; &gt; I'm happy to get feedback on specific drafts or feedback of the f=
orm they are all too low or too high. I do encourage people to realisticall=
y look at the data tracker to see how long other drafts have taken for each=
 stage to get a baseline instead of just
 going with their gut feel. <br>
&gt; &gt; <br>
&gt; &gt; Thanks, Cullen<br>
&gt; &gt; <br>
&gt; &gt; PS - if you are an author and look at the number next to your dra=
ft and think it is way off, please please, say something.
<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; draft-ietf-rtcweb-overview&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-use-cases-and-requirements&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; <a href=3D"http://dev.w3.org/2011/webrtc/editor/getusermedia.html=
">http://dev.w3.org/2011/webrtc/editor/getusermedia.html</a>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-burnett-rtcweb-constraints-registry&nbsp;&nbsp;&nbsp; 80<br=
>
&gt; &gt; <a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html">http=
://dev.w3.org/2011/webrtc/editor/webrtc.html</a>&nbsp;&nbsp;&nbsp;&nbsp; 50=
<br>
&gt; &gt; draft-nandakumar-rtcweb-stun-uri&nbsp;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-petithuguenin-behave-turn-uris&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-security&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-rtcweb-security-arch&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 80<=
br>
&gt; &gt; draft-muthu-behave-consent-freshness-02 50<br>
&gt; &gt; draft-ietf-rtcweb-data-channel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; 70<br>
&gt; &gt; draft-ietf-mmusic-sctp-sdp&nbsp;&nbsp; 70<br>
&gt; &gt; draft-jesup-rtcweb-data-protocol 50<br>
&gt; &gt; draft-ietf-tsvwg-sctp-dtls-encaps&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-ietf-rtcweb-rtp-usage&nbsp; 40<br>
&gt; &gt; draft-ietf-avtcore-6222bis&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-avtcore-avp-codecs&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; 80<br>
&gt; &gt; draft-ietf-avtcore-srtp-encrypted-header-ext&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; 80<br>
&gt; &gt; draft-ietf-avtext-multiple-clock-rates&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; 70<br>
&gt; &gt; draft-lennox-rtcweb-rtp-media-type-mux 80<br>
&gt; &gt; draft-ietf-avtcore-rtp-circuit-breakers&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; 70<br>
&gt; &gt; draft-ietf-avtcore-multi-media-rtp-session 70<br>
&gt; &gt; draft-lennox-avtcore-rtp-multi-stream&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-ietf-rtcweb-jsep&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-ietf-mmusic-msid&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 70<br>
&gt; &gt; draft-rescorla-mmusic-ice-trickle&nbsp;&nbsp;&nbsp; 25<br>
&gt; &gt; draft-ietf-mmusic-sdp-bundle-negotiation&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-nandakumar-mmusic-sdp-mux-attributes&nbsp; 25<br>
&gt; &gt; draft-dhesikan-tsvwg-rtcweb-qos&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10 =
or 90<br>
&gt; &gt; draft-ietf-rtcweb-audio&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 50<br>
&gt; &gt; draft-ietf-payload-rtp-opus&nbsp; 90<br>
&gt; &gt; draft-ietf-payload-vp8-08&nbsp;&nbsp;&nbsp; 95<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; rtcweb@ietf.org<br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
rtcweb@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><o:p></o:p></span></font></p>
</div>
</div>
</div>
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From: Cullen Jennings <fluffy@iii.ca>
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I'm in the process of updating it it all - I'll try and make reasonable =
estimates. Note some of the individual drafts are planned to be AD =
sponsored.=20


On Jul 26, 2013, at 12:56 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi Cullen,
> =20
> I may missed something, but it seems a little strange to me when =
individual drafts, that haven't even been adopted yet, are indicated as =
70-80% complete...
> =20
> Some drafts have also expired.
> =20
> Regards,
> =20
> Christer
> =20
>=20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)
>=20
> -----Original Message-----
> From: Cullen Jennings [fluffy@iii.ca]
> To: Bernard Aboba [bernard_aboba@hotmail.com]
> CC: rtcweb@ietf.org [rtcweb@ietf.org]
> Subject: Re: [rtcweb] Question about the status of various drafts
>=20
> Yah, I get your point. I will try and find a way to better represent =
this on the slides.=20
>=20
>=20
> On Jul 5, 2013, at 3:25 PM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:
>=20
> > I think you need to take dependencies into account.   There are =
several reasons for this:=20
> >=20
> >    a. Until you've gone through the dependency graph for the items, =
the full list of documents that need to be tracked isn't known.=20
> >    b. Estimates of completion are dependent on the progress of =
normative dependencies (e.g. they can't be published until the =
dependencies are resolved).=20
> >=20
> > As an example,  draft-ietf-rtcweb-overview (which is listed as 80 =
percent complete) has normative references to the following drafts:=20
> >=20
> >    [I-D.ietf-mmusic-sctp-sdp] 70
> >    [I-D.ietf-rtcweb-audio] 50
> >    [I-D.ietf-rtcweb-data-channel] 70
> >    [I-D.ietf-rtcweb-jsep] 50
> >    [I-D.ietf-rtcweb-rtp-usage] 40
> >    [I-D.ietf-rtcweb-security] 80
> >    [I-D.ietf-rtcweb-security-arch] 80
> >    [I-D.nandakumar-rtcweb-stun-uri] 80
> >    [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70
> >            =20
> > Since draft-ietf-rtcweb-overview is dependent on a document (rtp =
usage) which only has an estimate of 40 percent done) I'd say that the =
estimate is probably closer to 40 than 80.=20
> >=20
> >=20
> > > From: fluffy@iii.ca
> > > Date: Thu, 4 Jul 2013 17:47:20 -0700
> > > To: rtcweb@ietf.org
> > > Subject: [rtcweb] Question about the status of various drafts
> > >=20
> > >=20
> > > As part of putting together some chair slides for Berlin, I'm =
trying to get a very rough idea of where things are. I'd like to get =
folks input on what percent done various drafts are where Done is a =
published RFC.=20
> > >=20
> > > This is nearly impossible to do at any level of accuracy but I do =
want to get a vague idea. I took a very rough stab below just to have a =
starting point to get the conversation going. I'm sure my numbers are =
mostly wrong but I have tried to ask a few people and sort of take the =
central cluster of the guess. If you think I am way out, please provide =
some feedback of what you think the number actually is.
> > >=20
> > > I'm happy to get feedback on specific drafts or feedback of the =
form they are all too low or too high. I do encourage people to =
realistically look at the data tracker to see how long other drafts have =
taken for each stage to get a baseline instead of just going with their =
gut feel.=20
> > >=20
> > > Thanks, Cullen
> > >=20
> > > PS - if you are an author and look at the number next to your =
draft and think it is way off, please please, say something.=20
> > >=20
> > >=20
> > >=20
> > > draft-ietf-rtcweb-overview   80
> > > draft-ietf-rtcweb-use-cases-and-requirements        70
> > > http://dev.w3.org/2011/webrtc/editor/getusermedia.html       70
> > > draft-burnett-rtcweb-constraints-registry    80
> > > http://dev.w3.org/2011/webrtc/editor/webrtc.html     50
> > > draft-nandakumar-rtcweb-stun-uri     80
> > > draft-petithuguenin-behave-turn-uris         80
> > > draft-ietf-rtcweb-security   80
> > > draft-ietf-rtcweb-security-arch      80
> > > draft-muthu-behave-consent-freshness-02 50
> > > draft-ietf-rtcweb-data-channel       70
> > > draft-ietf-mmusic-sctp-sdp   70
> > > draft-jesup-rtcweb-data-protocol 50
> > > draft-ietf-tsvwg-sctp-dtls-encaps    70
> > > draft-ietf-rtcweb-rtp-usage  40
> > > draft-ietf-avtcore-6222bis   80
> > > draft-ietf-avtcore-avp-codecs        80
> > > draft-ietf-avtcore-srtp-encrypted-header-ext        80
> > > draft-ietf-avtext-multiple-clock-rates       70
> > > draft-lennox-rtcweb-rtp-media-type-mux 80
> > > draft-ietf-avtcore-rtp-circuit-breakers      70
> > > draft-ietf-avtcore-multi-media-rtp-session 70
> > > draft-lennox-avtcore-rtp-multi-stream        70
> > > draft-ietf-rtcweb-jsep       50
> > > draft-ietf-mmusic-msid       70
> > > draft-rescorla-mmusic-ice-trickle    25
> > > draft-ietf-mmusic-sdp-bundle-negotiation    50
> > > draft-nandakumar-mmusic-sdp-mux-attributes  25
> > > draft-dhesikan-tsvwg-rtcweb-qos      10 or 90
> > > draft-ietf-rtcweb-audio      50
> > > draft-ietf-payload-rtp-opus  90
> > > draft-ietf-payload-vp8-08    95
> > >=20
> > >=20
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Cullen Jennings <fluffy@iii.ca>
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To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [rtcweb] A compromise for SDES
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I'm a bit confused about your proposal.=20

=46rom a RTCWeb WG point of view, it seems that the key question is what =
are we suggesting that things that implement WebRTC (including browsers) =
do? Are you proposing that SDES in browsers be optional, mandatory, or =
not allowed?


On Jul 13, 2013, at 11:07 AM, Hadriel Kaplan <hadriel.kaplan@oracle.com> =
wrote:

