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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] Video codecs and the staw poll
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Date: Sun, 2 Feb 2014 22:02:17 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Video codecs and the staw poll
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+1
Silence is golden=20


On Jan 28, 2014, at 12:13 AM, Cullen Jennings <fluffy@iii.ca> wrote:

>=20
> Dear WG,
>=20
> After reviewing the poll results found here: =
http://www.ietf.org/mail-archive/web/rtcweb/current/pdfWd2PIhOY9y.pdf =
the chairs concludes that the working group still believes that an MTI =
is required for the WebRTC ecology to develop.    There are a number of =
options which did not garner significant support; essentially only =
options 1, 2, 3, 4 seem to have enough support that they might be the =
eventual basis of working group consensus.  The chairs do not view the =
other options as having sufficient support to warrant further working =
group activity or discussion.
>=20
> There is no obvious leader between VP8 and H.264, however, nor obvious =
support for selecting both.  Each has similar numbers of supporting =
positions and objections, and both have the support of well over half =
the participants in the straw poll.  Given that, we are no closer to =
being able to choose between them at this time. =20
>=20
> The chairs therefore propose tabling the discussion of a mandatory to =
implement video codec until about 6 week before the start of the IETF 91 =
meeting in November 2014. This is so that the working group can focus =
its energy on completing other work.  We do expect to begin work on the =
video document (draft-ietf-rtcweb-video) to meet its milestone of =
December, but initially without specifying which of the two codecs is =
the WG consensus for MTI.
>=20
> When we return to the discussion, the working group chairs currently =
expect to run a consensus call on support for each codec to be mandatory =
to implement.  This expectation may change, however, based on new =
information or working group experience.
>=20
> If anyone has concerns about tabling this discussion until September =
29, 2014 please let us know by February 4.
>=20
> Thank you,=20
>=20
> Cullen, Magnus, Ted <the chairs>
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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after-white-space;">+1<div>Silence is golden&nbsp;<br><div =
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</div>
<br><div><div>On Jan 28, 2014, at 12:13 AM, Cullen Jennings &lt;<a =
href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><br>Dear =
WG,<br><br>After reviewing the poll results found here: <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/pdfWd2PIhOY9y.=
pdf">http://www.ietf.org/mail-archive/web/rtcweb/current/pdfWd2PIhOY9y.pdf=
</a> the chairs concludes that the working group still believes that an =
MTI is required for the WebRTC ecology to develop. =
&nbsp;&nbsp;&nbsp;There are a number of options which did not garner =
significant support; essentially only options 1, 2, 3, 4 seem to have =
enough support that they might be the eventual basis of working group =
consensus. &nbsp;The chairs do not view the other options as having =
sufficient support to warrant further working group activity or =
discussion.<br><br>There is no obvious leader between VP8 and H.264, =
however, nor obvious support for selecting both. &nbsp;Each has similar =
numbers of supporting positions and objections, and both have the =
support of well over half the participants in the straw poll. =
&nbsp;Given that, we are no closer to being able to choose between them =
at this time. &nbsp;<br><br>The chairs therefore propose tabling the =
discussion of a mandatory to implement video codec until about 6 week =
before the start of the IETF 91 meeting in November 2014. This is so =
that the working group can focus its energy on completing other work. =
&nbsp;We do expect to begin work on the video document =
(draft-ietf-rtcweb-video) to meet its milestone of December, but =
initially without specifying which of the two codecs is the WG consensus =
for MTI.<br><br>When we return to the discussion, the working group =
chairs currently expect to run a consensus call on support for each =
codec to be mandatory to implement. &nbsp;This expectation may change, =
however, based on new information or working group experience.<br><br>If =
anyone has concerns about tabling this discussion until September 29, =
2014 please let us know by February 4.<br><br>Thank you, <br><br>Cullen, =
Magnus, Ted &lt;the =
chairs&gt;<br><br><br><br><br>____________________________________________=
___<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br></div></body></html>=

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Parthasarathi R <partha@parthasarathi.co.in>, "'Cb B'" <cb.list6@gmail.com>, "'Simon Perreault'" <simon.perreault@viagenie.ca>
Thread-Topic: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-and-requirements-12
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Subject: Re: [rtcweb] Query/Comment on	draft-ietf-rtcweb-use-cases-and-requirements-12
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> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus
> Westerlund
> Sent: Friday, January 31, 2014 2:32 PM
> To: Parthasarathi R; 'Cb B'; 'Simon Perreault'
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-and-
> requirements-12
>=20
> Hi Partha,
>=20
> Personal opinion:
>=20
> I think the below places the text in the wrong context. The note is in my=
 mind
> relevant in the context of the general NAT/FW traversal requirements, not
> the one discussing need to support multiple NAT/FW traversal servers. Thu=
s,
> I think Section 3.3.2 and thus requirement F29.
> Or potentially regarding Requirement F2. Is more appropriate places to
> include this.

F29, F2 does not mention any NAT/FW traversal techniques. F29 just discusse=
s the problems with NAT but IPv6 Firewalls could also be configured to bloc=
k UDP traffic. F19 Requirement seems to be missing in the latest version.

-Tiru.

>=20
> Cheers
>=20
> Magnus
>=20
> On 2014-01-31 02:26, Parthasarathi R wrote:
> > Hi Magnus,
> >
> > I can live with Simon text in case it is documented in Sec 4.2  as
> >
> >    F31     The browser must be able to use several STUN
> >            and TURN servers. Note that TURN support being mandatory
> >            does not preclude the browser from supporting
> >            additional traversal mechanisms.
> >    ----------------------------------------------------------------
> >    F32     There browser must support that STUN and TURN
> >            servers to use are supplied by other entities
> >            than via the web application (i.e. the network
> >            provider). Note that TURN support being mandatory
> >            does not preclude the browser from supporting
> >            additional traversal mechanisms.
> >
> > and also Appendix A:
> >
> > A22     The Web API must provide means for the
> >            application to specify several STUN and/or
> >            TURN servers to use. Note that TURN support being mandatory
> >            does not preclude a Web API from supporting
> >            additional traversal mechanisms.
> >
> > Please let me know in case you have any issue in the above text.
> >
> > BTW, just for the record,
> > draft-ietf-rtcweb-use-cases-and-requirements-12
> > does not specify the list of traversal mechanism requirements for
> > WebRTC Gateway/Server.
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Wednesday, January 29, 2014 2:59 PM
> >> To: Parthasarathi R; 'Cb B'; 'Simon Perreault'
> >> Cc: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Query/Comment on
> >> draft-ietf-rtcweb-use-cases-and-
> >> requirements-12
> >>
> >> Hi Partha and WG,
> >>
> >> I don't see any support for the changes you proposes in this
> >> discussion.
> >> What I see some support for is to add a statement making clear that
> >> there might be additional NAT/Firewall traversal mechanisms than
> >> STUN/TURN. Simon proposed:
> >>
> >> "Note that TURN support being mandatory does not preclude a WebRTC
> >> endpoint from supporting additional traversal mechanisms."
> >>
> >> However, looking at the document as it is currently written, I am
> >> uncertain where this would be added. The first mention of TURN is in
> >> Section 3.3.4.1, and that section is focused on the global service
> >> provider perspective and the need for location based provisioning of
> >> NAT/Firewall traversal server resources.
> >>
> >> I think it can be added to Section 3.3.5.1 without being misplaced,
> >> but it would be given a slightly narrower scope.
> >>
> >> I any of you want to be more explicit where this should be included,
> >> please be. If you are not forthcoming I will request the authors to
> >> add this in what they consider sensible position.
> >>
> >> Cheers
> >>
> >> Magnus
> >>
> >>
> >> On 2014-01-25 17:46, Parthasarathi R wrote:
> >>> Hi Simon,
> >>>
> >>>
> >>>
> >>> Thanks for your understanding about my firewall/NAT related problem
> >>> statement in this draft.
> >>>
> >>>
> >>>
> >>> I have proposed the firewall/NAT related text by which the specific
> >>> mechanism is not highlighted in the requirement document as there is
> >> no
> >>> WG consensus for any of the mechanism including TURN. It is possible
> >> to
> >>> argue hypothetically in PNTAW alias that PCP is the only mechanism
> >>> required in WebRTC endpoint.   Also, I'm more interested in WebRTC
> >>> gateway/server (Sec 4.3. of
> >>> draft-ietf-rtcweb-use-cases-and-requirements-12) requirements
> >>> wherein
> >> it
> >>> is not required to support TURN and the related mail thread is
> >>> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.
> >>>
> >>>
> >>>
> >>> IMO, my proposed text without mentioning any firewall/NAT mechanism
> >> in
> >>> the requirement document helps to move forward without depend on
> the
> >>> solution discussion in PNTAW alias.
> >>>
> >>>
> >>>
> >>> Thanks
> >>>
> >>> Partha
> >>>
> >>>
> >>>
> >>> *From:*Cb B [mailto:cb.list6@gmail.com]
> >>> *Sent:* Saturday, January 25, 2014 6:22 AM
> >>> *To:* Simon Perreault
> >>> *Cc:* rtcweb@ietf.org; Parthasarathi R
> >>> *Subject:* Re: [rtcweb] Query/Comment on
> >>> draft-ietf-rtcweb-use-cases-and-requirements-12
> >>>
> >>>
> >>>
> >>>
> >>> On Jan 24, 2014 10:17 AM, "Simon Perreault"
> >> <simon.perreault@viagenie.ca
> >>> <mailto:simon.perreault@viagenie.ca>> wrote:
> >>>>
> >>>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :
> >>>>
> >>>>> Please note that when non-IETFers read this requirement document,
> >>> they come
> >>>>> to the conclusion that IETF RTCWeb WG recommends TURN and not
> >>>>> other mechanisms. I'm saying that requirement document should not
> >>>>> be used
> >>> as the
> >>>>> mechanism to eliminate the other alternatives when there is a
> >> discussion
> >>>>> going-on in PNTAW alias. So, I'm asking for the change.
> >>>>
> >>>>
> >>>> I would totally agree with that sentiment, although I don't see
> >>>> your
> >>> proposed text change reflecting it accurately. How about simply:
> >>>>
> >>>> "Note that TURN support being mandatory does not preclude a
> WebRTC
> >>> endpoint from supporting additional traversal mechanisms."
> >>>>
> >>>>
> >>>
> >>> +1 for the above text.
> >>>
> >>> CB
> >>>
> >>>> Simon
> >>>> --
> >>>> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> >>>> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> >>>> STUN/TURN server               --> http://numb.viagenie.ca
> >>>> _______________________________________________
> >>>> 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
> >>>
> >>
> >>
> >> --
> >>
> >> Magnus Westerlund
> >>
> >> ---------------------------------------------------------------------
> >> - Services, Media and Network features, Ericsson Research EAB/TXM
> >> ----------------------------------------------------------------------
> >> Ericsson AB                 | Phone  +46 10 7148287
> >> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> >> SE-164 80 Stockholm, Sweden | mailto:
> magnus.westerlund@ericsson.com
> >> ---------------------------------------------------------------------
> >> -
> >
> >
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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While working on the latest JSEP update, I came across the following
problem:

When a answerer wants to reject a "media stream" in 3264, it sets the m=
line port in the answer to zero. However, when an endpoint wants to reject
a remote MediaStreamTrack without rejecting the m= line entirely, because
it is using the m= line for its own MediaStreamTrack, we don't have a clear
way to express that. While setting a=sendonly in the answer will tell the
other side to stop sending data, this won't kill the remote track. As far
as the remote side is concerned, its track is just 'paused'/'held', and is
still associated with the m= line.

Note that there is no such problem if an endpoint decides to kill its own
local MediaStreamTrack; in the offer it sends to indicate this, in addition
to setting the m= line to a=recvonly, it can also remove the MSID of the
track from the m= line, indicating the track is gone. In other words, the
a=msid attribute serves as the binding of MST to m= line, and its absence
indicates the track is gone.

Therefore, I think we need some similar way for a receiver to do the same
for a remote track. And fortunately, I think we have a surface that can be
used for this, namely the "recv-appId" attribute that has been discussed in
the context of unified-plan and explained in
http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt and
unified-plan. If an endpoint doesn't want to receive a particular track on
a m= line, it sets the m= line to a=sendonly and also removes any
a=recv-appId.

Below is an example of an audio call between A and B, with
MediaStreamTracks MSTA and MSTB. A initially generates an offer, and then I
show 4 different answer scenarios for various cases.

1) A generates offer with MSTA
m=audio X
a=msid:MSTA
a=recv-appId:1 # use 1 in the header extension when sending MSTA

2a) Normal answer: B adds its own track MSTB, generates answer:
m=audio Y
a=msid:MSTB
a=recv-appId:1

2b) Add+reject answer: B adds its own track MSTB, stops MSTA, generates
answer:
m=audio Y
a=msid:MSB
a=sendonly
# no appid; A can realize that MSA has been rejected

2c) Reject-only answer: B stops MSTA without adding a stream, generates
answer:
m=audio 0
# entire m-line has been rejected

2d) Do-nothing answer: B does nothing, generates answer:
m=audio Y
a=recv-appId:1
a=recvonly

If this seems reasonable, I plan to use this mechanism for handling m= line
allocation in JSEP.

Also, this essentially makes recv-appId the converse of MSID; for
simplicity, I think we should consider unifying these concepts into a
single document.

--001a11336a2e90057e04f1858368
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div style=3D"font-family:arial,sans-serif;font-size:13px"=
>While working on the latest JSEP update, I came across the following probl=
em:</div><div style=3D"font-family:arial,sans-serif;font-size:13px"><br></d=
iv>

<div style=3D"font-family:arial,sans-serif;font-size:13px">When a answerer =
wants to reject a &quot;media stream&quot; in 3264, it sets the m=3D line p=
ort in the answer to zero. However, when an endpoint wants to reject a remo=
te MediaStreamTrack without rejecting the m=3D line entirely, because it is=
 using the m=3D line for its own MediaStreamTrack, we don&#39;t have a clea=
r way to express that. While setting a=3Dsendonly in the answer will tell t=
he other side to stop sending data, this won&#39;t kill the remote track. A=
s far as the remote side is concerned, its track is just &#39;paused&#39;/&=
#39;held&#39;, and is still associated with the m=3D line.</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">Note that there is no =
such problem if an endpoint decides to kill its own local MediaStreamTrack;=
 in the offer it sends to indicate this, in addition to setting the m=3D li=
ne to a=3Drecvonly, it can also remove the MSID of the track from the m=3D =
line, indicating the track is gone. In other words, the a=3Dmsid attribute =
serves as the binding of MST to m=3D line, and its absence indicates the tr=
ack is gone.</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">Therefore, I think we =
need some similar way for a receiver to do the same for a remote track. And=
 fortunately, I think we have a surface that can be used for this, namely t=
he &quot;recv-appId&quot; attribute that has been discussed in the context =
of unified-plan and explained in <a href=3D"http://tools.ietf.org/id/draft-=
even-mmusic-application-token-02.txt">http://tools.ietf.org/id/draft-even-m=
music-application-token-02.txt</a> and unified-plan. If an endpoint doesn&#=
39;t want to receive a particular track on a m=3D line, it sets the m=3D li=
ne to a=3Dsendonly and also removes any a=3Drecv-appId.</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">Below is an example of=
 an audio call between A and B, with MediaStreamTracks MSTA and MSTB. A ini=
tially generates an offer, and then I show 4 different answer scenarios for=
 various cases.</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">1) A generates offer w=
ith MSTA</div><div style=3D"font-family:arial,sans-serif;font-size:13px">m=
=3Daudio X</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Dmsid:MSTA</d=
iv><div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecv-appI=
d:1 # use 1 in the header extension when sending MSTA</div><div style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">

<br></div><div style=3D"font-family:arial,sans-serif;font-size:13px">2a) No=
rmal answer: B adds its own track MSTB, generates answer:<br></div><div sty=
le=3D"font-family:arial,sans-serif;font-size:13px">m=3Daudio Y</div><div st=
yle=3D"font-family:arial,sans-serif;font-size:13px">

a=3Dmsid:MSTB</div><div style=3D"font-family:arial,sans-serif;font-size:13p=
x">a=3Drecv-appId:1</div><div style=3D"font-family:arial,sans-serif;font-si=
ze:13px"><br></div><div style=3D"font-family:arial,sans-serif;font-size:13p=
x">2b) Add+reject answer: B adds its own track MSTB, stops MSTA, generates =
answer:</div>

<div class=3D"im" style=3D"font-family:arial,sans-serif;font-size:13px"><di=
v>m=3Daudio Y</div><div>a=3Dmsid:MSB</div><div>a=3Dsendonly</div></div><div=
 style=3D"font-family:arial,sans-serif;font-size:13px"># no appid; A can re=
alize that MSA has been rejected</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">2c) Reject-only answer=
: B stops MSTA without adding a stream, generates answer:</div><div style=
=3D"font-family:arial,sans-serif;font-size:13px">

m=3Daudio 0=C2=A0</div><div style=3D"font-family:arial,sans-serif;font-size=
:13px"># entire m-line has been rejected</div><div style=3D"font-family:ari=
al,sans-serif;font-size:13px"><br></div><div style=3D"font-family:arial,san=
s-serif;font-size:13px">

2d) Do-nothing answer: B does nothing, generates answer:</div><div style=3D=
"font-family:arial,sans-serif;font-size:13px">m=3Daudio Y</div><div style=
=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecv-appId:1<br></div>=
<div style=3D"font-family:arial,sans-serif;font-size:13px">

a=3Drecvonly</div><div style=3D"font-family:arial,sans-serif;font-size:13px=
"><br></div><div style=3D"font-family:arial,sans-serif;font-size:13px">If t=
his seems reasonable, I plan to use this mechanism for handling m=3D line a=
llocation in JSEP.</div>

<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">Also, this essentially=
 makes recv-appId the converse of MSID; for simplicity, I think we should c=
onsider unifying these concepts into a single document.</div>

</div>

--001a11336a2e90057e04f1858368--

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From: Martin Thomson <martin.thomson@gmail.com>
To: Justin Uberti <juberti@google.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 3 February 2014 11:27, Justin Uberti <juberti@google.com> wrote:
> 2a) Normal answer: B adds its own track MSTB, generates answer:
> m=audio Y
> a=msid:MSTB
> a=recv-appId:1

Potential problems:

recv-appId looks like something that applies to what B is receiving
reject looks a little like does not support recv-appid as well

I think that the merging appid/msid is worth considering.  I think
that gets around these minor issues.

From christer.holmberg@ericsson.com  Mon Feb  3 12:08:39 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Cullen Jennings <fluffy@cisco.com>
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Hi,

Would this be solved by using uni-directional m- lines?

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Justin Uberti <juberti@google.com> wrote:



While working on the latest JSEP update, I came across the following proble=
m:

When a answerer wants to reject a "media stream" in 3264, it sets the m=3D =
line port in the answer to zero. However, when an endpoint wants to reject =
a remote MediaStreamTrack without rejecting the m=3D line entirely, because=
 it is using the m=3D line for its own MediaStreamTrack, we don't have a cl=
ear way to express that. While setting a=3Dsendonly in the answer will tell=
 the other side to stop sending data, this won't kill the remote track. As =
far as the remote side is concerned, its track is just 'paused'/'held', and=
 is still associated with the m=3D line.

Note that there is no such problem if an endpoint decides to kill its own l=
ocal MediaStreamTrack; in the offer it sends to indicate this, in addition =
to setting the m=3D line to a=3Drecvonly, it can also remove the MSID of th=
e track from the m=3D line, indicating the track is gone. In other words, t=
he a=3Dmsid attribute serves as the binding of MST to m=3D line, and its ab=
sence indicates the track is gone.

Therefore, I think we need some similar way for a receiver to do the same f=
or a remote track. And fortunately, I think we have a surface that can be u=
sed for this, namely the "recv-appId" attribute that has been discussed in =
the context of unified-plan and explained in http://tools.ietf.org/id/draft=
-even-mmusic-application-token-02.txt and unified-plan. If an endpoint does=
n't want to receive a particular track on a m=3D line, it sets the m=3D lin=
e to a=3Dsendonly and also removes any a=3Drecv-appId.

Below is an example of an audio call between A and B, with MediaStreamTrack=
s MSTA and MSTB. A initially generates an offer, and then I show 4 differen=
t answer scenarios for various cases.

1) A generates offer with MSTA
m=3Daudio X
a=3Dmsid:MSTA
a=3Drecv-appId:1 # use 1 in the header extension when sending MSTA

2a) Normal answer: B adds its own track MSTB, generates answer:
m=3Daudio Y
a=3Dmsid:MSTB
a=3Drecv-appId:1

2b) Add+reject answer: B adds its own track MSTB, stops MSTA, generates ans=
wer:
m=3Daudio Y
a=3Dmsid:MSB
a=3Dsendonly
# no appid; A can realize that MSA has been rejected

2c) Reject-only answer: B stops MSTA without adding a stream, generates ans=
wer:
m=3Daudio 0
# entire m-line has been rejected

2d) Do-nothing answer: B does nothing, generates answer:
m=3Daudio Y
a=3Drecv-appId:1
a=3Drecvonly

If this seems reasonable, I plan to use this mechanism for handling m=3D li=
ne allocation in JSEP.

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

--_000_37s61e9ckw02929t4v3gflwo1391458112947emailandroidcom_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body>
<pre style=3D"word-wrap:break-word; font-size:10.0pt; font-family:Tahoma; c=
olor:black">Hi,=0A=
=0A=
Would this be solved by using uni-directional m- lines?=0A=
=0A=
Regards,=0A=
=0A=
Christer=0A=
=0A=
Sent from my Sony Ericsson Xperia arc S=0A=
=0A=
Justin Uberti &lt;juberti@google.com&gt; wrote:=0A=
=0A=
</pre>
<div>
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif; font-size:13px">While working o=
n the latest JSEP update, I came across the following problem:</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">When a answerer=
 wants to reject a &quot;media stream&quot; in 3264, it sets the m=3D line =
port in the answer to zero. However, when an endpoint wants to reject a rem=
ote MediaStreamTrack without rejecting the m=3D
 line entirely, because it is using the m=3D line for its own MediaStreamTr=
ack, we don't have a clear way to express that. While setting a=3Dsendonly =
in the answer will tell the other side to stop sending data, this won't kil=
l the remote track. As far as the remote
 side is concerned, its track is just 'paused'/'held', and is still associa=
ted with the m=3D line.</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">Note that there=
 is no such problem if an endpoint decides to kill its own local MediaStrea=
mTrack; in the offer it sends to indicate this, in addition to setting the =
m=3D line to a=3Drecvonly, it can also
 remove the MSID of the track from the m=3D line, indicating the track is g=
one. In other words, the a=3Dmsid attribute serves as the binding of MST to=
 m=3D line, and its absence indicates the track is gone.</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">Therefore, I th=
ink we need some similar way for a receiver to do the same for a remote tra=
ck. And fortunately, I think we have a surface that can be used for this, n=
amely the &quot;recv-appId&quot; attribute that
 has been discussed in the context of unified-plan and explained in <a href=
=3D"http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt">
http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a> and=
 unified-plan. If an endpoint doesn't want to receive a particular track on=
 a m=3D line, it sets the m=3D line to a=3Dsendonly and also removes any a=
=3Drecv-appId.</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">Below is an exa=
mple of an audio call between A and B, with MediaStreamTracks MSTA and MSTB=
. A initially generates an offer, and then I show 4 different answer scenar=
ios for various cases.</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">1) A generates =
offer with MSTA</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">m=3Daudio X</di=
v>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Dmsid:MSTA</=
div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Drecv-appId:=
1 # use 1 in the header extension when sending MSTA</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">2a) Normal answ=
er: B adds its own track MSTB, generates answer:<br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">m=3Daudio Y</di=
v>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Dmsid:MSTB</=
div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Drecv-appId:=
1</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">2b) Add&#43;rej=
ect answer: B adds its own track MSTB, stops MSTA, generates answer:</div>
<div class=3D"im" style=3D"font-family:arial,sans-serif; font-size:13px">
<div>m=3Daudio Y</div>
<div>a=3Dmsid:MSB</div>
<div>a=3Dsendonly</div>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"># no appid; A c=
an realize that MSA has been rejected</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">2c) Reject-only=
 answer: B stops MSTA without adding a stream, generates answer:</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">m=3Daudio 0&nbs=
p;</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"># entire m-line=
 has been rejected</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">2d) Do-nothing =
answer: B does nothing, generates answer:</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">m=3Daudio Y</di=
v>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Drecv-appId:=
1<br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">a=3Drecvonly</d=
iv>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">If this seems r=
easonable, I plan to use this mechanism for handling m=3D line allocation i=
n JSEP.</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif; font-size:13px">Also, this esse=
ntially makes recv-appId the converse of MSID; for simplicity, I think we s=
hould consider unifying these concepts into a single document.</div>
</div>
</div>
</body>
</html>

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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 13:27, Justin Uberti wrote:
> Therefore, I think we need some similar way for a receiver to do the 
> same for a remote track.

Yes.

> And fortunately, I think we have a surface that can be used for this, 
> namely the "recv-appId" attribute that has been discussed in the 
> context of unified-plan and explained in 
> http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt 
> and unified-plan. If an endpoint doesn't want to receive a particular 
> track on a m= line, it sets the m= line to a=sendonly and also removes 
> any a=recv-appId.

No.

The problem is that you're defining semantics to a behavior that is 
indistinguishable from the behavior exhibited by something that does not 
support app-id: if the thing on the other end isn't using app-id, then 
an attempt to put a stream on hold will be misinterpreted as an attempt 
to reject or remove it.

I agree that we want an SDP operation that differentiates between "don't 
send me this now" and "don't send me this ever" [1]  -- but it needs to 
be explicit. Something more like "a=i-am-rejecting-this-stream", but 
with a less cumbersome name.