> Howdy,
> Someone's asked me to explain the compromise I plan on presenting in =
Berlin now, so we have more time to argue over it or chew on it.  The =
reason I was going to wait is because I wanted to explain how I got to =
the point of thinking such a compromise would make sense.  So I'll try =
to do that in this email, which will make this email long.  Nothin' much =
I can do about that.
>=20
> Obviously this is all my personal opinion, not that of my employer, =
and not a statement claiming to be objective fact, etc.
>=20
> Background:
> I think many people would agree that SDES has some benefits compared =
to DTLS for those of us wishing to interface to the SIP world.  Some of =
those benefits are commercial/cost factors, some are more tangible like =
reducing early media clipping.  Whether you agree with such benefits or =
not, so long as WebRTC could support SDES in a sufficiently-secure =
manner, there'd be less concern over having it.  Most of the arguments =
against SDES have been directed at whether it could in fact be =
"sufficiently secure".  Personally I find most of the arguments against =
it being secure enough to be essentially FUD, and/or also apply to =
DTLS-SRTP.
>=20
> On the flip-side, I think the commercial/cost factor for SDES is not a =
strong cause, because I believe the market will bear whatever the extra =
cost may be. (I wasn't sure of this 2 years ago when this all started, =
but I'm fairly confident of it now)  And I think having only DTLS-SRTP =
as MTI would have a marketing benefit for WebRTC as a technology, which =
I value highly.
>=20
> Given those personal feelings, I didn't much care either way, so long =
as a decision was made - although I still preferred having SDES to avoid =
the early media clipping problem.  So when I was going to present on =
this topic back in the Vancouver meeting, my last slide basically said: =
"If we support SDES and it turns out later we were wrong, we can always =
rescind it".  This was based on the notion that browsers get updated all =
the time, especially for security issues.
>=20
> But later it occurred to me that's actually the Achilles' heel of =
supporting SDES in WebRTC, for those of us wanting to gateway this stuff =
to SIP.  Imagine if it turns out we were wrong, and some =
expected-or-unexpected vulnerability is exploited against some large =
WebRTC service provider, and makes the news/slashdot.  Browser vendors =
would instantly have new code removing SDES support, and browsers in =
devices would be updated virtually overnight - some users may take a =
long time to upgrade their specific browsers on their specific devices - =
but many of them would do so within days.  That would be untenable for a =
WebRTC service provider.  Even the smaller Enterprises take a long time =
to upgrade code on their servers, so even for them changing their =
gateways to suddenly support DTLS-SRTP would be unrealistic.  Imagine =
what it would be for larger providers, who might even have to deploy =
more servers to handle the sudden additional overhead of DTLS-SRTP.  =
Meanwhile a growing perc
> entage of their users can no longer use the service.  That's bad mojo.
>=20
> The things that a WebRTC-to-SIP service provider can control, and =
arguably the much larger use-case for this stuff to begin with, are not =
really true "browsers" anyway - they're web-based-framework device Apps, =
provided by the service provider to begin with.  I.e., the apps they =
build using web application framework things like PhoneGap/Cordova, =
Appcelerator Titanium, etc.  I mean using a real Browser is nice for =
ad-hoc stuff, like being able to click on a "talk to a rep" button on a =
website, or a webex/meetecho type thing; but for real communication =
"service", whether it be as a subscriber of a traditional carrier, or =
Skype, or an employee of an Enterprise, or a call-center attendant, or =
whatever - for those most people would never want to have to open a =
browser just to receive/make calls.  They'd give you an app to use =
instead.
>=20
> So it's the app use-case that would benefit the most from SDES, =
especially for issues like media clipping.  And the app use-case doesn't =
have the security concerns nor concerns for unforeseen overnight =
updates.
>=20
> The Compromise:
> So given that background, I was planning to propose that the security =
doc keep DTLS-SRTP as the only MTI mechanism for browsers, BUT to add a =
statement that web-based application frameworks SHOULD also support =
SDES. (with text about why and how, etc.)
>=20
> I don't think this is too controversial, because web-based frameworks =
are never beholden to following browser behavior anyway - they're used =
to build a native application, and native applications have a very =
different security/threat model in practice.  They're written for a =
specific purpose, and installed by users from known sources for that =
known purpose.  Technically, afaik, nothing we do in RTCWEB WG or W3C's =
WEBRTC group have any requirements/mandates for native applications =
anyway - an app maker would just ignore something they don't think =
applies to them - but I think web-based frameworks do generally try to =
follow W3C models for Javascript APIs, and will likely read the IETF =
RFCs for the media-layer stuff too.  So I think having this SHOULD =
statement would be beneficial.
>=20
> Or if the WG prefers, we could even write a separate doc on what a =
web-based framework maker should consider supporting/not-supporting. (I =
can imagine a few other things they might want to offer that a browser =
can't)
>=20
> -hadriel
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Question about the status of various drafts
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On 07/05/2013 11:25 PM, Bernard Aboba wrote:
> I think you need to take dependencies into account.   There are
> several reasons for this: 
>
>    a. Until you've gone through the dependency graph for the items,
> the full list of documents that need to be tracked isn't known. 
>    b. Estimates of completion are dependent on the progress of
> normative dependencies (e.g. they can't be published until the
> dependencies are resolved). 
>
> As an example,  draft-ietf-rtcweb-overview (which is listed as 80
> percent complete) has normative references to the following drafts: 
>
>    [I-D.ietf-mmusic-sctp-sdp] 70
>    [I-D.ietf-rtcweb-audio] 50
>    [I-D.ietf-rtcweb-data-channel] 70
>    [I-D.ietf-rtcweb-jsep] 50
>    [I-D.ietf-rtcweb-rtp-usage] 40
>    [I-D.ietf-rtcweb-security] 80
>    [I-D.ietf-rtcweb-security-arch] 80
>    [I-D.nandakumar-rtcweb-stun-uri] 80
>    [I-D.tuexen-tsvwg-sctp-dtls-encaps] 70
>             
> Since draft-ietf-rtcweb-overview is dependent on a document (rtp
> usage) which only has an estimate of 40 percent done) I'd say that the
> estimate is probably closer to 40 than 80.

I don't know exactly what Cullen meant by the percentages, but I don't
think he intended them to be sequential the way publication dependencies
are.

If, for instance, he was thinking about how many percent of the text
needed to change semantically before it was done, I think 80% for
"overview" may be right (unless we choose to change the approach by some
large delta).

If you want to draw up a dependency graph .... well, since -overview-
was intended to be the one that points to all the others, it's
reasonable to put it last in the graph. That's what it's for.


>
>
> > From: fluffy@iii.ca
> > Date: Thu, 4 Jul 2013 17:47:20 -0700
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Question about the status of various drafts
> >
> >
> > As part of putting together some chair slides for Berlin, I'm trying
> to get a very rough idea of where things are. I'd like to get folks
> input on what percent done various drafts are where Done is a
> published RFC.
> >
> > This is nearly impossible to do at any level of accuracy but I do
> want to get a vague idea. I took a very rough stab below just to have
> a starting point to get the conversation going. I'm sure my numbers
> are mostly wrong but I have tried to ask a few people and sort of take
> the central cluster of the guess. If you think I am way out, please
> provide some feedback of what you think the number actually is.
> >
> > I'm happy to get feedback on specific drafts or feedback of the form
> they are all too low or too high. I do encourage people to
> realistically look at the data tracker to see how long other drafts
> have taken for each stage to get a baseline instead of just going with
> their gut feel.
> >
> > Thanks, Cullen
> >
> > PS - if you are an author and look at the number next to your draft
> and think it is way off, please please, say something.
> >
> >
> >
> > draft-ietf-rtcweb-overview 80
> > draft-ietf-rtcweb-use-cases-and-requirements 70
> > http://dev.w3.org/2011/webrtc/editor/getusermedia.html 70
> > draft-burnett-rtcweb-constraints-registry 80
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html 50
> > draft-nandakumar-rtcweb-stun-uri 80
> > draft-petithuguenin-behave-turn-uris 80
> > draft-ietf-rtcweb-security 80
> > draft-ietf-rtcweb-security-arch 80
> > draft-muthu-behave-consent-freshness-02 50
> > draft-ietf-rtcweb-data-channel 70
> > draft-ietf-mmusic-sctp-sdp 70
> > draft-jesup-rtcweb-data-protocol 50
> > draft-ietf-tsvwg-sctp-dtls-encaps 70
> > draft-ietf-rtcweb-rtp-usage 40
> > draft-ietf-avtcore-6222bis 80
> > draft-ietf-avtcore-avp-codecs 80
> > draft-ietf-avtcore-srtp-encrypted-header-ext 80
> > draft-ietf-avtext-multiple-clock-rates 70
> > draft-lennox-rtcweb-rtp-media-type-mux 80
> > draft-ietf-avtcore-rtp-circuit-breakers 70
> > draft-ietf-avtcore-multi-media-rtp-session 70
> > draft-lennox-avtcore-rtp-multi-stream 70
> > draft-ietf-rtcweb-jsep 50
> > draft-ietf-mmusic-msid 70
> > draft-rescorla-mmusic-ice-trickle 25
> > draft-ietf-mmusic-sdp-bundle-negotiation 50
> > draft-nandakumar-mmusic-sdp-mux-attributes 25
> > draft-dhesikan-tsvwg-rtcweb-qos 10 or 90
> > draft-ietf-rtcweb-audio 50
> > draft-ietf-payload-rtp-opus 90
> > draft-ietf-payload-vp8-08 95
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 07/05/2013 11:25 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W823344CCF942CC7B75815E937D0@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">I think you need to take dependencies into account.
        &nbsp; There are several reasons for this:&nbsp;
        <div><br>
          <div>&nbsp; &nbsp;a. Until you've gone through the dependency graph for
            the items, the full list of documents that need to be
            tracked isn't known.&nbsp;</div>
          <div>&nbsp; &nbsp;b. Estimates of completion are dependent on the
            progress of normative dependencies (e.g. they can't be
            published until the dependencies are resolved).&nbsp;<br>
            <div><br>
            </div>
            <div>As an example, &nbsp;draft-ietf-rtcweb-overview (which is
              listed as 80 percent complete) has normative references to
              the following drafts:&nbsp;
              <div>
                <div><br>
                </div>
                <div>&nbsp; &nbsp;[I-D.ietf-mmusic-sctp-sdp] 70</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-audio] 50</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-data-channel] 70</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-jsep] 50</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-rtp-usage] 40</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-security] 80</div>
                <div>&nbsp; &nbsp;[I-D.ietf-rtcweb-security-arch] 80</div>
                <div>&nbsp; &nbsp;[I-D.nandakumar-rtcweb-stun-uri] 80</div>
                <div>&nbsp; &nbsp;[I-D.tuexen-tsvwg-sctp-dtls-encaps] 70</div>
                <div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;</div>
                <div>Since draft-ietf-rtcweb-overview is dependent on a
                  document (rtp usage) which only has an estimate of 40
                  percent done) I'd say that the estimate is probably
                  closer to 40 than 80. <br>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I don't know exactly what Cullen meant by the percentages, but I
    don't think he intended them to be sequential the way publication
    dependencies are.<br>
    <br>
    If, for instance, he was thinking about how many percent of the text
    needed to change semantically before it was done, I think 80% for
    "overview" may be right (unless we choose to change the approach by
    some large delta).<br>
    <br>
    If you want to draw up a dependency graph .... well, since
    -overview- was intended to be the one that points to all the others,
    it's reasonable to put it last in the graph. That's what it's for.<br>
    <br>
    <br>
    <blockquote cite="mid:BLU169-W823344CCF942CC7B75815E937D0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <div>
            <div>
              <div>
                <div><br>
                </div>
                <div><br>
                  <div>&gt; From: <a class="moz-txt-link-abbreviated" href="mailto:fluffy@iii.ca">fluffy@iii.ca</a><br>
                    &gt; Date: Thu, 4 Jul 2013 17:47:20 -0700<br>
                    &gt; To: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    &gt; Subject: [rtcweb] Question about the status of
                    various drafts<br>
                    &gt; <br>
                    &gt; <br>
                    &gt; As part of putting together some chair slides
                    for Berlin, I'm trying to get a very rough idea of
                    where things are. I'd like to get folks input on
                    what percent done various drafts are where Done is a
                    published RFC. <br>
                    &gt; <br>
                    &gt; This is nearly impossible to do at any level of
                    accuracy but I do want to get a vague idea. I took a
                    very rough stab below just to have a starting point
                    to get the conversation going. I'm sure my numbers
                    are mostly wrong but I have tried to ask a few
                    people and sort of take the central cluster of the
                    guess. If you think I am way out, please provide
                    some feedback of what you think the number actually
                    is.<br>
                    &gt; <br>
                    &gt; I'm happy to get feedback on specific drafts or
                    feedback of the form they are all too low or too
                    high. I do encourage people to realistically look at
                    the data tracker to see how long other drafts have
                    taken for each stage to get a baseline instead of
                    just going with their gut feel. <br>
                    &gt; <br>
                    &gt; Thanks, Cullen<br>
                    &gt; <br>
                    &gt; PS - if you are an author and look at the
                    number next to your draft and think it is way off,
                    please please, say something. <br>
                    &gt; <br>
                    &gt; <br>
                    &gt; <br>
                    &gt; draft-ietf-rtcweb-overview 80<br>
                    &gt; draft-ietf-rtcweb-use-cases-and-requirements 70<br>
                    &gt;
                    <a class="moz-txt-link-freetext" href="http://dev.w3.org/2011/webrtc/editor/getusermedia.html">http://dev.w3.org/2011/webrtc/editor/getusermedia.html</a>
                    70<br>
                    &gt; draft-burnett-rtcweb-constraints-registry 80<br>
                    &gt;
                    <a class="moz-txt-link-freetext" href="http://dev.w3.org/2011/webrtc/editor/webrtc.html">http://dev.w3.org/2011/webrtc/editor/webrtc.html</a> 50<br>
                    &gt; draft-nandakumar-rtcweb-stun-uri 80<br>
                    &gt; draft-petithuguenin-behave-turn-uris 80<br>
                    &gt; draft-ietf-rtcweb-security 80<br>
                    &gt; draft-ietf-rtcweb-security-arch 80<br>
                    &gt; draft-muthu-behave-consent-freshness-02 50<br>
                    &gt; draft-ietf-rtcweb-data-channel 70<br>
                    &gt; draft-ietf-mmusic-sctp-sdp 70<br>
                    &gt; draft-jesup-rtcweb-data-protocol 50<br>
                    &gt; draft-ietf-tsvwg-sctp-dtls-encaps 70<br>
                    &gt; draft-ietf-rtcweb-rtp-usage 40<br>
                    &gt; draft-ietf-avtcore-6222bis 80<br>
                    &gt; draft-ietf-avtcore-avp-codecs 80<br>
                    &gt; draft-ietf-avtcore-srtp-encrypted-header-ext 80<br>
                    &gt; draft-ietf-avtext-multiple-clock-rates 70<br>
                    &gt; draft-lennox-rtcweb-rtp-media-type-mux 80<br>
                    &gt; draft-ietf-avtcore-rtp-circuit-breakers 70<br>
                    &gt; draft-ietf-avtcore-multi-media-rtp-session 70<br>
                    &gt; draft-lennox-avtcore-rtp-multi-stream 70<br>
                    &gt; draft-ietf-rtcweb-jsep 50<br>
                    &gt; draft-ietf-mmusic-msid 70<br>
                    &gt; draft-rescorla-mmusic-ice-trickle 25<br>
                    &gt; draft-ietf-mmusic-sdp-bundle-negotiation 50<br>
                    &gt; draft-nandakumar-mmusic-sdp-mux-attributes 25<br>
                    &gt; draft-dhesikan-tsvwg-rtcweb-qos 10 or 90<br>
                    &gt; draft-ietf-rtcweb-audio 50<br>
                    &gt; draft-ietf-payload-rtp-opus 90<br>
                    &gt; draft-ietf-payload-vp8-08 95<br>
                    &gt; <br>
                    &gt; <br>
                    &gt; _______________________________________________<br>
                    &gt; rtcweb mailing list<br>
                    &gt; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    &gt; <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                  </div>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------020804010305020801010101--