/a


____
[1] Along with a Javascript control that allows us to do the same thing, 
but that's a W3C issue.

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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 2/3/14 13:27, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-22T7hLMdC2je0nLk34MXQ8L+JFWLtAz--6Ryt+DMaMvQ@mail.gmail.com"
      type="cite">
      <div style="font-family:arial,sans-serif;font-size:13px">Therefore,
        I think we need some similar way for a receiver to do the same
        for a remote track.</div>
    </blockquote>
    <br>
    Yes.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22T7hLMdC2je0nLk34MXQ8L+JFWLtAz--6Ryt+DMaMvQ@mail.gmail.com"
      type="cite">
      <div style="font-family:arial,sans-serif;font-size:13px">And
        fortunately, I think we have a surface that can be used for
        this, namely the "recv-appId" attribute that has been discussed
        in the context of unified-plan and explained in <a
          moz-do-not-send="true"
href="http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt">http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a>
        and unified-plan. If an endpoint doesn't want to receive a
        particular track on a m= line, it sets the m= line to a=sendonly
        and also removes any a=recv-appId.</div>
    </blockquote>
    <br>
    No.<br>
    <br>
    The problem is that you're defining semantics to a behavior that is
    indistinguishable from the behavior exhibited by something that does
    not support app-id: if the thing on the other end isn't using
    app-id, then an attempt to put a stream on hold will be
    misinterpreted as an attempt to reject or remove it.<br>
    <br>
    I agree that we want an SDP operation that differentiates between
    "don't send me this now" and "don't send me this ever" [1]Â  -- but
    it needs to be explicit. Something more like
    "a=i-am-rejecting-this-stream", but with a less cumbersome name.<br>
    <br>
    /a<br>
    <br>
    <br>
    ____<br>
    [1] Along with a Javascript control that allows us to do the same
    thing, but that's a W3C issue.<br>
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 12:06 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 3 February 2014 11:27, Justin Uberti <juberti@google.com> wrote:
> > 2a) Normal answer: B adds its own track MSTB, generates answer:
> > m=audio Y
> > a=msid:MSTB
> > a=recv-appId:1
>
> Potential problems:
>
> recv-appId looks like something that applies to what B is receiving
>

It does, in fact. In B's answer, a=msid indicates what B is sending;
a=recv-appId indicates that B wants to receive the MST from A that is
associated with this m-line.


> reject looks a little like does not support recv-appid as well
>

True. But if, as you say, we merge appid and msid, then the only possible
ambiguity comes from the legacy no-msid, no-appid case, and even that is
not an issue since all the possible outcomes are separable: full duplex
(m=audio Y), one-way reject (m=audio Y, a=sendonly), full reject (m=audio
0). Of course, we can't know that the one-way reject is final, but the
number of add/remove stream operations done by legacy devices should be
limited.


> I think that the merging appid/msid is worth considering.  I think
> that gets around these minor issues.

--001a1136a478cb09ac04f1863ff7
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 12:06 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"><div class=3D"im">On 3 February 2014 11:27, Justin Uberti =
&lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:=
<br>


&gt; 2a) Normal answer: B adds its own track MSTB, generates answer:<br>
&gt; m=3Daudio Y<br>
&gt; a=3Dmsid:MSTB<br>
&gt; a=3Drecv-appId:1<br>
<br>
</div>Potential problems:<br>
<br>
recv-appId looks like something that applies to what B is receiving<br></bl=
ockquote><div><br></div><div>It does, in fact. In B&#39;s answer, a=3Dmsid =
indicates what B is sending; a=3Drecv-appId indicates that B wants to recei=
ve the MST from A that is associated with this m-line.</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">
reject looks a little like does not support recv-appid as well<br></blockqu=
ote><div><br></div><div>True. But if, as you say, we merge appid and msid, =
then the only possible ambiguity comes from the legacy no-msid, no-appid ca=
se, and even that is not an issue since all the possible outcomes are separ=
able: full duplex (m=3Daudio Y), one-way reject (m=3Daudio Y, a=3Dsendonly)=
, full reject (m=3Daudio 0). Of course, we can&#39;t know that the one-way =
reject is final, but the number of add/remove stream operations done by leg=
acy devices should be limited.</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">
<br>
I think that the merging appid/msid is worth considering. =C2=A0I think<br>
that gets around these minor issues.=C2=A0</blockquote></div></div></div>

--001a1136a478cb09ac04f1863ff7--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 12:20:53 -0800
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 12:08 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
> Would this be solved by using uni-directional m- lines?
>
> Yes, but at the cost of compatibility with legacy. There has been broad
support for bidirectional m= lines each time this has come up.

>
> Regards,
>
> Christer
>
> Sent from my Sony Ericsson Xperia arc S
>
> Justin Uberti <juberti@google.com> wrote:
>
>
>  While working on the latest JSEP update, I came across the following
> problem:
>
>  When a answerer wants to reject a "media stream" in 3264, it sets the m=
> line port in the answer to zero. However, when an endpoint wants to reject
> a remote MediaStreamTrack without rejecting the m= line entirely, because
> it is using the m= line for its own MediaStreamTrack, we don't have a clear
> way to express that. While setting a=sendonly in the answer will tell the
> other side to stop sending data, this won't kill the remote track. As far
> as the remote side is concerned, its track is just 'paused'/'held', and is
> still associated with the m= line.
>
>  Note that there is no such problem if an endpoint decides to kill its
> own local MediaStreamTrack; in the offer it sends to indicate this, in
> addition to setting the m= line to a=recvonly, it can also remove the MSID
> of the track from the m= line, indicating the track is gone. In other
> words, the a=msid attribute serves as the binding of MST to m= line, and
> its absence indicates the track is gone.
>
>  Therefore, I think we need some similar way for a receiver to do the
> same for a remote track. And fortunately, I think we have a surface that
> can be used for this, namely the "recv-appId" attribute that has been
> discussed in the context of unified-plan and explained in
> http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt and
> unified-plan. If an endpoint doesn't want to receive a particular track on
> a m= line, it sets the m= line to a=sendonly and also removes any
> a=recv-appId.
>
>  Below is an example of an audio call between A and B, with
> MediaStreamTracks MSTA and MSTB. A initially generates an offer, and then I
> show 4 different answer scenarios for various cases.
>
>  1) A generates offer with MSTA
> m=audio X
> a=msid:MSTA
> a=recv-appId:1 # use 1 in the header extension when sending MSTA
>
>  2a) Normal answer: B adds its own track MSTB, generates answer:
>  m=audio Y
> a=msid:MSTB
> a=recv-appId:1
>
>  2b) Add+reject answer: B adds its own track MSTB, stops MSTA, generates
> answer:
>  m=audio Y
> a=msid:MSB
> a=sendonly
>  # no appid; A can realize that MSA has been rejected
>
>  2c) Reject-only answer: B stops MSTA without adding a stream, generates
> answer:
> m=audio 0
> # entire m-line has been rejected
>
>  2d) Do-nothing answer: B does nothing, generates answer:
> m=audio Y
> a=recv-appId:1
>  a=recvonly
>
>  If this seems reasonable, I plan to use this mechanism for handling m=
> line allocation in JSEP.
>
>  Also, this essentially makes recv-appId the converse of MSID; for
> simplicity, I think we should consider unifying these concepts into a
> single document.
>

--047d7b873810e0cb5004f186434c
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, Feb 3, 2014 at 12:08 PM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@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>
<pre style=3D"font-size:10.0pt;font-family:Tahoma;word-wrap:break-word">Hi,

Would this be solved by using uni-directional m- lines?</pre></div></blockq=
uote><div>Yes, but at the cost of compatibility with legacy. There has been=
 broad support for bidirectional m=3D lines each time this has come up.<br>

</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div><pre style=3D"font-size:10.0pt;fo=
nt-family:Tahoma;word-wrap:break-word">
Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Justin Uberti &lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">j=
uberti@google.com</a>&gt; wrote:

</pre><div><div class=3D"h5">
<div>
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">While working on=
 the latest JSEP update, I came across the following problem:</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">When a answerer =
wants to reject a &quot;media stream&quot; in 3264, it sets the m=3D line p=
ort in the answer to zero. However, when an endpoint wants to reject a remo=
te MediaStreamTrack without rejecting the m=3D
 line entirely, because it is using the m=3D line for its own MediaStreamTr=
ack, we don&#39;t have a clear way to express that. While setting a=3Dsendo=
nly in the answer will tell the other side to stop sending data, this won&#=
39;t kill the remote track. As far as the remote
 side is concerned, its track is just &#39;paused&#39;/&#39;held&#39;, and =
is still associated with the m=3D line.</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">Note that there =
is no such problem if an endpoint decides to kill its own local MediaStream=
Track; in the offer it sends to indicate this, in addition to setting the m=
=3D line to a=3Drecvonly, it can also
 remove the MSID of the track from the m=3D line, indicating the track is g=
one. In other words, the a=3Dmsid attribute serves as the binding of MST to=
 m=3D line, and its absence indicates the track is gone.</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">Therefore, I thi=
nk we need some similar way for a receiver to do the same for a remote trac=
k. And fortunately, I think we have a surface that can be used for this, na=
mely the &quot;recv-appId&quot; attribute that
 has been discussed in the context of unified-plan and explained in <a href=
=3D"http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt" ta=
rget=3D"_blank">
http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a> and=
 unified-plan. If an endpoint doesn&#39;t want to receive a particular trac=
k on a m=3D line, it sets the m=3D line to a=3Dsendonly and also removes an=
y a=3Drecv-appId.</div>


<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">Below is an exam=
ple of an audio call between A and B, with MediaStreamTracks MSTA and MSTB.=
 A initially generates an offer, and then I show 4 different answer scenari=
os for various cases.</div>


<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">1) A generates o=
ffer with MSTA</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">m=3Daudio X</div=
>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Dmsid:MSTA</d=
iv>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecv-appId:1=
 # use 1 in the header extension when sending MSTA</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">2a) Normal answe=
r: B adds its own track MSTB, generates answer:<br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">m=3Daudio Y</div=
>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Dmsid:MSTB</d=
iv>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecv-appId:1=
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">2b) Add+reject a=
nswer: B adds its own track MSTB, stops MSTA, generates answer:</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">
<div>m=3Daudio Y</div>
<div>a=3Dmsid:MSB</div>
<div>a=3Dsendonly</div>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"># no appid; A ca=
n realize that MSA has been rejected</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">2c) Reject-only =
answer: B stops MSTA without adding a stream, generates answer:</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">m=3Daudio 0=C2=
=A0</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"># entire m-line =
has been rejected</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">2d) Do-nothing a=
nswer: B does nothing, generates answer:</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">m=3Daudio Y</div=
>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecv-appId:1=
<br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">a=3Drecvonly</di=
v>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">If this seems re=
asonable, I plan to use this mechanism for handling m=3D line allocation in=
 JSEP.</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</div>
</div></div></div>

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

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On Mon, Feb 3, 2014 at 12:19 PM, Adam Roach <adam@nostrum.com> wrote:

>  On 2/3/14 13:27, Justin Uberti wrote:
>
> Therefore, I think we need some similar way for a receiver to do the same
> for a remote track.
>
>
> Yes.
>
>
>  And fortunately, I think we have a surface that can be used for this,
> namely the "recv-appId" attribute that has been discussed in the context of
> unified-plan and explained in
> http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt and
> unified-plan. If an endpoint doesn't want to receive a particular track on
> a m= line, it sets the m= line to a=sendonly and also removes any
> a=recv-appId.
>
>
> No.
>
> The problem is that you're defining semantics to a behavior that is
> indistinguishable from the behavior exhibited by something that does not
> support app-id: if the thing on the other end isn't using app-id, then an
> attempt to put a stream on hold will be misinterpreted as an attempt to
> reject or remove it.
>

See my response to Martin. Putting a stream on hold will never result in
that being misinterpreted as a reject; on the contrary, without appid,
there is no way to 'fully' reject half of a stream, and, as a problem
confined to pre-unified-plan devices, I think that is probably OK.

>
> I agree that we want an SDP operation that differentiates between "don't
> send me this now" and "don't send me this ever" [1]  -- but it needs to be
> explicit. Something more like "a=i-am-rejecting-this-stream", but with a
> less cumbersome name.
>
> It's not clear to me that adding a new attribute would solve any problem
that appid doesn't solve; you still need to deal with the "other end not
using it" issue.


> ____
> [1] Along with a Javascript control that allows us to do the same thing,
> but that's a W3C issue.
>

Agree. But while we're on the subject, I am thinking that MST.stop() ==
"not ever", and RtpReceiver.active = false causes "not now".