From martin.thomson@gmail.com  Sat Jul 27 08:47:35 2013
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From: Martin Thomson <martin.thomson@gmail.com>
To: Dan Wing <dwing@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Security Architecture -07 review
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On 25 July 2013 17:34, Dan Wing <dwing@cisco.com> wrote:
> We could envision a more sophisticated mechanism, based on lots of differ=
ent criteria, where different public/private key pairs are used and where s=
ome are destroyed immediately after certain calls.  For example, I might wa=
nt to use a public/private key pair for all of my work calls and leave that=
 static (as I want that identity to stick with me) but an ephemeral, one-ti=
me-only public/private key pair when not engaging in phone calls related to=
 my work, such as calling my psychologist.

I believe that we've (EKR and I, I don't know if this made it to
meetings or lists) discussed the need for some application-level
control over these credentials.  I think that the idea was to allow
sites to specify a credential key that would be used to index a table
of credentials.  Of course, every origin would have to use a different
table, which could impose some storage requirements on the browser,
requiring some sort or space management to avoid exploitation.  By
selecting a random credential key for each call (bear with me
regarding terminology, I'm making this up as I go) a site could
effectively create a new set of credentials for that call.

Of course, the site could also choose to generate new credentials for
every call to make it impossible (or at least very difficult) to audit
calls made on that site, if that is something we cared about.  I do
think that in the absence of the identity provider system, that is the
only advantage that DTLS-SRTP brings, so we'd have to be quite careful
here.  I don't know what the right answer is.

From lars@netapp.com  Sun Jul 28 05:06:22 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] No Interim on SDES at this juncture
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Hadriel Kaplan wrote:
> Right, and I'm hoping the Browser vendors can tell us if browsers can safely know whether all the JS running on a given page is from HTTPS or not, and not allow SDES if the JS is not all from (signed) HTTPS.  If not, then I agree that's another negative downside for SDES.  In fact I think it's even a problem for DTLS-SRTP.  I don't know why we'd be ok with using HTTP at all, if we're *truly* concerned about MitM and malicious 3rd party JS attacks.

Yes, this is detectable. As I said at the mic in Beijing, however, this 
is a property that can change at any time, because a page can load 
script from a new, programmatically constructed URL at any point. See 
Section 5.1 of the updated draft-ietf-rtcweb-security-arch-07.

That makes it difficult to design security policy guarantees (it might 
seem "safe" to access a resource when you ask to access it, but very 
difficult to revoke access later when you realize it's not safe). For 
that reason among many others, browsers have been moving to simply block 
mixed content entirely. See 
<https://blog.mozilla.org/security/2013/06/27/mixed-content-blocker-hits-firefox-beta/>. 
Firefox will release this in August. Chrome already does this. IE has 
prompted users since at least IE 7.

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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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IÃ±aki Baz Castillo wrote:
> for WebRTC" (this is true). I also remember that somebody proposed
> that the browser should include a "call history" panel, which clearly
> shows a limited and wrong WWW vision.

I recall EKR proposing an audit log for camera access. But the reasons 
for this were entirely related to security (an app with persistent 
permissions won't be as ready to abuse them if it knows it will leave a 
trail in logs not under its control).

I thought it was a really good idea. I still do.

From ibc@aliax.net  Mon Jul 29 02:37:45 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Summary of Application Developers' opinions of the current WebRTC API and SDP as a control surface
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2013/7/29 Timothy B. Terriberry <tterriberry@mozilla.com>:
> I=C3=B1aki Baz Castillo wrote:
>>
>> for WebRTC" (this is true). I also remember that somebody proposed
>> that the browser should include a "call history" panel, which clearly
>> shows a limited and wrong WWW vision.
>
>
> I recall EKR proposing an audit log for camera access. But the reasons fo=
r
> this were entirely related to security (an app with persistent permission=
s
> won't be as ready to abuse them if it knows it will leave a trail in logs
> not under its control).
>
> I thought it was a really good idea. I still do.

What I meant was somebody requesting a "Call history panel" in the
browser (similar to the "Favorites panel" or the "Navigation History
panel"). ;)


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Mon, 29 Jul 2013 19:31:55 +0200
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Subject: [rtcweb] SDP is not suitable for WebRTC
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Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
but it was moved to MMUSIC maillist (don't know why since it is about
WebRTC applications design):

http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html

Sorry for the cross-posting but at this point I'm a bit lost and do
not know which is the appropriate group for my concern.



So my concern is:


- Web application with a SIP over WebSocket client running in the web.

- The web user is provided with a conference SIP URI in which there
are *already* 8 participants (5 of them emitting audio and video and 3
just emitting audio).

- The user calls, from his webphone, to the given URI to join the conferenc=
e.



Let's imagine that the JS app knows the number of participant in the confer=
ence.
Let's imagine my browser have mic and webcam.



QUESTION:

How can my browser join the conference without requiring SDP
renegotiation from the server and, at the same time, being able to
send audio/video and receive audio/video from others (different tracks
/ m=3Dlines)?




"SOLUTIONS":



1)

I tell my browser to generate a SDP offer with:

  - 1 send/receive m=3Daudio line.
  - 7 recvonly m=3Daudio line.
  - 1 send/only m=3Dvideo line.
  - 4 recvonly m=3Dvideo line.

(Obviously this is a joke)



2)

SDP seems to allow that the offer and the answer have different number
of m lines (I'm not aware of that but I believe that SDP can do
"everything"). So my browser generates a SDP offer with 1 m=3Daudio line
and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4
m=3Dvideo lines.

Will my browser understand such a SDP answer with more m lines than
its generated offer? I assume NOT.



3)

My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo
line and the server too. And later the server sends re-INVITE with all
the m lines.

Oppss, SDP renegotiation...




SDP is bad for WebRTC. SDP is good for legacy symmetric communications
in which there is a single-track audio communication and, of course,
both endpoints emit audio. But SDP is bad for modern RTC protocols in
which an endpoint can emit tons of tracks to a single endpoint.


Do we really want this for WebRTC 1.0 ?