--047d7b33d85620aa3104f1865571
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, Feb 3, 2014 at 12:19 PM, 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:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im">
    <div>On 2/3/14 13:27, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">Therefore,
        I think we need some similar way for a receiver to do the same
        for a remote track.</div>
    </blockquote>
    <br></div>
    Yes.<div class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">And
        fortunately, I think we have a surface that can be used for
        this, namely the &quot;recv-appId&quot; attribute that has been dis=
cussed
        in the context of unified-plan and explained in <a href=3D"http://t=
ools.ietf.org/id/draft-even-mmusic-application-token-02.txt" target=3D"_bla=
nk">http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a>
        and unified-plan. If an endpoint doesn&#39;t want to receive a
        particular track on a m=3D line, it sets the m=3D line to a=3Dsendo=
nly
        and also removes any a=3Drecv-appId.</div>
    </blockquote>
    <br></div>
    No.<br>
    <br>
    The problem is that you&#39;re defining semantics to a behavior that is
    indistinguishable from the behavior exhibited by something that does
    not support app-id: if the thing on the other end isn&#39;t using
    app-id, then an attempt to put a stream on hold will be
    misinterpreted as an attempt to reject or remove it.<br></div></blockqu=
ote><div><br></div><div>See my response to Martin. Putting a stream on hold=
 will never result in that being misinterpreted as a reject; on the contrar=
y, without appid, there is no way to &#39;fully&#39; reject half of a strea=
m, and, as a problem confined to pre-unified-plan devices, I think that is =
probably OK.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <br>
    I agree that we want an SDP operation that differentiates between
    &quot;don&#39;t send me this now&quot; and &quot;don&#39;t send me this=
 ever&quot; [1]=C2=A0 -- but
    it needs to be explicit. Something more like
    &quot;a=3Di-am-rejecting-this-stream&quot;, but with a less cumbersome =
name.<br><br></div></blockquote><div>It&#39;s not clear to me that adding a=
 new attribute would solve any problem that appid doesn&#39;t solve; you st=
ill need to deal with the &quot;other end not using it&quot; issue.</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"><div bgcolor=3D"#FFFFFF" te=
xt=3D"#000000">
    ____<br>
    [1] Along with a Javascript control that allows us to do the same
    thing, but that&#39;s a W3C issue.<br></div></blockquote><div><br></div=
><div>Agree. But while we&#39;re on the subject, I am thinking that MST.sto=
p() =3D=3D &quot;not ever&quot;, and RtpReceiver.active =3D false causes &q=
uot;not now&quot;.=C2=A0</div>

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

--047d7b33d85620aa3104f1865571--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 12:28:03 -0800
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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--001a11c1e7fc8718b604f1865d89
Content-Type: text/plain; charset=UTF-8

+mmusic (meant to include the first time around)


On Mon, Feb 3, 2014 at 12:25 PM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Mon, Feb 3, 2014 at 12:19 PM, Adam Roach <adam@nostrum.com> wrote:
>
>>  On 2/3/14 13:27, Justin Uberti wrote:
>>
>> Therefore, I think we need some similar way for a receiver to do the same
>> for a remote track.
>>
>>
>> Yes.
>>
>>
>>  And fortunately, I think we have a surface that can be used for this,
>> namely the "recv-appId" attribute that has been discussed in the context of
>> unified-plan and explained in
>> http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt and
>> unified-plan. If an endpoint doesn't want to receive a particular track on
>> a m= line, it sets the m= line to a=sendonly and also removes any
>> a=recv-appId.
>>
>>
>> No.
>>
>> The problem is that you're defining semantics to a behavior that is
>> indistinguishable from the behavior exhibited by something that does not
>> support app-id: if the thing on the other end isn't using app-id, then an
>> attempt to put a stream on hold will be misinterpreted as an attempt to
>> reject or remove it.
>>
>
> See my response to Martin. Putting a stream on hold will never result in
> that being misinterpreted as a reject; on the contrary, without appid,
> there is no way to 'fully' reject half of a stream, and, as a problem
> confined to pre-unified-plan devices, I think that is probably OK.
>
>>
>> I agree that we want an SDP operation that differentiates between "don't
>> send me this now" and "don't send me this ever" [1]  -- but it needs to be
>> explicit. Something more like "a=i-am-rejecting-this-stream", but with a
>> less cumbersome name.
>>
>> It's not clear to me that adding a new attribute would solve any problem
> that appid doesn't solve; you still need to deal with the "other end not
> using it" issue.
>
>
>> ____
>> [1] Along with a Javascript control that allows us to do the same thing,
>> but that's a W3C issue.
>>
>
> Agree. But while we're on the subject, I am thinking that MST.stop() ==
> "not ever", and RtpReceiver.active = false causes "not now".
>
>

--001a11c1e7fc8718b604f1865d89
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">+mmusic (meant to include the first time around)</div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Feb 3, 20=
14 at 12:25 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juber=
ti@google.com" target=3D"_blank">juberti@google.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 dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"im">On Mon, Feb 3, 201=
4 at 12:19 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nost=
rum.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>On 2/3/14 13:27, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">Therefore,
        I think we need some similar way for a receiver to do the same
        for a remote track.</div>
    </blockquote>
    <br></div>
    Yes.<div><br>
    <br>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">And
        fortunately, I think we have a surface that can be used for
        this, namely the &quot;recv-appId&quot; attribute that has been dis=
cussed
        in the context of unified-plan and explained in <a href=3D"http://t=
ools.ietf.org/id/draft-even-mmusic-application-token-02.txt" target=3D"_bla=
nk">http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a>
        and unified-plan. If an endpoint doesn&#39;t want to receive a
        particular track on a m=3D line, it sets the m=3D line to a=3Dsendo=
nly
        and also removes any a=3Drecv-appId.</div>
    </blockquote>
    <br></div>
    No.<br>
    <br>
    The problem is that you&#39;re defining semantics to a behavior that is
    indistinguishable from the behavior exhibited by something that does
    not support app-id: if the thing on the other end isn&#39;t using
    app-id, then an attempt to put a stream on hold will be
    misinterpreted as an attempt to reject or remove it.<br></div></blockqu=
ote><div><br></div></div><div>See my response to Martin. Putting a stream o=
n hold will never result in that being misinterpreted as a reject; on the c=
ontrary, without appid, there is no way to &#39;fully&#39; reject half of a=
 stream, and, as a problem confined to pre-unified-plan devices, I think th=
at is probably OK.=C2=A0</div>

<div class=3D"im">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <br>
    I agree that we want an SDP operation that differentiates between
    &quot;don&#39;t send me this now&quot; and &quot;don&#39;t send me this=
 ever&quot; [1]=C2=A0 -- but
    it needs to be explicit. Something more like
    &quot;a=3Di-am-rejecting-this-stream&quot;, but with a less cumbersome =
name.<br><br></div></blockquote></div><div>It&#39;s not clear to me that ad=
ding a new attribute would solve any problem that appid doesn&#39;t solve; =
you still need to deal with the &quot;other end not using it&quot; issue.</=
div>

<div class=3D"im">
<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"><div bgcolor=3D"#FFFFFF" te=
xt=3D"#000000">
    ____<br>
    [1] Along with a Javascript control that allows us to do the same
    thing, but that&#39;s a W3C issue.<br></div></blockquote><div><br></div=
></div><div>Agree. But while we&#39;re on the subject, I am thinking that M=
ST.stop() =3D=3D &quot;not ever&quot;, and RtpReceiver.active =3D false cau=
ses &quot;not now&quot;.=C2=A0</div>


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

--001a11c1e7fc8718b604f1865d89--

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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 14:25, Justin Uberti wrote:
> It's not clear to me that adding a new attribute would solve any 
> problem that appid doesn't solve; you still need to deal with the 
> "other end not using it" issue.

Regardless of whether you think there's a problem in the legacy interop 
case, the big benefit here is that an explicit indicator by definition 
makes the operation explicit, and that in-and-of-itself is a good thing.

The core issue is that you're trying to make the operation implicit; 
your proposal edges awfully close to the DWIM model of design, which we 
know causes problems when we try to add further extensions.

/a

From juberti@google.com  Mon Feb  3 13:03:44 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 13:03:22 -0800
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com> wrote:

> On 2/3/14 14:25, Justin Uberti wrote:
>
>> It's not clear to me that adding a new attribute would solve any problem
>> that appid doesn't solve; you still need to deal with the "other end not
>> using it" issue.
>>
>
> Regardless of whether you think there's a problem in the legacy interop
> case, the big benefit here is that an explicit indicator by definition
> makes the operation explicit, and that in-and-of-itself is a good thing.
>
> The core issue is that you're trying to make the operation implicit; your
> proposal edges awfully close to the DWIM model of design, which we know
> causes problems when we try to add further extensions.


Good point, but since we are still defining exactly what appId does, I
think we have the opportunity to make it explicit.

--047d7b343296d8404004f186db09
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 12:42 PM, 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:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 2/3/14 14:25, 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">
It&#39;s not clear to me that adding a new attribute would solve any proble=
m that appid doesn&#39;t solve; you still need to deal with the &quot;other=
 end not using it&quot; issue.<br>
</blockquote>
<br></div>
Regardless of whether you think there&#39;s a problem in the legacy interop=
 case, the big benefit here is that an explicit indicator by definition mak=
es the operation explicit, and that in-and-of-itself is a good thing.<br>


<br>
The core issue is that you&#39;re trying to make the operation implicit; yo=
ur proposal edges awfully close to the DWIM model of design, which we know =
causes problems when we try to add further extensions.</blockquote><div>

<br></div><div>Good point, but since we are still defining exactly what app=
Id does, I think we have the opportunity to make it explicit.=C2=A0</div></=
div><br></div></div>

--047d7b343296d8404004f186db09--

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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 15:03, Justin Uberti wrote:
>
> On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com 
> <mailto:adam@nostrum.com>> wrote:
>
>     On 2/3/14 14:25, Justin Uberti wrote:
>
>         It's not clear to me that adding a new attribute would solve
>         any problem that appid doesn't solve; you still need to deal
>         with the "other end not using it" issue.
>
>
>     Regardless of whether you think there's a problem in the legacy
>     interop case, the big benefit here is that an explicit indicator
>     by definition makes the operation explicit, and that
>     in-and-of-itself is a good thing.
>
>     The core issue is that you're trying to make the operation
>     implicit; your proposal edges awfully close to the DWIM model of
>     design, which we know causes problems when we try to add further
>     extensions.
>
>
> Good point, but since we are still defining exactly what appId does, I 
> think we have the opportunity to make it explicit.
>

I think I agree, but this isn't congruent with the proposal you made 
before. Omitting app-id as a means of rejecting a stream is an implicit 
operation, regardless of how well it's documented.

If you want to have something like "a=recv-appId:reject", that would be 
suitably explicit, but I really have a hard time imagining how something 
like that would be preferable to just having a new attribute.

/a

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    <div class="moz-cite-prefix">On 2/3/14 15:03, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-1iLOuJ-MhHpGp99j1qNieQTP9=EmTpibCh21fj5-6hMA@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra">On Mon, Feb 3, 2014 at 12:42 PM, Adam
          Roach <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:adam@nostrum.com" target="_blank">adam@nostrum.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">
              <div class="im">On 2/3/14 14:25, Justin Uberti wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  It's not clear to me that adding a new attribute would
                  solve any problem that appid doesn't solve; you still
                  need to deal with the "other end not using it" issue.<br>
                </blockquote>
                <br>
              </div>
              Regardless of whether you think there's a problem in the
              legacy interop case, the big benefit here is that an
              explicit indicator by definition makes the operation
              explicit, and that in-and-of-itself is a good thing.<br>
              <br>
              The core issue is that you're trying to make the operation
              implicit; your proposal edges awfully close to the DWIM
              model of design, which we know causes problems when we try
              to add further extensions.</blockquote>
            <div>
              <br>
            </div>
            <div>Good point, but since we are still defining exactly
              what appId does, I think we have the opportunity to make
              it explicit.Â </div>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
    I think I agree, but this isn't congruent with the proposal you made
    before. Omitting app-id as a means of rejecting a stream is an
    implicit operation, regardless of how well it's documented.<br>
    <br>
    If you want to have something like "a=recv-appId:reject", that would
    be suitably explicit, but I really have a hard time imagining how
    something like that would be preferable to just having a new
    attribute.<br>
    <br>
    /a<br>
  </body>
</html>

--------------020506060903060308030504--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 13:22:00 -0800
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 1:10 PM, Adam Roach <adam@nostrum.com> wrote:

>  On 2/3/14 15:03, Justin Uberti wrote:
>
>
> On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 2/3/14 14:25, Justin Uberti wrote:
>>
>>> It's not clear to me that adding a new attribute would solve any problem
>>> that appid doesn't solve; you still need to deal with the "other end not
>>> using it" issue.
>>>
>>
>>  Regardless of whether you think there's a problem in the legacy interop
>> case, the big benefit here is that an explicit indicator by definition
>> makes the operation explicit, and that in-and-of-itself is a good thing.
>>
>> The core issue is that you're trying to make the operation implicit; your
>> proposal edges awfully close to the DWIM model of design, which we know
>> causes problems when we try to add further extensions.
>
>
>  Good point, but since we are still defining exactly what appId does, I
> think we have the opportunity to make it explicit.
>
>
> I think I agree, but this isn't congruent with the proposal you made
> before. Omitting app-id as a means of rejecting a stream is an implicit
> operation, regardless of how well it's documented.
>
> If you want to have something like "a=recv-appId:reject", that would be
> suitably explicit, but I really have a hard time imagining how something
> like that would be preferable to just having a new attribute.
>

I was thinking a=recv-appId:0, which is both explicit, and has parallels
with the zero port form of rejecting a m= line.