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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No, this scenario seems more of a beta or maybe an alpha than a polished
1.0 version.
El 29/07/2013 19:34, "I=F1aki Baz Castillo" <ibc@aliax.net> escribi=F3:

> Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
> but it was moved to MMUSIC maillist (don't know why since it is about
> WebRTC applications design):
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html
>
> Sorry for the cross-posting but at this point I'm a bit lost and do
> not know which is the appropriate group for my concern.
>
>
>
> So my concern is:
>
>
> - Web application with a SIP over WebSocket client running in the web.
>
> - The web user is provided with a conference SIP URI in which there
> are *already* 8 participants (5 of them emitting audio and video and 3
> just emitting audio).
>
> - The user calls, from his webphone, to the given URI to join the
> conference.
>
>
>
> Let's imagine that the JS app knows the number of participant in the
> conference.
> Let's imagine my browser have mic and webcam.
>
>
>
> QUESTION:
>
> How can my browser join the conference without requiring SDP
> renegotiation from the server and, at the same time, being able to
> send audio/video and receive audio/video from others (different tracks
> / m=3Dlines)?
>
>
>
>
> "SOLUTIONS":
>
>
>
> 1)
>
> I tell my browser to generate a SDP offer with:
>
>   - 1 send/receive m=3Daudio line.
>   - 7 recvonly m=3Daudio line.
>   - 1 send/only m=3Dvideo line.
>   - 4 recvonly m=3Dvideo line.
>
> (Obviously this is a joke)
>
>
>
> 2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines (I'm not aware of that but I believe that SDP can do
> "everything"). So my browser generates a SDP offer with 1 m=3Daudio line
> and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4
> m=3Dvideo lines.
>
> Will my browser understand such a SDP answer with more m lines than
> its generated offer? I assume NOT.
>
>
>
> 3)
>
> My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo
> line and the server too. And later the server sends re-INVITE with all
> the m lines.
>
> Oppss, SDP renegotiation...
>
>
>
>
> SDP is bad for WebRTC. SDP is good for legacy symmetric communications
> in which there is a single-track audio communication and, of course,
> both endpoints emit audio. But SDP is bad for modern RTC protocols in
> which an endpoint can emit tons of tracks to a single endpoint.
>
>
> Do we really want this for WebRTC 1.0 ?
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>
>

--001a11c2195afa347f04e2aa0538
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">No, this scenario seems more of a beta or maybe an alpha tha=
n a polished 1.0 version.</p>
<div class=3D"gmail_quote">El 29/07/2013 19:34, &quot;I=F1aki Baz Castillo&=
quot; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; escribi=F3=
:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,<br>
but it was moved to MMUSIC maillist (don&#39;t know why since it is about<b=
r>
WebRTC applications design):<br>
<br>
<a href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.htm=
l" target=3D"_blank">http://www.ietf.org/mail-archive/web/mmusic/current/ms=
g11966.html</a><br>
<br>
Sorry for the cross-posting but at this point I&#39;m a bit lost and do<br>
not know which is the appropriate group for my concern.<br>
<br>
<br>
<br>
So my concern is:<br>
<br>
<br>
- Web application with a SIP over WebSocket client running in the web.<br>
<br>
- The web user is provided with a conference SIP URI in which there<br>
are *already* 8 participants (5 of them emitting audio and video and 3<br>
just emitting audio).<br>
<br>
- The user calls, from his webphone, to the given URI to join the conferenc=
e.<br>
<br>
<br>
<br>
Let&#39;s imagine that the JS app knows the number of participant in the co=
nference.<br>
Let&#39;s imagine my browser have mic and webcam.<br>
<br>
<br>
<br>
QUESTION:<br>
<br>
How can my browser join the conference without requiring SDP<br>
renegotiation from the server and, at the same time, being able to<br>
send audio/video and receive audio/video from others (different tracks<br>
/ m=3Dlines)?<br>
<br>
<br>
<br>
<br>
&quot;SOLUTIONS&quot;:<br>
<br>
<br>
<br>
1)<br>
<br>
I tell my browser to generate a SDP offer with:<br>
<br>
=A0 - 1 send/receive m=3Daudio line.<br>
=A0 - 7 recvonly m=3Daudio line.<br>
=A0 - 1 send/only m=3Dvideo line.<br>
=A0 - 4 recvonly m=3Dvideo line.<br>
<br>
(Obviously this is a joke)<br>
<br>
<br>
<br>
2)<br>
<br>
SDP seems to allow that the offer and the answer have different number<br>
of m lines (I&#39;m not aware of that but I believe that SDP can do<br>
&quot;everything&quot;). So my browser generates a SDP offer with 1 m=3Daud=
io line<br>
and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4<b=
r>
m=3Dvideo lines.<br>
<br>
Will my browser understand such a SDP answer with more m lines than<br>
its generated offer? I assume NOT.<br>
<br>
<br>
<br>
3)<br>
<br>
My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo<br>
line and the server too. And later the server sends re-INVITE with all<br>
the m lines.<br>
<br>
Oppss, SDP renegotiation...<br>
<br>
<br>
<br>
<br>
SDP is bad for WebRTC. SDP is good for legacy symmetric communications<br>
in which there is a single-track audio communication and, of course,<br>
both endpoints emit audio. But SDP is bad for modern RTC protocols in<br>
which an endpoint can emit tons of tracks to a single endpoint.<br>
<br>
<br>
Do we really want this for WebRTC 1.0 ?<br>
<br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
</blockquote></div>

--001a11c2195afa347f04e2aa0538--

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Date: Mon, 29 Jul 2013 18:45:10 +0100 (BST)
From: Bossiel thioriguel <bossiel@yahoo.fr>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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--1670751155-1967887231-1375119910=:76535
Content-Type: text/plain; charset=iso-8859-1
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You said: "2)=0ASDP seems to allow that the offer and the answer have diffe=
rent number=0Aof m lines=A0"=0A=0A=0ANo at all:=0ARFC 3264:=0AFor each "m=
=3D" line in the offer, there MUST be a corresponding "m=3D"=0Aline in the =
answer.  The answer MUST contain exactly the same number of "m=3D" lines as=
 the offer.  This allows for streams to be matched up based on their order.=
  This implies that if the offer contained zero "m=3D" lines, the answer MU=
ST contain zero "m=3D" lines.=0AMamadou.=0A=0A_____________________________=
___=0A De=A0: I=F1aki Baz Castillo <ibc@aliax.net>=0A=C0=A0: "rtcweb@ietf.o=
rg" <rtcweb@ietf.org>; "public-webrtc@w3.org" <public-webrtc@w3.org> =0AEnv=
oy=E9 le : Lundi 29 juillet 2013 19h31=0AObjet=A0: [rtcweb] SDP is not suit=
able for WebRTC=0A =0A=0AHi, I initiated a thread [*] about Plan-Unified an=
d multiple m lines,=0Abut it was moved to MMUSIC maillist (don't know why s=
ince it is about=0AWebRTC applications design):=0A=0Ahttp://www.ietf.org/ma=
il-archive/web/mmusic/current/msg11966.html=0A=0ASorry for the cross-postin=
g but at this point I'm a bit lost and do=0Anot know which is the appropria=
te group for my concern.=0A=0A=0A=0ASo my concern is:=0A=0A=0A- Web applica=
tion with a SIP over WebSocket client running in the web.=0A=0A- The web us=
er is provided with a conference SIP URI in which there=0Aare *already* 8 p=
articipants (5 of them emitting audio and video and 3=0Ajust emitting audio=
).=0A=0A- The user calls, from his webphone, to the given URI to join the c=
onference.=0A=0A=0A=0ALet's imagine that the JS app knows the number of par=
ticipant in the conference.=0ALet's imagine my browser have mic and webcam.=
=0A=0A=0A=0AQUESTION:=0A=0AHow can my browser join the conference without r=
equiring SDP=0Arenegotiation from the server and, at the same time, being a=
ble to=0Asend audio/video and receive audio/video from others (different tr=
acks=0A/ m=3Dlines)?=0A=0A=0A=0A=0A"SOLUTIONS":=0A=0A=0A=0A1)=0A=0AI tell m=
y browser to generate a SDP offer with:=0A=0A=A0 - 1 send/receive m=3Daudio=
 line.=0A=A0 - 7 recvonly m=3Daudio line.=0A=A0 - 1 send/only m=3Dvideo lin=
e.=0A=A0 - 4 recvonly m=3Dvideo line.=0A=0A(Obviously this is a joke)=0A=0A=
=0A=0A2)=0A=0ASDP seems to allow that the offer and the answer have differe=
nt number=0Aof m lines (I'm not aware of that but I believe that SDP can do=
=0A"everything"). So my browser generates a SDP offer with 1 m=3Daudio line=
=0Aand 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and =
4=0Am=3Dvideo lines.=0A=0AWill my browser understand such a SDP answer with=
 more m lines than=0Aits generated offer? I assume NOT.=0A=0A=0A=0A3)=0A=0A=
My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo=0Ali=
ne and the server too. And later the server sends re-INVITE with all=0Athe =
m lines.=0A=0AOppss, SDP renegotiation...=0A=0A=0A=0A=0ASDP is bad for WebR=
TC. SDP is good for legacy symmetric communications=0Ain which there is a s=
ingle-track audio communication and, of course,=0Aboth endpoints emit audio=
. But SDP is bad for modern RTC protocols in=0Awhich an endpoint can emit t=
ons of tracks to a single endpoint.=0A=0A=0ADo we really want this for WebR=
TC 1.0 ?=0A=0A=0A-- =0AI=F1aki Baz Castillo=0A<ibc@aliax.net>=0A___________=
____________________________________=0Artcweb mailing list=0Artcweb@ietf.or=
g=0Ahttps://www.ietf.org/mailman/listinfo/rtcweb
--1670751155-1967887231-1375119910=:76535
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div><span>You said: =
"</span><span style=3D"font-size: 12pt;">2)</span></div><br>SDP seems to al=
low that the offer and the answer have different number<br>of m lines&nbsp;=
"<div><br><div>No at all:</div><div><span style=3D"font-size: 12pt;">RFC 32=
64:</span></div><div><span style=3D"font-size: 12pt;">For each "m=3D" line =
in the offer, there MUST be a corresponding "m=3D"</span></div><div><pre st=
yle=3D"word-wrap: break-word; white-space: pre-wrap;">   line in the answer=
.  The answer MUST contain exactly the same number=0A   of "m=3D" lines as =
the offer.  This allows for streams to be matched up=0A   based on their or=
der.  This implies that if the offer contained zero=0A   "m=3D" lines, the =
answer MUST contain zero "m=3D" lines.</pre><pre style=3D"word-wrap: break-=
word; white-space: pre-wrap;">Mamadou.</pre>  <div style=3D"font-family: 't=
imes new roman', 'new york', times, serif; font-size: 12pt;"> <div style=3D=
"font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;=
"> <div dir=3D"ltr"> <hr size=3D"1">  <font size=3D"2" face=3D"Arial"> <b><=
span style=3D"font-weight:bold;">De&nbsp;:</span></b> I=F1aki Baz Castillo =
&lt;ibc@aliax.net&gt;<br> <b><span style=3D"font-weight: bold;">=C0&nbsp;:<=
/span></b> "rtcweb@ietf.org" &lt;rtcweb@ietf.org&gt;; "public-webrtc@w3.org=
" &lt;public-webrtc@w3.org&gt; <br> <b><span style=3D"font-weight: bold;">E=
nvoy=E9 le :</span></b> Lundi 29 juillet 2013 19h31<br> <b><span style=3D"f=
ont-weight: bold;">Objet&nbsp;:</span></b> [rtcweb] SDP is not suitable for=
 WebRTC<br> </font> </div> <div class=3D"y_msg_container"><br>Hi, I initiat=
ed a thread [*] about Plan-Unified and multiple m lines,<br>but it was move=
d
 to MMUSIC maillist (don't know why since it is about<br>WebRTC application=
s design):<br><br><a href=3D"http://www.ietf.org/mail-archive/web/mmusic/cu=
rrent/msg11966.html" target=3D"_blank">http://www.ietf.org/mail-archive/web=
/mmusic/current/msg11966.html</a><br><br>Sorry for the cross-posting but at=
 this point I'm a bit lost and do<br>not know which is the appropriate grou=
p for my concern.<br><br><br><br>So my concern is:<br><br><br>- Web applica=
tion with a SIP over WebSocket client running in the web.<br><br>- The web =
user is provided with a conference SIP URI in which there<br>are *already* =
8 participants (5 of them emitting audio and video and 3<br>just emitting a=
udio).<br><br>- The user calls, from his webphone, to the given URI to join=
 the conference.<br><br><br><br>Let's imagine that the JS app knows the num=
ber of participant in the conference.<br>Let's imagine my browser have mic =
and webcam.<br><br><br><br>QUESTION:<br><br>How can my browser join the
 conference without requiring SDP<br>renegotiation from the server and, at =
the same time, being able to<br>send audio/video and receive audio/video fr=
om others (different tracks<br>/ m=3Dlines)?<br><br><br><br><br>"SOLUTIONS"=
:<br><br><br><br>1)<br><br>I tell my browser to generate a SDP offer with:<=
br><br>&nbsp; - 1 send/receive m=3Daudio line.<br>&nbsp; - 7 recvonly m=3Da=
udio line.<br>&nbsp; - 1 send/only m=3Dvideo line.<br>&nbsp; - 4 recvonly m=
=3Dvideo line.<br><br>(Obviously this is a joke)<br><br><br><br>2)<br><br>S=
DP seems to allow that the offer and the answer have different number<br>of=
 m lines (I'm not aware of that but I believe that SDP can do<br>"everythin=
g"). So my browser generates a SDP offer with 1 m=3Daudio line<br>and 1 m=
=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4<br>m=3Dv=
ideo lines.<br><br>Will my browser understand such a SDP answer with more m=
 lines than<br>its generated offer? I assume NOT.<br><br><br><br>3)<br><br>=
My browser
 generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo<br>line and th=
e server too. And later the server sends re-INVITE with all<br>the m lines.=
<br><br>Oppss, SDP renegotiation...<br><br><br><br><br>SDP is bad for WebRT=
C. SDP is good for legacy symmetric communications<br>in which there is a s=
ingle-track audio communication and, of course,<br>both endpoints emit audi=
o. But SDP is bad for modern RTC protocols in<br>which an endpoint can emit=
 tons of tracks to a single endpoint.<br><br><br>Do we really want this for=
 WebRTC 1.0 ?<br><br><br>-- <br>I=F1aki Baz Castillo<br>&lt;<a ymailto=3D"m=
ailto:ibc@aliax.net" href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br=
>_______________________________________________<br>rtcweb mailing list<br>=
<a ymailto=3D"mailto:rtcweb@ietf.org" href=3D"mailto:rtcweb@ietf.org">rtcwe=
b@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br><br><=
br></div>
 </div> </div>  </div></div></div></body></html>
--1670751155-1967887231-1375119910=:76535--

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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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--0015175df19ef975ed04e2aa1e7a
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Thanks a lot for clarifying it. So Christer was wrong ;)

And it makes my scenario even worse. I really hope something will happen
and WebRTC will get rid of SDP...

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 29/07/2013 19:45, "Bossiel thioriguel" <bossiel@yahoo.fr> escribi=C3=B3:

> You said: "2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines "
>
> No at all:
> RFC 3264:
> For each "m=3D" line in the offer, there MUST be a corresponding "m=3D"
>
>    line in the answer.  The answer MUST contain exactly the same number
>    of "m=3D" lines as the offer.  This allows for streams to be matched u=
p
>    based on their order.  This implies that if the offer contained zero
>    "m=3D" lines, the answer MUST contain zero "m=3D" lines.
>
> Mamadou.
>
>   ------------------------------
>  *De :* I=C3=B1aki Baz Castillo <ibc@aliax.net>
> *=C3=80 :* "rtcweb@ietf.org" <rtcweb@ietf.org>; "public-webrtc@w3.org" <
> public-webrtc@w3.org>
> *Envoy=C3=A9 le :* Lundi 29 juillet 2013 19h31
> *Objet :* [rtcweb] SDP is not suitable for WebRTC
>
> Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
> but it was moved to MMUSIC maillist (don't know why since it is about
> WebRTC applications design):
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html
>
> Sorry for the cross-posting but at this point I'm a bit lost and do
> not know which is the appropriate group for my concern.
>
>
>
> So my concern is:
>
>
> - Web application with a SIP over WebSocket client running in the web.
>
> - The web user is provided with a conference SIP URI in which there
> are *already* 8 participants (5 of them emitting audio and video and 3
> just emitting audio).
>
> - The user calls, from his webphone, to the given URI to join the
> conference.
>
>
>
> Let's imagine that the JS app knows the number of participant in the
> conference.
> Let's imagine my browser have mic and webcam.
>
>
>
> QUESTION:
>
> How can my browser join the conference without requiring SDP
> renegotiation from the server and, at the same time, being able to
> send audio/video and receive audio/video from others (different tracks
> / m=3Dlines)?
>
>
>
>
> "SOLUTIONS":
>
>
>
> 1)
>
> I tell my browser to generate a SDP offer with:
>
>   - 1 send/receive m=3Daudio line.
>   - 7 recvonly m=3Daudio line.
>   - 1 send/only m=3Dvideo line.
>   - 4 recvonly m=3Dvideo line.
>
> (Obviously this is a joke)
>
>
>
> 2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines (I'm not aware of that but I believe that SDP can do
> "everything"). So my browser generates a SDP offer with 1 m=3Daudio line
> and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4
> m=3Dvideo lines.
>
> Will my browser understand such a SDP answer with more m lines than
> its generated offer? I assume NOT.
>
>
>
> 3)
>
> My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo
> line and the server too. And later the server sends re-INVITE with all
> the m lines.
>
> Oppss, SDP renegotiation...
>
>
>
>
> SDP is bad for WebRTC. SDP is good for legacy symmetric communications
> in which there is a single-track audio communication and, of course,
> both endpoints emit audio. But SDP is bad for modern RTC protocols in
> which an endpoint can emit tons of tracks to a single endpoint.
>
>
> Do we really want this for WebRTC 1.0 ?
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>

--0015175df19ef975ed04e2aa1e7a
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<p dir=3D"ltr">Thanks a lot for clarifying it. So Christer was wrong ;)</p>
<p dir=3D"ltr">And it makes my scenario even worse. I really hope something=
 will happen and WebRTC will get rid of SDP... <br></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 29/07/2013 19:45, &quot;Bossiel thioriguel&qu=
ot; &lt;<a href=3D"mailto:bossiel@yahoo.fr">bossiel@yahoo.fr</a>&gt; escrib=
i=C3=B3:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style=3D"font-size:12pt;font-family:times new roman,new york,time=
s,serif"><div><span>You said: &quot;</span><span style=3D"font-size:12pt">2=
)</span></div><br>SDP seems to allow that the offer and the answer have dif=
ferent number<br>
of m lines=C2=A0&quot;<div><br><div>No at all:</div><div><span style=3D"fon=
t-size:12pt">RFC 3264:</span></div><div><span style=3D"font-size:12pt">For =
each &quot;m=3D&quot; line in the offer, there MUST be a corresponding &quo=
t;m=3D&quot;</span></div>
<div><pre style=3D"word-wrap:break-word;white-space:pre-wrap">   line in th=
e answer.  The answer MUST contain exactly the same number
   of &quot;m=3D&quot; lines as the offer.  This allows for streams to be m=
atched up
   based on their order.  This implies that if the offer contained zero
   &quot;m=3D&quot; lines, the answer MUST contain zero &quot;m=3D&quot; li=
nes.</pre><pre style=3D"word-wrap:break-word;white-space:pre-wrap">Mamadou.=
</pre>  <div style=3D"font-family:&#39;times new roman&#39;,&#39;new york&#=
39;,times,serif;font-size:12pt">
 <div style=3D"font-family:&#39;times new roman&#39;,&#39;new york&#39;,tim=
es,serif;font-size:12pt"> <div dir=3D"ltr"> <hr size=3D"1">  <font face=3D"=
Arial"> <b><span style=3D"font-weight:bold">De=C2=A0:</span></b> I=C3=B1aki=
 Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@al=
iax.net</a>&gt;<br>
 <b><span style=3D"font-weight:bold">=C3=80=C2=A0:</span></b> &quot;<a href=
=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>&g=
t;; &quot;<a href=3D"mailto:public-webrtc@w3.org" target=3D"_blank">public-=
webrtc@w3.org</a>&quot; &lt;<a href=3D"mailto:public-webrtc@w3.org" target=
=3D"_blank">public-webrtc@w3.org</a>&gt; <br>
 <b><span style=3D"font-weight:bold">Envoy=C3=A9 le :</span></b> Lundi 29 j=
uillet 2013 19h31<br> <b><span style=3D"font-weight:bold">Objet=C2=A0:</spa=
n></b> [rtcweb] SDP is not suitable for WebRTC<br> </font> </div> <div><br>=
Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,<br>
but it was moved
 to MMUSIC maillist (don&#39;t know why since it is about<br>WebRTC applica=
tions design):<br><br><a href=3D"http://www.ietf.org/mail-archive/web/mmusi=
c/current/msg11966.html" target=3D"_blank">http://www.ietf.org/mail-archive=
/web/mmusic/current/msg11966.html</a><br>
<br>Sorry for the cross-posting but at this point I&#39;m a bit lost and do=
<br>not know which is the appropriate group for my concern.<br><br><br><br>=
So my concern is:<br><br><br>- Web application with a SIP over WebSocket cl=
ient running in the web.<br>
<br>- The web user is provided with a conference SIP URI in which there<br>=
are *already* 8 participants (5 of them emitting audio and video and 3<br>j=
ust emitting audio).<br><br>- The user calls, from his webphone, to the giv=
en URI to join the conference.<br>
<br><br><br>Let&#39;s imagine that the JS app knows the number of participa=
nt in the conference.<br>Let&#39;s imagine my browser have mic and webcam.<=
br><br><br><br>QUESTION:<br><br>How can my browser join the
 conference without requiring SDP<br>renegotiation from the server and, at =
the same time, being able to<br>send audio/video and receive audio/video fr=
om others (different tracks<br>/ m=3Dlines)?<br><br><br><br><br>&quot;SOLUT=
IONS&quot;:<br>
<br><br><br>1)<br><br>I tell my browser to generate a SDP offer with:<br><b=
r>=C2=A0 - 1 send/receive m=3Daudio line.<br>=C2=A0 - 7 recvonly m=3Daudio =
line.<br>=C2=A0 - 1 send/only m=3Dvideo line.<br>=C2=A0 - 4 recvonly m=3Dvi=
deo line.<br><br>(Obviously this is a joke)<br>
<br><br><br>2)<br><br>SDP seems to allow that the offer and the answer have=
 different number<br>of m lines (I&#39;m not aware of that but I believe th=
at SDP can do<br>&quot;everything&quot;). So my browser generates a SDP off=
er with 1 m=3Daudio line<br>
and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4<b=
r>m=3Dvideo lines.<br><br>Will my browser understand such a SDP answer with=
 more m lines than<br>its generated offer? I assume NOT.<br><br><br><br>3)<=
br>
<br>My browser
 generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo<br>line and th=
e server too. And later the server sends re-INVITE with all<br>the m lines.=
<br><br>Oppss, SDP renegotiation...<br><br><br><br><br>SDP is bad for WebRT=
C. SDP is good for legacy symmetric communications<br>
in which there is a single-track audio communication and, of course,<br>bot=
h endpoints emit audio. But SDP is bad for modern RTC protocols in<br>which=
 an endpoint can emit tons of tracks to a single endpoint.<br><br><br>Do we=
 really want this for WebRTC 1.0 ?<br>
<br><br>-- <br>I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.n=
et" target=3D"_blank">ibc@aliax.net</a>&gt;<br>____________________________=
___________________<br>rtcweb mailing list<br><a href=3D"mailto:rtcweb@ietf=
.org" target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br><br><br></div>
 </div> </div>  </div></div></div></div></blockquote></div>