--001a1136b6107ac4b404f1871e5f
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, Feb 3, 2014 at 1:10 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">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div class=3D"h5">
    <div>On 2/3/14 15:03, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><br>
        <div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 12:42 PM, Adam
          Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" t=
arget=3D"_blank">adam@nostrum.com</a>&gt;</span>
          wrote:<br>
          <div class=3D"gmail_quote">
            <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">
              <div>On 2/3/14 14:25, Justin Uberti wrote:<br>
                <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
                  It&#39;s not clear to me that adding a new attribute woul=
d
                  solve any problem that appid doesn&#39;t solve; you still
                  need to deal with the &quot;other end not using it&quot; =
issue.<br>
                </blockquote>
                <br>
              </div>
              Regardless of whether you think there&#39;s a problem in the
              legacy interop case, the big benefit here is that an
              explicit indicator by definition makes the operation
              explicit, and that in-and-of-itself is a good thing.<br>
              <br>
              The core issue is that you&#39;re trying to make the operatio=
n
              implicit; your proposal edges awfully close to the DWIM
              model of design, which we know causes problems when we try
              to add further extensions.</blockquote>
            <div>
              <br>
            </div>
            <div>Good point, but since we are still defining exactly
              what appId does, I think we have the opportunity to make
              it explicit.=C2=A0</div>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br></div></div>
    I think I agree, but this isn&#39;t congruent with the proposal you mad=
e
    before. Omitting app-id as a means of rejecting a stream is an
    implicit operation, regardless of how well it&#39;s documented.<br>
    <br>
    If you want to have something like &quot;a=3Drecv-appId:reject&quot;, t=
hat would
    be suitably explicit, but I really have a hard time imagining how
    something like that would be preferable to just having a new
    attribute.</div></blockquote><div><br></div><div>I was thinking a=3Drec=
v-appId:0, which is both explicit, and has parallels with the zero port for=
m of rejecting a m=3D line.=C2=A0</div></div><br></div></div>

--001a1136b6107ac4b404f1871e5f--

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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 15:22, Justin Uberti wrote:
>
> I was thinking a=recv-appId:0, which is both explicit, and has 
> parallels with the zero port form of rejecting a m= line.
>

I have no objection to such an approach.

/a

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From: Jonathan Lennox <jonathan@vidyo.com>
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On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;&nbsp;wrote:</div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
<div><br>
</div>
</div>
</body>
</html>

--_000_C433A27C348A402AB03607060746FADAvidyocom_--

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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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--047d7b6da02872d46104f1887ec4
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On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>
>  Also, this essentially makes recv-appId the converse of MSID; for
> simplicity, I think we should consider unifying these concepts into a
> single document.
>
>
>  AppID (as I envision it, as the author) and MSID (as I understand it)
> have rather different scopes, which are visible once sources are sent using
> more than one packet stream.
>
>  For example, an m-line which has both a primary encoded stream (of
> media) and a redundancy stream (of rtx packets) would have two appid
> values, but only one msid.  On the receive side, if you wanted to receive
> both a primary encoded stream and a redundancy stream, you would have two
> recv-appid values.
>
>  As such, appid and msid are solving different (though related) problems,
> and I don't think unifying them is a good idea.
>
>  Perhaps we need a recv-msid, explicitly?
>

That would be fine by me - it would allow a single invention (msid +
recv-msid) to focus on solving the three identified problems with MSTs in
unified-plan:
- correlation of a MediaStreamTrack to m= line in signaling (msid)
- correlation of packets from a MST to m= line ahead of signaling
(recv-msid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a
MST to an individual RTP flow, but we can let that develop on its own,
separate from the timetable of webrtc 1.0 and unified plan.

--047d7b6da02872d46104f1887ec4
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <span dir=3D"ltr">&=
lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.c=
om</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 style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
><div class=3D"im">
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div></div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br></div></div></blockquote><=
div><br></div><div>That would be fine by me - it would allow a single inven=
tion (msid + recv-msid) to focus on solving the three identified problems w=
ith MSTs in unified-plan:</div>

<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div><div>- correlation of packets from a MST to m=3D line ahead of signalin=
g (recv-msid)</div><div>- rejection of an individual MST (recv-msid)</div>
<div>
<br></div><div>Based on your description, appid seems to go one or two leve=
ls down from a MST to an individual RTP flow, but we can let that develop o=
n its own, separate from the timetable of webrtc 1.0 and unified plan.</div=
>

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

--047d7b6da02872d46104f1887ec4--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 15:03:36 -0800
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To: Jonathan Lennox <jonathan@vidyo.com>, Harald Alvestrand <hta@google.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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--089e011844dad5304004f1888911
Content-Type: text/plain; charset=UTF-8

+Harald as this affects the MSID doc


On Mon, Feb 3, 2014 at 3:00 PM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com>wrote:
>
>>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  Also, this essentially makes recv-appId the converse of MSID; for
>> simplicity, I think we should consider unifying these concepts into a
>> single document.
>>
>>
>>  AppID (as I envision it, as the author) and MSID (as I understand it)
>> have rather different scopes, which are visible once sources are sent using
>> more than one packet stream.
>>
>>  For example, an m-line which has both a primary encoded stream (of
>> media) and a redundancy stream (of rtx packets) would have two appid
>> values, but only one msid.  On the receive side, if you wanted to receive
>> both a primary encoded stream and a redundancy stream, you would have two
>> recv-appid values.
>>
>>  As such, appid and msid are solving different (though related)
>> problems, and I don't think unifying them is a good idea.
>>
>>  Perhaps we need a recv-msid, explicitly?
>>
>
> That would be fine by me - it would allow a single invention (msid +
> recv-msid) to focus on solving the three identified problems with MSTs in
> unified-plan:
> - correlation of a MediaStreamTrack to m= line in signaling (msid)
> - correlation of packets from a MST to m= line ahead of signaling
> (recv-msid)
> - rejection of an individual MST (recv-msid)
>
> Based on your description, appid seems to go one or two levels down from a
> MST to an individual RTP flow, but we can let that develop on its own,
> separate from the timetable of webrtc 1.0 and unified plan.
>
>
>

--089e011844dad5304004f1888911
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">+Harald as this affects the MSID doc</div><div class=3D"gm=
ail_extra"><br><br><div class=3D"gmail_quote">On Mon, Feb 3, 2014 at 3:00 P=
M, 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"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Mon, Feb 3=
, 2014 at 2:32 PM, Jonathan Lennox <span dir=3D"ltr">&lt;<a href=3D"mailto:=
jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.com</a>&gt;</span> wro=
te:<br>


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



<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
><div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div></div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br></div></div></blockquote><=
div><br></div></div></div><div>That would be fine by me - it would allow a =
single invention (msid + recv-msid) to focus on solving the three identifie=
d problems with MSTs in unified-plan:</div>


<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div><div>- correlation of packets from a MST to m=3D line ahead of signalin=
g (recv-msid)</div><div>- rejection of an individual MST (recv-msid)</div>

<div>
<br></div><div>Based on your description, appid seems to go one or two leve=
ls down from a MST to an individual RTP flow, but we can let that develop o=
n its own, separate from the timetable of webrtc 1.0 and unified plan.</div=
>


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

--089e011844dad5304004f1888911--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 3:10 PM, Adam Roach <adam@nostrum.com> wrote:

> On 2/3/14 15:22, Justin Uberti wrote:
>>=20
>> I was thinking a=3Drecv-appId:0, which is both explicit, and has paralle=
ls with the zero port form of rejecting a m=3D line.
>>=20
>=20
> I have no objection to such an approach.
>=20
> /a

+1 with Adam on we need something explicitly somewhere.=20

If we do it with some combination msid / app-id - I just want to make sure =
the behavior is still reasonable with things that don=92t low how to signal=
 ssrc.=20

 =

From jonathan@vidyo.com  Mon Feb  3 15:11:52 2014
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.



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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;</div>
<div>&nbsp;wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;&nbsp;wrote:</div=
>
<div class=3D"im">
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid &#4=
3; recv-msid) to focus on solving the three identified problems with MSTs i=
n unified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don't think point 2 will work.</div>
<div><br>
</div>
<div>I don't think a single, m-line-wide, identifier will work for the corr=
elation-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows i=
n the m-line. &nbsp;That's why we've designed appid the way we have.</div>
<div><br>
</div>
<div>I think appid is the right solution to point 2. &nbsp;More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid's
 job.</div>
<div><br>
</div>
<br>
</body>
</html>

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From juberti@google.com  Mon Feb  3 16:05:39 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 16:05:15 -0800
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To: Jonathan Lennox <jonathan@vidyo.com>
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>
>  On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com>
>  wrote:
>
>  On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com>wrote:
>
>>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  Also, this essentially makes recv-appId the converse of MSID; for
>> simplicity, I think we should consider unifying these concepts into a
>> single document.
>>
>>
>>  AppID (as I envision it, as the author) and MSID (as I understand it)
>> have rather different scopes, which are visible once sources are sent using
>> more than one packet stream.
>>
>>  For example, an m-line which has both a primary encoded stream (of
>> media) and a redundancy stream (of rtx packets) would have two appid
>> values, but only one msid.  On the receive side, if you wanted to receive
>> both a primary encoded stream and a redundancy stream, you would have two
>> recv-appid values.
>>
>>  As such, appid and msid are solving different (though related)
>> problems, and I don't think unifying them is a good idea.
>>
>>  Perhaps we need a recv-msid, explicitly?
>>
>
>  That would be fine by me - it would allow a single invention (msid +
> recv-msid) to focus on solving the three identified problems with MSTs in
> unified-plan:
> - correlation of a MediaStreamTrack to m= line in signaling (msid)
> - correlation of packets from a MST to m= line ahead of signaling
> (recv-msid)
> - rejection of an individual MST (recv-msid)
>
>  Based on your description, appid seems to go one or two levels down from
> a MST to an individual RTP flow, but we can let that develop on its own,
> separate from the timetable of webrtc 1.0 and unified plan.
>
>
>  Points 1 and 3 make sense to me, and are I think what msid is for; but I
> don't think point 2 will work.
>
>  I don't think a single, m-line-wide, identifier will work for the
> correlation-ahead-of-signaling problem, once you have multiple RTP flows in
> the m-line.  That's why we've designed appid the way we have.
>

>  I think appid is the right solution to point 2.  More specifically, the
> problem appid is solving is better expressed as the correlation of packets
> from an RTP flow to an m-line; the correlation from the m-line to a MST is
> a separate step, and is msid's job.
>

Agree that msid should map m-line to MST, but it would seem recv-msid could
map RTP flow to m=line as well as indicate whether the receiver is in fact
interested in receiving the MST associated with said m= line. Note that
recv-msid doesn't specify an actual MSID; its purpose is to indicate that
the receiver wants to receive data from the remote side's MST, and provide
a mechanism for the sender to mark the data accordingly. (Which ends up
seeming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by
with only 2.

--001a1133ed3a51b0a304f1896696
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, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <span dir=3D"ltr">&=
lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.c=
om</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 style=3D"word-wrap:break-word">
<br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;</div><div><div c=
lass=3D"h5">
<div>=C2=A0wrote:</div>
<br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
>
<div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid + r=
ecv-msid) to focus on solving the three identified problems with MSTs in un=
ified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div></div></div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don&#39;t think point 2 will work.</div>
<div><br>
</div>
<div>I don&#39;t think a single, m-line-wide, identifier will work for the =
correlation-ahead-of-signaling=C2=A0problem, once you have multiple RTP flo=
ws in the m-line. =C2=A0That&#39;s why we&#39;ve designed appid the way we =
have.</div>

</div></blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:b=
reak-word">
<div><br>
</div>
<div>I think appid is the right solution to point 2. =C2=A0More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid&#39;s
 job.</div></div></blockquote><div><br></div><div>Agree that msid should ma=
p m-line to MST, but it would seem recv-msid could map RTP flow to m=3Dline=
 as well as indicate whether the receiver is in fact interested in receivin=
g the MST associated with said m=3D line. Note that recv-msid doesn&#39;t s=
pecify an actual MSID; its purpose is to indicate that the receiver wants t=
o receive data from the remote side&#39;s MST, and provide a mechanism for =
the sender to mark the data accordingly. (Which ends up seeming a lot like =
appid.)</div>

<div><br></div><div>Generally, I am hesitant to depend on 3 things if we th=
ink we could get by with only 2.</div></div><br></div></div>

--001a1133ed3a51b0a304f1896696--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Date: Tue, 4 Feb 2014 14:13:35 +0000
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Cc: Cullen Jennings <fluffy@cisco.com>, Harald Alvestrand <hta@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Subject: [rtcweb] Data Channel question: correlating streams with channels
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Section 6.4 of draft-ietf-rtcweb-data-channel-06 says:

    The realization of a bidirectional Data Channel is a pair of one
    incoming stream and one outgoing SCTP stream.

    Note that there's no requirement for the SCTP streams used to create
    a bidirectional channel have the same number in each direction.  How
    stream values are selected is protocol and implementation dependent.

The note seems to be something of a cop-out. But I'll accept for the 
moment that this can be true when external negotiation is used to define 
the channels.

But I've looked through draft-ietf-rtcweb-data-protocol-01, and I don't 
find any specification there of how the pairing is done.

Section 4 of the protocol draft says:

    The data channel protocol uses a two way handshake to open a data
    channel.  The side wanting to open a data channel selects an unused
    Stream and sends a DATA_CHANNEL_OPEN message.  The peer responds with
    a DATA_CHANNEL_ACK message.  Then the data channel is open.

The side doing the OPEN chooses what will be its outgoing stream id. 
That then becomes the incoming stream id of the peer. I had always 
assumed that the same stream id would be used for the ACK. But I find no 
evidence to support that assumption. For it to be anything else I would 
think the ACK would need to carry more information to correlate back to 
the OPEN message.