--0015175df19ef975ed04e2aa1e7a--

From silviapfeiffer1@gmail.com  Mon Jul 29 15:53:12 2013
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Tue, 30 Jul 2013 08:52:51 +1000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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On Tue, Jul 30, 2013 at 3:31 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:
> Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
> but it was moved to MMUSIC maillist (don't know why since it is about
> WebRTC applications design):
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html
>
> Sorry for the cross-posting but at this point I'm a bit lost and do
> not know which is the appropriate group for my concern.
>
>
>
> So my concern is:
>
>
> - Web application with a SIP over WebSocket client running in the web.
>
> - The web user is provided with a conference SIP URI in which there
> are *already* 8 participants (5 of them emitting audio and video and 3
> just emitting audio).
>
> - The user calls, from his webphone, to the given URI to join the confere=
nce.
>
>
>
> Let's imagine that the JS app knows the number of participant in the conf=
erence.
> Let's imagine my browser have mic and webcam.
>
>
>
> QUESTION:
>
> How can my browser join the conference without requiring SDP
> renegotiation from the server and, at the same time, being able to
> send audio/video and receive audio/video from others (different tracks
> / m=3Dlines)?
>
>
>
>
> "SOLUTIONS":
>
>
>
> 1)
>
> I tell my browser to generate a SDP offer with:
>
>   - 1 send/receive m=3Daudio line.
>   - 7 recvonly m=3Daudio line.
>   - 1 send/only m=3Dvideo line.
>   - 4 recvonly m=3Dvideo line.
>
> (Obviously this is a joke)


Since you seem to be doing a full mesh and you seem to be wanting to
receive from everyone and be able to talk back to everyone, wouldn't
this need to be 8 different SDP offers in a full mesh network rather
than one SDP offer with 13 different m-lines? I don't follow how each
of the recipients would pick the right m-line for themselves...



> 2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines (I'm not aware of that but I believe that SDP can do
> "everything"). So my browser generates a SDP offer with 1 m=3Daudio line
> and 1 m=3Dvideo line, and the server replies with 8 m=3Daudio lines and 4
> m=3Dvideo lines.
>
> Will my browser understand such a SDP answer with more m lines than
> its generated offer? I assume NOT.

I'm don't understand SDP, but logic would tell me that it would send a
single SDP offer with its own capabilities to each of the other 8
participants:
m=3Daudio send/receive

(not sure if it also needs m=3Dvideo, since it's not a video capable device=
 IIUC)


> 3)
>
> My browser generates a SDP offer with 1 m=3Daudio line and 1 m=3Dvideo
> line and the server too. And later the server sends re-INVITE with all
> the m lines.
>
> Oppss, SDP renegotiation...

Yes, I would expect a negotiation to happen between each of the 8
current participants and the one new participant individually.
It would end up in 8 O/A negotiations that should all end up with 5
m=3Daudio send/receive and 3 m=3Daudio receive-only (from the POV of the
webphone.



I'm taking a stab at this because I want to understand more about how
it works, but I actually have no idea if I am correct. Please,
somebody with SDP knowlege, clarify how this would actually work. I,
too, am very curious.

Thanks,
Silvia.

From ron@debian.org  Mon Jul 29 16:15:05 2013
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From: Ron <ron@debian.org>
To: rtcweb@ietf.org
Message-ID: <20130729231436.GQ25981@audi.shelbyville.oz>
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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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On Mon, Jul 29, 2013 at 07:31:55PM +0200, Iñaki Baz Castillo wrote:
> Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
> but it was moved to MMUSIC maillist (don't know why since it is about
> WebRTC applications design):
> 
> http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html
> 
> Sorry for the cross-posting but at this point I'm a bit lost and do
> not know which is the appropriate group for my concern.

Have you tried a.r.k?



From christer.holmberg@ericsson.com  Mon Jul 29 20:52:54 2013
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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, "Bossiel thioriguel" <bossiel@yahoo.fr>
Thread-Topic: [rtcweb] SDP is not suitable for WebRTC
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Jul 2013 09:45:59 +0200
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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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2013/7/30 Silvia Pfeiffer <silviapfeiffer1@gmail.com>:
> Since you seem to be doing a full mesh and you seem to be wanting to
> receive from everyone and be able to talk back to everyone, wouldn't
> this need to be 8 different SDP offers in a full mesh network rather
> than one SDP offer with 13 different m-lines? I don't follow how each
> of the recipients would pick the right m-line for themselves...


Silvia, I don't feel I am doing a full mesh, but something really
common, which is:

- I contact, from my browser, to a conference server in which there is
an active conference with N participants.

- I provide my audio and video tracks to the conference server.

- The server provides me with N audio/video tracks belonging to N participa=
nts.

- I don't want to do a second SDP O/A round trip.

Please, note that my browser has a signaling and a media session with
the conference server. This is, the endpoints are my browser and the
conference server. The fact that the conference server acts as a mixer
by joining all the participants changes nothing. The browser
establishes a single RTC session with the server. This is not an
exotic scenario at all.

And no, I do not want to send 8 different SDP offers since I will
mantain just a single RTC session (with the conference server). All
the participants do the same, and the conference server is responsible
of taking all the tracks from participants and send them to all the
participants (via separate tracks).


I would like to hear the opinion of those in favour of SDP for WebRTC,
since what I am clearly stating is that SDP and Plan-Unified is *bad*
for this common scenario.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2013/7/30 Christer Holmberg <christer.holmberg@ericsson.com>:
> I was not wrong, but I was maybe unclear.

OK


> I meant that each entity would
> offer it=E2=80=99s m- lines as part of separate Offer/Answer transactions=
. It IS
> allowed to add m- lines in a subsequent Offer.

Yes, but that seems 100% unnecessary. The only we need is that peer-A
tells peer-B about its tracks and peer-B tells peer-A about its
tracks. We don't have that with SDP.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Jul 2013 10:01:10 +0200
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Subject: Re: [rtcweb] SDP is not suitable for WebRTC
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>> The only we need is that peer-A tells peer-B about its tracks and peer-B=
 tells peer-A about its tracks. We don't have that with SDP.
>
> Well, the peers have to use SOMETHING to do that.

Could you please explain it a bit more? If I am not wrong you insist
on the fact that, *theorically*, if a device knows (by other means)
the number of participants in a conference it can generate a SDP with
so many m lines (as in the "solution 1" I told about).

Then I recall: that is NOT possible in WebRTC. The API does not let me
to add 8 m=3Daudio lines (all of them with a=3Drecvonly but one with
a=3Dsendrec).

It is not valid for me that SDP allows "magic". I am interested in
which I can do with SDP *within WebRTC* via the "SDP-blob based API".

Thanks.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ted.ietf@gmail.com  Tue Jul 30 01:18:31 2013
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Subject: [rtcweb] Cross posting: Re:  SDP is not suitable for WebRTC
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--047d7b41407e6b484604e2b63ffa
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Howdy,

We have asked that folks avoid cross-posting, in order to keep it easier
for folks to follow a thread.  Please keep this discussion on public-webrtc=
.

thanks,

Ted Hardie

On Tue, Jul 30, 2013 at 9:45 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2013/7/30 Silvia Pfeiffer <silviapfeiffer1@gmail.com>:
> > Since you seem to be doing a full mesh and you seem to be wanting to
> > receive from everyone and be able to talk back to everyone, wouldn't
> > this need to be 8 different SDP offers in a full mesh network rather
> > than one SDP offer with 13 different m-lines? I don't follow how each
> > of the recipients would pick the right m-line for themselves...
>
>
> Silvia, I don't feel I am doing a full mesh, but something really
> common, which is:
>
> - I contact, from my browser, to a conference server in which there is
> an active conference with N participants.
>
> - I provide my audio and video tracks to the conference server.
>
> - The server provides me with N audio/video tracks belonging to N
> participants.
>
> - I don't want to do a second SDP O/A round trip.
>
> Please, note that my browser has a signaling and a media session with
> the conference server. This is, the endpoints are my browser and the
> conference server. The fact that the conference server acts as a mixer
> by joining all the participants changes nothing. The browser
> establishes a single RTC session with the server. This is not an
> exotic scenario at all.
>
> And no, I do not want to send 8 different SDP offers since I will
> mantain just a single RTC session (with the conference server). All
> the participants do the same, and the conference server is responsible
> of taking all the tracks from participants and send them to all the
> participants (via separate tracks).
>
>
> I would like to hear the opinion of those in favour of SDP for WebRTC,
> since what I am clearly stating is that SDP and Plan-Unified is *bad*
> for this common scenario.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b41407e6b484604e2b63ffa
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Howdy,<br><br>We have asked that folks avoid cross-posting, in order to kee=
p it easier for folks to follow a thread.=A0 Please keep this discussion on=
 public-webrtc.<br><br>thanks,<br><br>Ted Hardie<br><br><div class=3D"gmail=
_quote">
On Tue, Jul 30, 2013 at 9:45 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
2013/7/30 Silvia Pfeiffer &lt;<a href=3D"mailto:silviapfeiffer1@gmail.com">=
silviapfeiffer1@gmail.com</a>&gt;:<br>
<div class=3D"im">&gt; Since you seem to be doing a full mesh and you seem =
to be wanting to<br>
&gt; receive from everyone and be able to talk back to everyone, wouldn&#39=
;t<br>
&gt; this need to be 8 different SDP offers in a full mesh network rather<b=
r>
&gt; than one SDP offer with 13 different m-lines? I don&#39;t follow how e=
ach<br>
&gt; of the recipients would pick the right m-line for themselves...<br>
<br>
<br>
</div>Silvia, I don&#39;t feel I am doing a full mesh, but something really=
<br>
common, which is:<br>
<br>
- I contact, from my browser, to a conference server in which there is<br>
an active conference with N participants.<br>
<br>
- I provide my audio and video tracks to the conference server.<br>
<br>
- The server provides me with N audio/video tracks belonging to N participa=
nts.<br>
<br>
- I don&#39;t want to do a second SDP O/A round trip.<br>
<br>
Please, note that my browser has a signaling and a media session with<br>
the conference server. This is, the endpoints are my browser and the<br>
conference server. The fact that the conference server acts as a mixer<br>
by joining all the participants changes nothing. The browser<br>
establishes a single RTC session with the server. This is not an<br>
exotic scenario at all.<br>
<br>
And no, I do not want to send 8 different SDP offers since I will<br>
mantain just a single RTC session (with the conference server). All<br>
the participants do the same, and the conference server is responsible<br>
of taking all the tracks from participants and send them to all the<br>
participants (via separate tracks).<br>
<br>
<br>
I would like to hear the opinion of those in favour of SDP for WebRTC,<br>
since what I am clearly stating is that SDP and Plan-Unified is *bad*<br>
for this common scenario.<br>
<div class=3D"im HOEnZb"><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

--047d7b41407e6b484604e2b63ffa--

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Classification: UNCLASSIFIED
Caveats: NONE

I=C3=B1aki:

Yes, the conference setup scenario is itself is restricted in a kind of =
start-conference-arch, as if, it is the third party call setup in the =
SIP-conf-scenario in your conf-flow.

So, the discussions need to be confined based on this particular =
conf-call-flow arch. Once we agree on this, we can then limit our =
discussions with thiis particular case instead of making a general case.

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of I=C3=B1aki Baz Castillo
Sent: Tuesday, July 30, 2013 3:46 AM
To: Silvia Pfeiffer
Cc: rtcweb@ietf.org; public-webrtc@w3.org
Subject: Re: [rtcweb] SDP is not suitable for WebRTC

2013/7/30 Silvia Pfeiffer <silviapfeiffer1@gmail.com>:
> Since you seem to be doing a full mesh and you seem to be wanting to=20
> receive from everyone and be able to talk back to everyone, wouldn't=20
> this need to be 8 different SDP offers in a full mesh network rather=20
> than one SDP offer with 13 different m-lines? I don't follow how each=20
> of the recipients would pick the right m-line for themselves...


Silvia, I don't feel I am doing a full mesh, but something really =
common, which is:

- I contact, from my browser, to a conference server in which there is =
an active conference with N participants.

- I provide my audio and video tracks to the conference server.

- The server provides me with N audio/video tracks belonging to N =
participants.

- I don't want to do a second SDP O/A round trip.

Please, note that my browser has a signaling and a media session with =
the conference server. This is, the endpoints are my browser and the =
conference server. The fact that the conference server acts as a mixer =
by joining all the participants changes nothing. The browser establishes =
a single RTC session with the server. This is not an exotic scenario at =
all.

And no, I do not want to send 8 different SDP offers since I will =
mantain just a single RTC session (with the conference server). All the =
participants do the same, and the conference server is responsible of =
taking all the tracks from participants and send them to all the =
participants (via separate tracks).


I would like to hear the opinion of those in favour of SDP for WebRTC, =
since what I am clearly stating is that SDP and Plan-Unified is *bad* =
for this common scenario.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb



Classification: UNCLASSIFIED
Caveats: NONE


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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Date: Tue, 30 Jul 2013 15:11:16 +0200
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Subject: [rtcweb] Requesting a new use case / requirement for draft-ietf-rtcweb-use-cases-and-requirements
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Hi, I would like to request a new use case / requirement for WebRTC to
be included in draft-ietf-rtcweb-use-cases-and-requirements:





Contacting a conference server
---------------------------------------------

The browser must be able to generate a media offer to be sent to a
conference server, by providing its audio and video information (media
tracks to be sent by the browser), and the conference server must be
able to reply a single response accepting the offer and indicating the
browser all the media tracks from other participants it will send to
it. This must be possible without requiring a new media negotiation
round trip, and without requiring that the browser knows the number of
participants in the conference before contacting it.

Note that in this scenario the browser is the session initiator and
thus the media description offerer, while the conference server
provides the media description answer.




Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ted.ietf@gmail.com  Tue Jul 30 07:16:23 2013
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Date: Tue, 30 Jul 2013 16:16:12 +0200
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From: Ted Hardie <ted.ietf@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
Subject: Re: [rtcweb] Requesting a new use case / requirement for draft-ietf-rtcweb-use-cases-and-requirements
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--089e015370964d0af004e2bb40ae
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This seems to conflate what you want the service to do and how you believe
it ought to operate.  The core use case appears to be a variant of the
"Multiparty video communication" in section 4.2.10.1.  If I understand
correctly, this variant retains the browser behavior in respect to
rendering of peer media, but gets all the audio and video streams from a
single peer (the conferencing server).

This text:

"The conference server must be able to reply a single response accepting
the offer and indicating the browser all the media tracks from other
participants it will send to it. This must be possible without requiring a
new media negotiation round trip, and without requiring that the browser
knows the number of participants in the conference before contacting it."

is not appropriate for a use case.  It's a proposed constraint on the
solution, rather than a description of the use.

regards,

Ted Hardie


On Tue, Jul 30, 2013 at 3:11 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Hi, I would like to request a new use case / requirement for WebRTC to
> be included in draft-ietf-rtcweb-use-cases-and-requirements:
>
>
>
>
>
> Contacting a conference server
> ---------------------------------------------
>
> The browser must be able to generate a media offer to be sent to a
> conference server, by providing its audio and video information (media
> tracks to be sent by the browser), and the conference server must be
> able to reply a single response accepting the offer and indicating the
> browser all the media tracks from other participants it will send to
> it. This must be possible without requiring a new media negotiation
> round trip, and without requiring that the browser knows the number of
> participants in the conference before contacting it.
>
> Note that in this scenario the browser is the session initiator and
> thus the media description offerer, while the conference server
> provides the media description answer.
>
>
>
>
> Thanks a lot.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e015370964d0af004e2bb40ae
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This seems to conflate what you want the service to do and how you believe =
it ought to operate.=A0 The core use case appears to be a variant of the &q=
uot;Multiparty video communication<span class=3D"h4"></span>&quot; in secti=
on 4.2.10.1.=A0 If I understand correctly, this variant retains the browser=
 behavior in respect to rendering of peer media, but gets all the audio and=
 video streams from a single peer (the conferencing server).=A0 <br>
<br>This text:<br><br>&quot;The conference server must be able to reply a s=
ingle response accepting the offer and indicating the browser all the media=
 tracks from other participants it will send to it. This must be possible w=
ithout requiring a new media negotiation round trip, and without requiring =
that the browser knows the number of participants in the conference before =
contacting it.&quot;<br>
<br>is not appropriate for a use case.=A0 It&#39;s a proposed constraint on=
 the solution, rather than a description of the use.<br><br>regards,<br><br=
>Ted Hardie<br><br><br><div class=3D"gmail_quote">On Tue, Jul 30, 2013 at 3=
:11 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@al=
iax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi, I would like to request a new use case /=
 requirement for WebRTC to<br>
be included in draft-ietf-rtcweb-use-cases-and-requirements:<br>
<br>
<br>
<br>
<br>
<br>
Contacting a conference server<br>
---------------------------------------------<br>
<br>
The browser must be able to generate a media offer to be sent to a<br>
conference server, by providing its audio and video information (media<br>
tracks to be sent by the browser), and the conference server must be<br>
able to reply a single response accepting the offer and indicating the<br>
browser all the media tracks from other participants it will send to<br>
it. This must be possible without requiring a new media negotiation<br>
round trip, and without requiring that the browser knows the number of<br>
participants in the conference before contacting it.<br>
<br>
Note that in this scenario the browser is the session initiator and<br>
thus the media description offerer, while the conference server<br>
provides the media description answer.<br>
<br>
<br>
<br>
<br>
Thanks a lot.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br>

--089e015370964d0af004e2bb40ae--

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Date: Tue, 30 Jul 2013 17:05:50 +0200
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Subject: Re: [rtcweb] Requesting a new use case / requirement for draft-ietf-rtcweb-use-cases-and-requirements
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2013/7/30 Ted Hardie <ted.ietf@gmail.com>:
> "The conference server must be able to reply a single response accepting =
the
> offer and indicating the browser all the media tracks from other
> participants it will send to it. This must be possible without requiring =
a
> new media negotiation round trip, and without requiring that the browser
> knows the number of participants in the conference before contacting it."
>
> is not appropriate for a use case.  It's a proposed constraint on the
> solution, rather than a description of the use.

Hi Ted, I agree that the above is not an use case but a requirement.
But in contrast to what you say, I don't consider it a "constrain".
Instead I consider a constrain the current SDP O/A mandate which does
not let me implementing the above (the opposite is not true).

Should I request a new "requirement" for the draft instead of a new use cas=
e?


Best regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From partha@parthasarathi.co.in  Tue Jul 30 09:47:22 2013
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Date: Tue, 30 Jul 2013 22:13:57 +0530
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Subject: [rtcweb] Offer event handling for Pranswer state missing in JSEP
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Hi all,

In JSEP state machine, OFFER event is not possible to be handled at
Local-Pranswer, Remote-Pranswer state. OFFER event is required in
local-pranswer for multiple offer/answer handling like BUNDLE, ICE
negotiations. The example callflow for early-dialog with multiple
offer/answer handling is available at Sec 5.2 of
draft-partha-rtcweb-jsep-sip-00. It is open item in JSEP.

Thanks
Partha

 


From Ruediger.Geib@telekom.de  Wed Jul 31 01:12:01 2013
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From: <Ruediger.Geib@telekom.de>
To: <eckert@cisco.com>
Date: Wed, 31 Jul 2013 10:11:05 +0200
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

discussion seems to conclude with:

- fairness must be enforced if senders don't set DSCPs according to the glo=
bal
  fairness scheme.
- video applications have a hard time setting DSCPs anyway due to a missing=
 API.
- ECN is not meant to indicate drop priorities and offers an insufficient
  number of states.

It might be easier to use a single PHB/DSCP and add PCN like ECN. In an e2e=
 mode with receivers giving PCN congestion feedback to senders. My definiti=
on of PCN would be "rate control on a congested link" (rather than admissio=
n control and traffic termination as applied by the WG - I've been involved=
 in the WGs work).

This should result in:
- avoidance of sudden overload in that class (no burst drops)
- rate adaption of senders if congestion is pending
- a simpler design as compared to what you and Ken discussed

I'm going to drop another message soon.

Regards,

Ruediger

-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of T=
oerless Eckert (eckert)
Sent: Friday, July 12, 2013 3:00 AM
To: Dave Taht
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

On Wed, Jul 10, 2013 at 12:02:26PM -0700, Dave Taht wrote:
> I've begun to look at webrtc of late in the context of fq_codel.
>
> One thing that might work would be ECN marking important frames and
> not ECN marking others.

True. But i think there are at least 3 layers of priority that would
really be great to have: really important packet (long term reference
frames), "normal" packets, and low-proirity-packets (eg: discardable P).
And ECN could only do two.

Most of the time when you would want intelligent dropping, your drop
rate should be fairly low, eg< 20% instantaneous within the queue. If
you can do discardable P-frames it's of course best to drop those first.
Only when you instantaneously have really big bursts would you want to
drop "normal" packets and refrain to protection really important packets.

This does work really well with different queue-length (or WRED profiles).
If you only had two priorities (via ECN) it's hard to say how you
would best assign these two priorities to at least those three priority
levels.