	Thanks,
	Paul

From jonathan@vidyo.com  Tue Feb  4 08:13:11 2014
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
Thread-Index: AQHPIS46pADKsTFIzECMWWmLIWbna5qkgc6AgAAH3gCAAAMqAIAADvKAgAEOagA=
Date: Tue, 4 Feb 2014 16:13:06 +0000
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 7:05 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:

On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.

Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual MSID; its purpose is to indicate that=
 the receiver wants to receive data from the remote side's MST, and provide=
 a mechanism for the sender to mark the data accordingly. (Which ends up se=
eming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.

I understand that, but I'm also hesitant to have one mechanism do two (it s=
eems to me) fairly unrelated things, especially where the scenarios in whic=
h the problems it's solving are applicable vary fairly widely.

As far as I can tell, the only use case anyone's identified for MSID is Web=
RTC, whereas the disambiguation problem occurs (and thus appid and recv-app=
id are applicable) for anything using Bundle with Unified-like semantics.


--_000_2DF6D66078B34F53AF39B3651E9BBB46vidyocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Feb 3, 2014, at 7:05 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;</div>
<div>&nbsp;wrote:</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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 style=3D"word-wrap:break-word"><br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;</div>
<div>
<div class=3D"h5">
<div>&nbsp;wrote:</div>
<br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;&nbsp;wrote:</div=
>
<div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid &#4=
3; recv-msid) to focus on solving the three identified problems with MSTs i=
n unified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
</div>
</div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don't think point 2 will work.</div>
<div><br>
</div>
<div>I don't think a single, m-line-wide, identifier will work for the corr=
elation-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows i=
n the m-line. &nbsp;That's why we've designed appid the way we have.</div>
</div>
</blockquote>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div><br>
</div>
<div>I think appid is the right solution to point 2. &nbsp;More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid's
 job.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Agree that msid should map m-line to MST, but it would seem recv-msid =
could map RTP flow to m=3Dline as well as indicate whether the receiver is =
in fact interested in receiving the MST associated with said m=3D line. Not=
e that recv-msid doesn't specify an
 actual MSID; its purpose is to indicate that the receiver wants to receive=
 data from the remote side's MST, and provide a mechanism for the sender to=
 mark the data accordingly. (Which ends up seeming a lot like appid.)</div>
<div><br>
</div>
<div>Generally, I am hesitant to depend on 3 things if we think we could ge=
t by with only 2.</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
I understand that, but I'm also hesitant to have one mechanism do two (it s=
eems to me) fairly unrelated things, especially where the scenarios in whic=
h the problems it's solving are applicable vary fairly widely.</div>
<div><br>
</div>
<div>As far as I can tell, the only use case anyone's identified for MSID i=
s WebRTC, whereas the disambiguation problem occurs (and thus appid and rec=
v-appid are applicable) for anything using Bundle with Unified-like semanti=
cs.</div>
<br>
</body>
</html>

--_000_2DF6D66078B34F53AF39B3651E9BBB46vidyocom_--

From jonathan@vidyo.com  Tue Feb  4 08:13:50 2014
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Date: Tue, 4 Feb 2014 16:13:42 +0000
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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MSID doesn't affect the transport level.  Appid, however, defines an RTCP S=
DES item, and an RTP header extension.

On Feb 4, 2014, at 9:13 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om<mailto:christer.holmberg@ericsson.com>>
 wrote:

Hi,

Just for my clarification: none of this would have any impact on transport =
level features related to RTCP, STUN, etc?

Regards,

Christer


From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:bounces@ietf.org>] On B=
ehalf Of Justin Uberti
Sent: 4. helmikuuta 2014 2:05
To: Jonathan Lennox
Cc: Cullen Jennings; Harald Alvestrand; rtcweb@ietf.org<mailto:rtcweb@ietf.=
org>; mmusic
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP



On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:

On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:


On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:


Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.

Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual MSID; its purpose is to indicate that=
 the receiver wants to receive data from the remote side's MST, and provide=
 a mechanism for the sender to mark the data accordingly. (Which ends up se=
eming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.



--_000_B9399713DDEF4680B5E4E9C817372133vidyocom_
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</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div>MSID doesn't affect the transport level. &nbsp;Appid, however, defines=
 an RTCP SDES item, and an RTP header extension.</div>
<br>
<div>
<div>On Feb 4, 2014, at 9:13 AM, Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt;</div>
<div>&nbsp;wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: He=
lvetica; font-size: medium; font-style: normal; font-variant: normal; font-=
weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; te=
xt-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space=
: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -we=
bkit-text-stroke-width: 0px; ">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Hi,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Just for my clarification: none of this would have any im=
pact on transport level features related to RTCP, STUN, etc?<o:p></o:p></sp=
an></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Regards,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Christer<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; ">From:=
</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif;=
 "><span class=3D"Apple-converted-space">&nbsp;</span>rtcweb [mailto:rtcweb=
-<a href=3D"mailto:bounces@ietf.org" style=3D"color: purple; text-decoratio=
n: underline; ">bounces@ietf.org</a>]<span class=3D"Apple-converted-space">=
&nbsp;</span><b>On
 Behalf Of<span class=3D"Apple-converted-space">&nbsp;</span></b>Justin Ube=
rti<br>
<b>Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>4. helmikuut=
a 2014 2:05<br>
<b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span>Jonathan Lenno=
x<br>
<b>Cc:</b><span class=3D"Apple-converted-space">&nbsp;</span>Cullen Jenning=
s; Harald Alvestrand;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; mmusic<br>
<b>Subject:</b><span class=3D"Apple-converted-space">&nbsp;</span>Re: [rtcw=
eb] Rejecting MediaStreamTracks in JSEP<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<p class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font=
-family: 'Times New Roman', serif; ">
<o:p>&nbsp;</o:p></p>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox &lt;<a href=3D"mailto:jonat=
han@vidyo.com" target=3D"_blank" style=3D"color: purple; text-decoration: u=
nderline; ">jonathan@vidyo.com</a>&gt; wrote:<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank" style=3D"color: purple; text-decoration: underli=
ne; ">juberti@google.com</a>&gt;<o:p></o:p></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
&nbsp;wrote:<o:p></o:p></div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<br>
<br>
<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox &lt;<a href=3D"mailto:jonat=
han@vidyo.com" target=3D"_blank" style=3D"color: purple; text-decoration: u=
nderline; ">jonathan@vidyo.com</a>&gt; wrote:<o:p></o:p></div>
<div>
<blockquote style=3D"border-style: none none none solid; border-left-width:=
 1pt; border-left-color: rgb(204, 204, 204); padding: 0cm 0cm 0cm 6pt; marg=
in-left: 4.8pt; margin-right: 0cm; ">
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank" style=3D"color: purple; text-decoration: underli=
ne; ">juberti@google.com</a>&gt;&nbsp;wrote:<o:p></o:p></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<br>
<br>
<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 10pt; font-family: Arial, sans-serif; ">Also, thi=
s essentially makes recv-appId the converse of MSID; for simplicity, I thin=
k we should consider unifying these concepts into a single document.<o:p></=
o:p></span></div>
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid. &nbsp;On the receive side, if you wanted to receive both a pri=
mary encoded stream and a redundancy
 stream, you would have two recv-appid values.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Perhaps we need a recv-msid, explicitly?<o:p></o:p></div>
</div>
</blockquote>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
That would be fine by me - it would allow a single invention (msid &#43; re=
cv-msid) to focus on solving the three identified problems with MSTs in uni=
fied-plan:<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)<o:p></=
o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- rejection of an individual MST (recv-msid)<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.<o:p></o:p></div>
</div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows in the=
 m-line. &nbsp;That's why we've designed appid the way we have.<o:p></o:p><=
/div>
</div>
</div>
<blockquote style=3D"border-style: none none none solid; border-left-width:=
 1pt; border-left-color: rgb(204, 204, 204); padding: 0cm 0cm 0cm 6pt; marg=
in-left: 4.8pt; margin-right: 0cm; ">
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
I think appid is the right solution to point 2. &nbsp;More specifically, th=
e problem appid is solving is better expressed as the correlation of packet=
s from an RTP flow to an m-line; the correlation from the m-line to a MST i=
s a separate step, and is msid's job.<o:p></o:p></div>
</div>
</blockquote>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual
 MSID; its purpose is to indicate that the receiver wants to receive data f=
rom the remote side's MST, and provide a mechanism for the sender to mark t=
he data accordingly. (Which ends up seeming a lot like appid.)<o:p></o:p></=
div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.<o:p></o:p></div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</body>
</html>

--_000_B9399713DDEF4680B5E4E9C817372133vidyocom_--

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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: Comments on draft-ietf-rtcweb-stun-consent-freshness-00
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-stun-consent-freshness-00
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Hi Magnus,

Sorry for delay. Please see inline for answers


>-----Original Message-----
>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>Sent: Friday, November 15, 2013 2:32 PM
>To: Ram Mohan R (rmohanr);
>draft-ietf-rtcweb-stun-consent-freshness@tools.ietf.org; rtcweb@ietf.org
>Subject: Re: Comments on draft-ietf-rtcweb-stun-consent-freshness-00
>
>Hi,
>
>Removing things where I am in agreement or like your response.
>
>On 2013-11-14 18:23, Ram Mohan R (rmohanr) wrote:
>> Hi Magnus,
>>=20
>> Thanks for your comments. Please see inline for responses.
>>=20
>>=20
>> -----Original Message-----
>> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
>> Date: Tuesday, 12 November 2013 8:28 PM
>
>>>
>>> 3. Section 1:
>>> "This document describes a new STUN usage with a request and response
>>>   which verifies the remote peer consents to receive traffic, and
>>>   detects loss of liveness."
>>>
>>> I am missing a word after "request and response", transaction or
>>> messages maybe?
>>=20
>> NEW:
>> This document describes a new STUN usage with a request and response
>> messages which verifies the remote peer consents to receive traffic,
>> and detects loss of liveness.
>
>Isn't it either "which verifies the remote peer's consent to" or "which
>verifies that the remote peer consents to"?


I will change to "which verifies the remote peer's consent to=B2



>
>
>
>>
>>>
>>> 5. Section 4:
>>>   Consent freshness serves as a circuit breaker (so that if the path or
>>>   remote peer fails the WebRTC browser stops sending all traffic on
>>>   that remote peer), determining session liveness serves the purpose of
>>>   notifying the application of connectivity failure so that the
>>>   application can take appropriate action.
>>>
>>=20
>>>
>>> What is the definition of a peer here?
>>=20
>> Peer is the sink for the traffic originated from the sender for that
>> flow
>> (5 tuple). If a browser uses different 5 tuple for each stream(Audio,
>> video, data) it sends then it should do consent for each flow. If it
>> uses same 5-tuple (mux case) then there will be only one consent.
>
>Please be explicit about this. I think it is easy to interpret that this
>would apply to all flows from one end-point to a given destination
>address.

Sure. We will add the above text and make it clearer in Section 4


>
>>=20
>>=20
>>=20
>>> What scope does the stop sending
>>> have?
>>=20
>> The stream that is being sent on a flow(5 tuple) for which a consent
>> has failed will be stopped. If all the streams are muxed on a same 5
>> tuple the entire traffic will be stopped.
>>=20
>
>I am fine with this, as long as the document is explicit about this being
>the scope.

Will make this explicit in the draft


>
>
>>>
>>> 6. Section 4:
>>>   1.  A consent timer, Tc, whose value is determined by the browser.
>>>       This value MUST be 15 seconds.
>>>
>>> I see a contradiction here, should it be determined by the browser or
>>> be
>>> 15 s? If it is 15, can you please motivate why it is this value, or
>>> point to where it is motivated?
>>=20
>> The default value of 15 was some thing that EKR and Justin gave us
>> based on the implementation/testing and fine tuning they have done on
>> their browsers. I agree we can have some text around it that explains
>> how this number is arrived. We will add the same.
>
>Good, I think it is important we understand the considerations of why
>this is a reasonable choice.

Sure. Will check with Justin/EKR and add some text that justifies why 15
sec is default.



>
>
>>>
>>> 8. Section 4:
>>>   If a valid STUN Binding Response is not received after 500ms, the
>>>   STUN Binding Request is retransmitted (with the same Transaction ID
>>>   and all other fields).  As long as a valid STUN Binding Response is
>>>   not received, this retransmission is repeated every 500ms until Tf
>>>   seconds have elapsed or a valid response is received.
>>>
>>> So no exponential back-off on the retransmission timer. I think that
>>> is fine. However, I think it do require you to expand a bit on what
>>> happens when one gets multiple response on the same Transaction ID.
>>> For example additional responses do they or do they not reset the Tc
>>>timer?
>>=20
>> Additional responses MUST reset the Tc timer.
>>=20
>
>Okay, that appears fine.
>
>>=20
>>>
>>> Then I wonder about the cases where the RTT is above 500 ms, should
>>> one then scale this factor to be once per RTT?
>
>What about this question?

The ICE agent can learn the RTT of a given path from RTCP reports (if
present). If there is no RTCP reports then it can use STUN BindReq/Res to
learn RTT.

STUN allows you to learn the RTT of a path when you sending Binding
request and get a Binding response for a candidate pair. Before consent
timer
is started the browser=B9s ICE agent would have done ICE checks and
concluded on a candidate for each media stream.  So an ICE agent can use
the RTT learnt here.
Thoughts on this ?


>
>>>
>>> 9. Section 4:
>>> "with the same Transaction ID"
>>>
>>> Why not use a new transaction ID for each sent packet? It would allow
>>> tracking loss and determine RTT more reliable. All for the small cost
>>> of keeping track of slightly more Transaction IDs?
>>=20
>> Yes, new transaction ID will help to determine the packet loss and RTT.
>>=20
>
>Yes, so what is preferable here? Doing cloned retransmission, or
>generating new TIDs for each outgoing request?

Using same Transaction ID would lead the peer ICE agent think this request
as a retransmission and it may not reset its Consent timer.  So I think
using new ID is better.



>
>>=20
>>=20
>>>
>>> 10. Section 4.
>>>   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.).  Note the
>>>   definition of a received packet is liberal, and includes an SRTP
>>>   packet that fails authentication, a STUN Binding Request with an
>>>   invalid USERNAME or PASSWORD, or any other packet.
>>>
>>> I think this requires some considerations in regards to RTT. If the
>>> RTT is 700 ms, high for a real-time app but not unheard of at all.
>>> Then a Tr =3D 1 second would fire on a single loss.
>>=20
>> Ok.  will start a discussion for this item in a separate email.
>>=20

As discussed above the ICE agent can learn the RTT of a given path from
RTCP reports (if present) or compute the RTT from STUN BindReq/Response.
It can then use the learnt value
to influence the Tr value.

Regards,

Ram

>
>Good
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>


From christer.holmberg@ericsson.com  Wed Feb  5 03:23:41 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Subject: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Hi,

This is related to the e-mail Paul sent yesterday (Data Channel question: c=
orrelating streams with channels).

Section 5 of draft-ietf-rtcweb-data-channel-06 says:

"Each SCTP user message contains a so called Payload Protocol
                Identifier (PPID) that is passed to SCTP by its upper layer=
 and sent
                to its peer.  This value can be used to multiplex multiple =
protocols
                over a single SCTP association.  The sender provides for ea=
ch
                protocol a specific PPID and the receiver can demultiplex t=
he
                messages based on the received PPID."

Now, when I look at the IANA registration page for PPID values, most seem t=
o be associated with specific protocols.

However, if I e.g. use PPID values 51-54 (as rtcweb applications will do) t=
hey don't say anything about the protocol. They refer to an encoding format=
, and multiple protocols may have the same format.

I would assume that the SCTP stream id would have to be used to multiplex p=
rotocols.

Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">This is related to the e-mail Paul sent yesterday (D=
ata Channel question: correlating streams with channels).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Section 5</b> of <b>draft-ietf-rtcweb-data-channe=
l-06</b> says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;Each SCTP user m=
essage contains a so called Payload Protocol<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Identifier (PPID) that is passed to SCTP =
by its upper layer and sent<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; to its peer.&nbsp; This value can be used=
 to multiplex multiple protocols<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; over a single SCTP association.&nbsp; The=
 sender provides for each<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; protocol a specific PPID and the receiver=
 can demultiplex the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; messages based on the received PPID.&#822=
1;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Now, when I look at the IANA registration page for P=
PID values, most seem to be associated with specific protocols.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">However, if I e.g. use PPID values 51-54 (as rtcweb =
applications will do) they don&#8217;t say anything about the protocol. The=
y refer to an encoding format, and multiple protocols may have the same for=
mat.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I would assume that the SCTP stream id would have to=
 be used to multiplex protocols.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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The PPID isn't exposed to the application.  It's just used to
distinguish text/binary/control.    If an application uses different
data channels with different protocols, it should use the SID to
distinguish between them, not the PPID.  In fact, even if it wanted to
use the PPID, it wouldn't be able to.

On Wed, Feb 5, 2014 at 3:23 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> Hi,
>
>
>
> This is related to the e-mail Paul sent yesterday (Data Channel question:
> correlating streams with channels).
>
>
>
> Section 5 of draft-ietf-rtcweb-data-channel-06 says:
>
>
>
> =E2=80=9CEach SCTP user message contains a so called Payload Protocol
>
>                 Identifier (PPID) that is passed to SCTP by its upper lay=
er
> and sent
>
>                 to its peer.  This value can be used to multiplex multipl=
e
> protocols
>
>                 over a single SCTP association.  The sender provides for
> each
>
>                 protocol a specific PPID and the receiver can demultiplex
> the
>
>                 messages based on the received PPID.=E2=80=9D
>
>
>
> Now, when I look at the IANA registration page for PPID values, most seem=
 to
> be associated with specific protocols.
>
>
>
> However, if I e.g. use PPID values 51-54 (as rtcweb applications will do)
> they don=E2=80=99t say anything about the protocol. They refer to an enco=
ding
> format, and multiple protocols may have the same format.
>
>
>
> I would assume that the SCTP stream id would have to be used to multiplex
> protocols.
>
>
>
> Regards,
>
>
>
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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On Wed, Feb 5, 2014 at 12:40 PM, Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com> wrote:
>> The PPID isn't exposed to the application.  It's just used to
>> distinguish text/binary/control.    If an application uses different
>> data channels with different protocols, it should use the SID to
>> distinguish between them, not the PPID.  In fact, even if it wanted to
>> use the PPID, it wouldn't be able to.
> [Raju] PPID is also used by browsers (and other webrtc implementors like native clients) to fragment user given large data into smaller fragments (via "XYZ partial", "XYZ last" PPIDs) to avoid monopolizing the SCTP link for a single data channel's data.
> See http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-6.6 for more details.

I believe that Firefox is the browser that does that.  At the last
IETF, I think we agreed that it's a temporary fix at best while we
wait until this problem is solved at the SCTP layer and not above it.

I'm not aware of any native clients that use this technique, unless
there are native clients that use Firefox's code.   Perhaps there are;
 I just don't know of any.

In any case, it's not part of the standard, and I think the plan is
that it won't be needed long-term.

>
> -Raju
>

From Michael.Tuexen@lurchi.franken.de  Wed Feb  5 13:11:55 2014
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On Feb 5, 2014, at 10:09 PM, Peter Thatcher <pthatcher@google.com> =
wrote:

> On Wed, Feb 5, 2014 at 12:40 PM, Makaraju, Maridi Raju (Raju)
> <Raju.Makaraju@alcatel-lucent.com> wrote:
>>> The PPID isn't exposed to the application.  It's just used to
>>> distinguish text/binary/control.    If an application uses different
>>> data channels with different protocols, it should use the SID to
>>> distinguish between them, not the PPID.  In fact, even if it wanted =
to
>>> use the PPID, it wouldn't be able to.
>> [Raju] PPID is also used by browsers (and other webrtc implementors =
like native clients) to fragment user given large data into smaller =
fragments (via "XYZ partial", "XYZ last" PPIDs) to avoid monopolizing =
the SCTP link for a single data channel's data.
>> See =
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-6.6 =
for more details.
>=20
> I believe that Firefox is the browser that does that.  At the last
> IETF, I think we agreed that it's a temporary fix at best while we
> wait until this problem is solved at the SCTP layer and not above it.
>=20
> I'm not aware of any native clients that use this technique, unless
> there are native clients that use Firefox's code.   Perhaps there are;
> I just don't know of any.
>=20
> In any case, it's not part of the standard, and I think the plan is
> that it won't be needed long-term.
As discussed in the last IETF meeting, this PPID based fragmentation and =
reassembly
will be removed from the ID.

Best regards
Michael
>=20
>>=20
>> -Raju
>>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Date: Wed, 5 Feb 2014 21:32:46 +0000
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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> > I believe that Firefox is the browser that does that.  At the last
> > IETF, I think we agreed that it's a temporary fix at best while we
> > wait until this problem is solved at the SCTP layer and not above it.
[Raju] Using NDATA I believe.

> >
> > I'm not aware of any native clients that use this technique, unless
> > there are native clients that use Firefox's code.   Perhaps there are;
> > I just don't know of any.
> >
> > In any case, it's not part of the standard, and I think the plan is
> > that it won't be needed long-term.
> As discussed in the last IETF meeting, this PPID based fragmentation and
> reassembly
> will be removed from the ID.
[Raju] Can I then assume the reason for such removal is the availability of=
 NDATA (http://tools.ietf.org/html/draft-stewart-tsvwg-sctp-ndata-03) soone=
r than later, which will make such PPID based fragmentation/reassembly imme=
diately obsolete?

[Raju] -Raju

From Michael.Tuexen@lurchi.franken.de  Wed Feb  5 13:47:42 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 5, 2014, at 10:32 PM, "Makaraju, Maridi Raju (Raju)" =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>>> I believe that Firefox is the browser that does that.  At the last
>>> IETF, I think we agreed that it's a temporary fix at best while we
>>> wait until this problem is solved at the SCTP layer and not above =
it.
> [Raju] Using NDATA I believe.
>=20
>>>=20
>>> I'm not aware of any native clients that use this technique, unless
>>> there are native clients that use Firefox's code.   Perhaps there =
are;
>>> I just don't know of any.
>>>=20
>>> In any case, it's not part of the standard, and I think the plan is
>>> that it won't be needed long-term.
>> As discussed in the last IETF meeting, this PPID based fragmentation =
and
>> reassembly
>> will be removed from the ID.
> [Raju] Can I then assume the reason for such removal is the =
availability of NDATA =
(http://tools.ietf.org/html/draft-stewart-tsvwg-sctp-ndata-03) sooner =
than later, which will make such PPID based fragmentation/reassembly =
immediately obsolete?
Right. The point was not to have two solutions, but use one.

Best regards
Michael
>=20
> [Raju] -Raju
>=20


From pkyzivat@alum.mit.edu  Wed Feb  5 13:59:53 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Will NDATA support be required for rtcweb?

	Thanks,
	Paul

On 2/5/14 4:47 PM, Michael Tuexen wrote:
> On Feb 5, 2014, at 10:32 PM, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com> wrote:
>
>>>> I believe that Firefox is the browser that does that.  At the last
>>>> IETF, I think we agreed that it's a temporary fix at best while we
>>>> wait until this problem is solved at the SCTP layer and not above it.
>> [Raju] Using NDATA I believe.
>>
>>>>
>>>> I'm not aware of any native clients that use this technique, unless
>>>> there are native clients that use Firefox's code.   Perhaps there are;
>>>> I just don't know of any.
>>>>
>>>> In any case, it's not part of the standard, and I think the plan is
>>>> that it won't be needed long-term.
>>> As discussed in the last IETF meeting, this PPID based fragmentation and
>>> reassembly
>>> will be removed from the ID.
>> [Raju] Can I then assume the reason for such removal is the availability of NDATA (http://tools.ietf.org/html/draft-stewart-tsvwg-sctp-ndata-03) sooner than later, which will make such PPID based fragmentation/reassembly immediately obsolete?
> Right. The point was not to have two solutions, but use one.
>
> Best regards
> Michael
>>
>> [Raju] -Raju
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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> > [Raju] Can I then assume the reason for such removal is the availabilit=
y
> of NDATA (http://tools.ietf.org/html/draft-stewart-tsvwg-sctp-ndata-03)
> sooner than later, which will make such PPID based fragmentation/reassemb=
ly
> immediately obsolete?
> Right. The point was not to have two solutions, but use one.
[Raju] That makes perfect sense and NDATA is a better solution too! That al=
so pushes browser vendors to support NDATA sooner than later.

-Raju

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Date: Wed, 5 Feb 2014 22:35:54 +0000
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol	multiplexing
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> Will NDATA support be required for rtcweb?
[Raju] Yes. It is mentioned in http://tools.ietf.org/html/draft-ietf-rtcweb=
-data-channel-06#page-3 which says the following:

   The SCTP base protocol specified in [RFC4960] does not support the
   interleaving of user messages.  Therefore sending a large user
   message can monopolize the SCTP association.  To overcome this
   limitation, [I-D.stewart-tsvwg-sctp-ndata]  defines an extension to
   support message interleaving.  Once such an extension is available,=20
   it SHOULD be used.

From pkyzivat@alum.mit.edu  Wed Feb  5 14:45:34 2014
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Data Channel question: correlating streams with channels
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Sorry. I finally found the normative statement I was wanting:

    If a DATA_CHANNEL_OPEN message is received on an unused stream, the
    stream identifier corresponds to the role of the peer and all
    parameters in the DATA_CHANNEL_OPEN message are valid, then a
    corresponding DATA_CHANNEL_ACK message is sent on the stream with the
    same stream identifier as the one the DATA_CHANNEL_OPEN message was
    received on.

This satisfies my concern.