> Problem overall is that ecn can be gamed and is largely not
> enabled, and so on...

Well. The gaming could be inhibited exactly by this draft because it
would try to guarantee the same amount of drops on a per-flow or "class"
basis, so if you're trying to game the system, you would be put into
your own class and would still see the same amount of drops as other flows.

Well, the main issue is really to redesignate ECN for a different
semantic than what it has today. And indicating packet priority is just
not quite the same as congestion feedback.

> In the case of the fq_codel algorithm I can see a "delayed drop"
> mechanism happening,
> where upon deciding to drop, it could defer a drop up to a few packets
> based on the classification
> of the packet, say 3 packets at most. Since flows are already
> identified, this is kind of easy
> by adding another state variable...

Yes, this is all quite interesting. We also played around with droppping
during enqueuing vs. during dequeuing, so there are a bunch of things
that could optimize intelligent dropping if you can build a more
intelligent queue.

[...]

>
> So I can see trying to utilize the AF markings as you describe in your
> draft in codel's
> drop decision. While this is also subject to being gamed, the
> disincentives are slight.

Lets separate the marking from the mechanism to inhibit gaming. This
draft really is about the latter, and the marking described with AF is
really just an example. Yes, it would be lovely to agree on markings
too.

> And what codel depends on is a definition of "TCP friendly". That if a
> stream is not going to behave in a tcp friendly manner, then an entirely
> different  strategy for such flows seems necessary, and glomming on some
> additional state isn't going to be the right thing.

Who stood up at RMCAT some time back and claimed to have worked for
> 10 years on TCP and declared TCP not self friendly to itself, so RMCAT
shouldn't bother ? ;-))

Kidding aside: Intelligent dropping IMHO makes perfect sense when being
applied to RT traffic sharing a queue friendly with TCP. And intelligent
dropping can then still improve the performance.

I am pretty positive though that RMCAT traffic in general will behave
better and get lower delay and more video-appropriate loss if it can have
its own queue. Primarily because it will just take decates to root out
all the bad TCP traffic.

I don't think that the strategies if you do or if you don't share the queue
with TCP would be fundamentally be different.

> a) how well can it work in routers to shift packet drops to low-prio pack=
ets
>
> I guess where I fall apart here is how to shift the encoders to send thin=
ner
> streams at  any level of packet drop.

Video codec coders are known to be able to do crazy cool things.
There are a bunch of known mechanisms and probably more proprietary ones.

> I'd be interested in specific codecs, example video streams, and test
> scripts to try and fiddle with this, as well as viable metrics.

Sure. Best done in person. If we're going to be at the WG for this, we can
talk on the side how we've been doing this.

Cheers
    Toerless

> > Thanks!
> >     Toerless
> >
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisc=
o.com>
> >
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >>
> >> We have submitted a new Internet draft to describe a Normalization Mar=
ker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal o=
f NM deployment is to create a new incentive for video encoders to generate=
 more discardable packets when using AF PHB in DIffServ. We are looking for=
ward to your review comments.
> >>
> >> Best regards,
> >> CJ
> >>
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (ecke=
rt); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.tx=
t
> >>
> >>
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >>
> >> Filename:      draft-lai-tsvwg-normalizer
> >> Revision:      00
> >> Title:                 Normalization Marker for AF PHB Group in DiffSe=
rv
> >> Creation date:         2013-02-08
> >> WG ID:                 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-n=
ormalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-norma=
lizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer=
-00
> >>
> >>
> >> Abstract:
> >>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>    nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>    useful for traffic conditioning with Active Queue Management (AQM)
> >>    when the AF PHB group is deployed for multimedia service classes th=
at
> >>    carry video packets with advanced codec technologies.  Using NM can
> >>    offer an incentive that is however missing in today's ecosystem of
> >>    practical deployment, i.e., when congestion occurs in the AF PHB
> >>    group, the network should allow a video codec to benefit from its
> >>    effort of generating finer layers of intra-flow precedence (IFP) wi=
th
> >>    discardable packets in a more balanced distribution.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
> >>
> >
> > --
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> >
>
>
>
> --
> Dave T=E4ht
>
> Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscrib=
e.html

--
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


From Ruediger.Geib@telekom.de  Wed Jul 31 01:28:44 2013
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From: <Ruediger.Geib@telekom.de>
To: <eckert@cisco.com>
Date: Wed, 31 Jul 2013 10:26:48 +0200
Thread-Topic: [tsvwg] Using intelligent dropping to improve video.....adding admission control?
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Toerless,

prior to setting up a new call, some kind of available bandwidth measuremen=
t or indication might be added. This isn't part of your draft. But an addit=
ional video stream may cause congestion on a bottleneck.

Measurement based admission control has been controversial so far. If used,=
 the main point is to use a measurement class sharing the same queue as the=
 traffic to be admitted, but having a higher drop probability.

- PCN is an option, but requires a new marking behaviour
 (threshold marker). End to end PCN as mentioned in another email.
- Using AF41 for admitted traffic and AF42 to perform measurements
  is an option to me, both should support ECN marking (AF42 seeing
  the higher marking and drop probability).

As marked packets indicating an issue can be interpreted by routers along a=
 path, while packets dropped earlier elsewhere can't be, I prefer marking t=
o dropping. I also think, it is easier to detect packet marks than delay va=
riation measurements in routers, if it is about detecting congestion.

Take the above as an example. I used a separate thread as this isn't part o=
f your draft an I accept if this shouldn't be discussed in the context of t=
his draft. It fit's well to the RTCWeb QoS discussion though, I think.

Regards,

Ruediger

From ted.ietf@gmail.com  Wed Jul 31 06:52:00 2013
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We still need scribes for the meeting tomorrow; please volunteer if you can
do that for us.

thanks,

Ted Hardie

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We still need scribes for the meeting tomorrow; please volunteer if you can=
 do that for us.=A0 <br><br>thanks,<br><br>Ted Hardie<br>

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On 7/31/13 3:51 PM, Ted Hardie wrote:
> We still need scribes for the meeting tomorrow; please volunteer if you
> can do that for us. 

Do you need minute takers, jabber relays, or both?

(And yes, we need to define these roles more clearly...)

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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On Wed, Jul 31, 2013 at 3:57 PM, Peter Saint-Andre <stpeter@stpeter.im>wrote:

> On 7/31/13 3:51 PM, Ted Hardie wrote:
> > We still need scribes for the meeting tomorrow; please volunteer if you
> > can do that for us.
>
> Do you need minute takers, jabber relays, or both?
>
> Most urgently minute takers (which are sometimes called "scribes").  It is
somewhat easier for the chairs to follow the jabber channel and, honestly,
easier to press someone into service for that role, since it requires relay
rather than writing.  If someone can actually scribe *into* the jabber
channel, that would be great, but we have meetecho as a fall back.

Ted



> (And yes, we need to define these roles more clearly...)
>
> Peter
>
> --
> Peter Saint-Andre
> https://stpeter.im/
>
>
>

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On Wed, Jul 31, 2013 at 3:57 PM, Peter Saint-Andre <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:stpeter@stpeter.im" target=3D"_blank">stpeter@stpeter.im</a=
>&gt;</span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">
<div class=3D"im">On 7/31/13 3:51 PM, Ted Hardie wrote:<br>
&gt; We still need scribes for the meeting tomorrow; please volunteer if yo=
u<br>
&gt; can do that for us.<br>
<br>
</div>Do you need minute takers, jabber relays, or both?<br>
<br></blockquote><div>Most urgently minute takers (which are sometimes call=
ed &quot;scribes&quot;).=A0 It is somewhat easier for the chairs to follow =
the jabber channel and, honestly, easier to press someone into service for =
that role, since it requires relay rather than writing.=A0 If someone can a=
ctually scribe *into* the jabber channel, that would be great, but we have =
meetecho as a fall back.<br>
<br>Ted<br><br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
(And yes, we need to define these roles more clearly...)<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Peter<br>
<br>
--<br>
Peter Saint-Andre<br>
<a href=3D"https://stpeter.im/" target=3D"_blank">https://stpeter.im/</a><b=
r>
<br>
<br>
</font></span></blockquote></div><br>

--20cf3036361f93483004e2cf2955--

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Adam Fineberg <fineberg@vline.me>
Thread-Topic: [avtext] [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.txt
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Subject: Re: [rtcweb] [avtext] Fwd: New Version Notification for	draft-fineberg-avtext-temporal-layer-ext-00.txt
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Hi Adam,

Why do you include PictureID? I can understand how the "Y" bit for base lay=
er sync can be useful to include. But how does including PictureID help ide=
ntify temporal layers?

Regards,
Mo

________________________________________
Date: Wed, 17 Jul 2013 17:09:46 -0700
From: fineberg@vline.me
To: rtcweb@ietf.org
Subject: [rtcweb] Fwd: New Version Notification for draft-fineberg-avtext-t=
emporal-layer-ext-00.txt

Hi,

I'm working on WebRTC services and have found that while developing service=
s that forward VP8 video streams if we want to take advantage of the VP8 te=
mporal scaling we must get the temporal layer information from the RTP head=
er which requires us to decrypt the SRTP packets. This is undesirable both =
because the middle-box needs to have access to the keys as well as the beca=
use of the added overhead of the decrypt/encrypt cycle. This draft proposes=
 an RTP header extension that will allow us to use the VP8 temporal layer i=
nformation included in the header extension and therefore do forwarding wit=
hout SRTP decryption. Comments welcome.

Regards,
Adam Fineberg
fineberg at vline.com

-------- Original Message --------=20
Subject:
New Version Notification for draft-fineberg-avtext-temporal-layer-ext-00.tx=
t
Date:
Tue, 09 Jul 2013 10:02:05 -0700
From:
internet-drafts at ietf.org
To:
Adam Fineberg=A0<fineberg at vline.com>



A new version of I-D, draft-fineberg-avtext-temporal-layer-ext-00.txt
has been successfully submitted by Adam Fineberg and posted to the
IETF repository.

Filename:	 draft-fineberg-avtext-temporal-layer-ext
Revision:	 00
Title:		 A Real-Time Transport Protocol (RTP) Header Extension for VP8 Temp=
oral Layer Information
Creation date:	 2013-07-08
Group:		 Individual Submission
Number of pages: 6
URL:             http://www.ietf.org/internet-drafts/draft-fineberg-avtext-=
temporal-layer-ext-00.txt
Status:          http://datatracker.ietf.org/doc/draft-fineberg-avtext-temp=
oral-layer-ext
Htmlized:        http://tools.ietf.org/html/draft-fineberg-avtext-temporal-=
layer-ext-00


Abstract:
   This document defines a mechanism by which packets of Real-Time
   Tranport Protocol (RTP) video streams encoded with the VP8 codec can
   indicate, in an RTP header extension, the temporal layer information
   about the frame encoded in the RTP packet.  This information can be
   used in a middlebox performing bandwidth management of streams
   without requiring it to decrypt the streams.

--=20
Regards,
Adam