	Thanks,
	Paul

On 2/4/14 11:07 AM, Paul Kyzivat wrote:
> Section 6.4 of draft-ietf-rtcweb-data-channel-06 says:
>
>     The realization of a bidirectional Data Channel is a pair of one
>     incoming stream and one outgoing SCTP stream.
>
>     Note that there's no requirement for the SCTP streams used to create
>     a bidirectional channel have the same number in each direction.  How
>     stream values are selected is protocol and implementation dependent.
>
> The note seems to be something of a cop-out. But I'll accept for the
> moment that this can be true when external negotiation is used to define
> the channels.
>
> But I've looked through draft-ietf-rtcweb-data-protocol-01, and I don't
> find any specification there of how the pairing is done.
>
> Section 4 of the protocol draft says:
>
>     The data channel protocol uses a two way handshake to open a data
>     channel.  The side wanting to open a data channel selects an unused
>     Stream and sends a DATA_CHANNEL_OPEN message.  The peer responds with
>     a DATA_CHANNEL_ACK message.  Then the data channel is open.
>
> The side doing the OPEN chooses what will be its outgoing stream id.
> That then becomes the incoming stream id of the peer. I had always
> assumed that the same stream id would be used for the ACK. But I find no
> evidence to support that assumption. For it to be anything else I would
> think the ACK would need to carry more information to correlate back to
> the OPEN message.
>
>      Thanks,
>      Paul
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From pkyzivat@alum.mit.edu  Wed Feb  5 15:03:37 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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On 2/5/14 5:35 PM, Makaraju, Maridi Raju (Raju) wrote:
>> Will NDATA support be required for rtcweb?
> [Raju] Yes. It is mentioned in http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#page-3 which says the following:
>
>     The SCTP base protocol specified in [RFC4960] does not support the
>     interleaving of user messages.  Therefore sending a large user
>     message can monopolize the SCTP association.  To overcome this
>     limitation, [I-D.stewart-tsvwg-sctp-ndata]  defines an extension to
>     support message interleaving.  Once such an extension is available,
>     it SHOULD be used.
>

I guess that begs the question of what "Once such an extension is 
available" means, and what "it SHOULD be used" means. :-(

"available" could mean:

- once it is defined in a published RFC
- once it is defined in a WG draft or RFC
- once it is defined in an individual draft
   (e.g., draft-stewart-tsvwg-sctp-ndata-03)
- once it it is implemented in the browser you are using

"SHOULD be used" could apply to:
- browser implementers
- application implementers

And then there is the question of what "SHOULD" means. Specifically, 
what are the conditions where it would be acceptable to *not* follow this?

I *hope* this is intended to mean that browser implementers MUST 
*implement* it, and *use* it in their implementation of DataChannel once 
draft-stewart-tsvwg-sctp-ndata becomes an RFC. And SHOULD do so now 
based on draft-stewart-tsvwg-sctp-ndata-03.

	Thanks,
	Paul

From Raju.Makaraju@alcatel-lucent.com  Wed Feb  5 15:33:18 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB Data Channel: message interleaving
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Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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> "available" could mean:
>=20
> - once it is defined in a published RFC
> - once it is defined in a WG draft or RFC
> - once it is defined in an individual draft
>    (e.g., draft-stewart-tsvwg-sctp-ndata-03)
[Raju] I believe this is the case.
> - once it it is implemented in the browser you are using
[Raju] Browser implementations probably be more proactive in adapting this =
draft similar to number of other drafts that are being adapted in individua=
l draft state.
I think this is one way browsers can differentiate themselves from others.
So, in my opinion, the "SHOULD" below probably ok and does not have to be a=
 "MUST" as this NDATA is not a MUST have to deliver data channel requiremen=
ts. If it has to be a "MUST" then I believe it has to be part of the requir=
ement section 4 of http://tools.ietf.org/html/draft-ietf-rtcweb-data-channe=
l-06#section-4 and list it as "MUST".

>=20
> "SHOULD be used" could apply to:
> - browser implementers
> - application implementers
>=20
> And then there is the question of what "SHOULD" means. Specifically,
> what are the conditions where it would be acceptable to *not* follow this=
?
>=20
> I *hope* this is intended to mean that browser implementers MUST
> *implement* it, and *use* it in their implementation of DataChannel once
> draft-stewart-tsvwg-sctp-ndata becomes an RFC. And SHOULD do so now
> based on draft-stewart-tsvwg-sctp-ndata-03.
[Raju] I am not sure if this NDATA is a deal breaker for webrtc 1.0?! Some =
implementations may support while others not as NDATA is negotiated during =
association setup; if one end does not support then it is disabled.

If you don't have NDATA then one data channel stream may monopolize the SCT=
P link and could affect scenarios like:
- You sent 1GB file using data channel stream 0 (all at once without using =
something like MSRP).
- Then immediately, you send a small "Hi, some important stuff!" chat messa=
ge using data channel stream 1.
Without NDATA, the other end will receive the text msg ONLY AFTER receiving=
 the 1GB file!! With NDATA chat msg gets equal share of the SCTP link and g=
ets delivered much sooner.
You can apply this to even more sensitive gaming apps.

-Raju


From christer.holmberg@ericsson.com  Wed Feb  5 23:56:54 2014
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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Peter Thatcher <pthatcher@google.com>
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Hi,

> As discussed in the last IETF meeting, this PPID based fragmentation and =
reassembly will be removed from the ID.

I assume the associated PPID values will also be un-registered with IANA?

Regards,

Christer


From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 00:08:42 2014
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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On Feb 6, 2014, at 12:33 AM, "Makaraju, Maridi Raju (Raju)" =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>> "available" could mean:
>>=20
>> - once it is defined in a published RFC
>> - once it is defined in a WG draft or RFC
>> - once it is defined in an individual draft
>>   (e.g., draft-stewart-tsvwg-sctp-ndata-03)
> [Raju] I believe this is the case.
draft-stewart-tsvwg-sctp-ndata has been adopted as a WG item in the =
TSVWG. We need
to update and resubmit it.
>> - once it it is implemented in the browser you are using
> [Raju] Browser implementations probably be more proactive in adapting =
this draft similar to number of other drafts that are being adapted in =
individual draft state.
> I think this is one way browsers can differentiate themselves from =
others.
> So, in my opinion, the "SHOULD" below probably ok and does not have to =
be a "MUST" as this NDATA is not a MUST have to deliver data channel =
requirements. If it has to be a "MUST" then I believe it has to be part =
of the requirement section 4 of =
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-4 =
and list it as "MUST".
>=20
>>=20
>> "SHOULD be used" could apply to:
>> - browser implementers
>> - application implementers
>>=20
>> And then there is the question of what "SHOULD" means. Specifically,
>> what are the conditions where it would be acceptable to *not* follow =
this?
>>=20
>> I *hope* this is intended to mean that browser implementers MUST
>> *implement* it, and *use* it in their implementation of DataChannel =
once
>> draft-stewart-tsvwg-sctp-ndata becomes an RFC. And SHOULD do so now
>> based on draft-stewart-tsvwg-sctp-ndata-03.
> [Raju] I am not sure if this NDATA is a deal breaker for webrtc 1.0?! =
Some implementations may support while others not as NDATA is negotiated =
during association setup; if one end does not support then it is =
disabled.
>=20
> If you don't have NDATA then one data channel stream may monopolize =
the SCTP link and could affect scenarios like:
> - You sent 1GB file using data channel stream 0 (all at once without =
using something like MSRP).
> - Then immediately, you send a small "Hi, some important stuff!" chat =
message using data channel stream 1.
> Without NDATA, the other end will receive the text msg ONLY AFTER =
receiving the 1GB file!! With NDATA chat msg gets equal share of the =
SCTP link and gets delivered much sooner.
If NDATA is not supported, the message size is limited to avoid the =
monopolization. This limit
will be signalled through SDP.

Best regards
Michael
> You can apply this to even more sensitive gaming apps.
>=20
> -Raju
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 00:25:41 2014
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On Feb 6, 2014, at 8:56 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>> As discussed in the last IETF meeting, this PPID based fragmentation =
and reassembly will be removed from the ID.
>=20
> I assume the associated PPID values will also be un-registered with =
IANA?
I guess they will stay... They are supported by Firefox and PPIDs are a =
pretty
cheap resource. They are from a pool of 32-bit entities. That is why
we registered the values before the RFCs get published.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Hi,

>>> As discussed in the last IETF meeting, this PPID based fragmentation an=
d reassembly will be removed from the ID.
>>=20
>> I assume the associated PPID values will also be un-registered with IANA=
?
> I guess they will stay... They are supported by Firefox and PPIDs are a p=
retty cheap resource. They are from a pool of 32-bit entities. That is why =
we registered the values before the RFCs get published.

So, does that mean my WebRTC entity has to support them, in case the remote=
 peer is on Firefox?

Also, if they are removed from the data channel draft, what reference will =
be used in the IANA table?

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] RTCWEB Data Channel: message interleaving
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Date: Thu, 6 Feb 2014 08:38:45 +0000
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Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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Hi,

> If NDATA is not supported, the message size is limited to avoid the monop=
olization. This limit will be signalled through SDP.

What do you mean by "signaled through SDP"?

Will the browser inform the JS app, using SDP, if the local and/or remote p=
eer doesn't support NDATA?

Regards,

Christer


From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 00:45:55 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 6, 2014, at 9:36 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>> As discussed in the last IETF meeting, this PPID based =
fragmentation and reassembly will be removed from the ID.
>>>=20
>>> I assume the associated PPID values will also be un-registered with =
IANA?
>> I guess they will stay... They are supported by Firefox and PPIDs are =
a pretty cheap resource. They are from a pool of 32-bit entities. That =
is why we registered the values before the RFCs get published.
>=20
> So, does that mean my WebRTC entity has to support them, in case the =
remote peer is on Firefox?
I don't think so. Not sure what the plan for Firefox is, but I expect =
Firefox to change the
code as soon as the final solution is specified.
>=20
> Also, if they are removed from the data channel draft, what reference =
will be used in the IANA table?
I think it will stay. For PPIDs you even don't have to have an RFC. You =
can even register PPIDs
as a person.
>=20
> Regards,
>=20
> Christer
>=20
>=20


From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 00:52:28 2014
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On Feb 6, 2014, at 9:38 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>> If NDATA is not supported, the message size is limited to avoid the =
monopolization. This limit will be signalled through SDP.
>=20
> What do you mean by "signaled through SDP"?
>=20
> Will the browser inform the JS app, using SDP, if the local and/or =
remote peer doesn't support NDATA?
What will be signalled in SDP is the maximum message size the end-point =
can handle. This is needed
anyway... If you don't support NDATA, you can signal a smaller value of, =
let's say 16KB or so.

NDATA is negotiated via the SCTP setup and there is no need for another =
signalling channel.

After the association is up, both sides know if NDATA is supported and, =
if I remember correctly,
there is a JS way of making message size limits available to the =
application. So you can
limit the message size if NDATA is not supported by both peers. So that =
is the plan:
* If NDATA is supported by both peers, message size are only limited by =
reassembly
  memory and any limit is signalled via SDP. The corresponding support =
will be in the
  next revision of the SDP ID.
* If NDATA is not supported by both peers, a message size limit applies =
to avoid monopolisation.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


From internet-drafts@ietf.org  Thu Feb  6 02:42:17 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-13.txt
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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           : Web Real-Time Communication Use-cases and Requirements
        Authors         : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-13.txt
	Pages           : 26
	Date            : 2014-02-06

Abstract:
   This document describes web based real-time communication use-cases.
   Requirements on the browser functionality are derived from the use-
   cases.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-13

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-use-cases-and-requirements-13


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From stefan.lk.hakansson@ericsson.com  Thu Feb  6 02:52:49 2014
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Subject: [rtcweb] New version of use-case draft uploaded
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Hi,=0A=
=0A=
I've uploaded a new version -13 [1]. I have made changes based on input=0A=
from Mary, Magnus and Chenxing+Andy, for details read further down.=0A=
=0A=
I hope I have caught most things; what remains to my knowledge is:=0A=
* Include the requirement text the first time a requirement is derived=0A=
* Group (and re-number) the claims=0A=
=0A=
I am hesitating to do the last thing 'cause of the risk of getting all=0A=
internal references into a mess, but one day soon I will feel brave=0A=
enough :-)=0A=
=0A=
Stefan=0A=
=0A=
[1]=0A=
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirement=
s/=0A=
=0A=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
On 2014-01-15 21:57, Mary Barnes wrote:=0A=
> I had thought I had previously done a detailed review of this doc, but=0A=
I can't find it to know whether changes suggested have been=0A=
incorporated. So, I have re-reviewed the document and I think it's=0A=
almost ready to progress. I think it needs some editorial clarifications=0A=
and nits to be fixed as summarized below. Note, I did not review the=0A=
appendix.=0A=
> Regards,=0A=
> Mary.=0A=
> In general, I still find the style of this document very difficult to=0A=
grok since the requirements are not grouped in categories and one has to=0A=
keep switching back and forth in the document to match requirements to=0A=
use cases.=0A=
> Questions/Comments for clarification:=0A=
> ----------------------------------------------------=0A=
> 1) Section 1, 1st paragraph, last sentence. It's not clear to me that=0A=
"e.g., a telephone" is meaningful. I don't think you're intending to=0A=
interworking with a legacy PSTN connected black phone. So, it might be=0A=
more accurate to say " e.g., a mobile phone or a SIP UA".=0A=
=0A=
Changed.=0A=
=0A=
> 2) Section 3.3.1.1. Next to last paragraph. I'm not sure what you mean=0A=
by different "makes". I think you mean different types of devices (e.g.,=0A=
mobile, SIP UA, etc. ). That all said, I don't think that's not so=0A=
relevant. I think simply stating different OSs and different browsers is=0A=
sufficient.=0A=
=0A=
Changed.=0A=
=0A=
> 3) Section 3.3.6.1. It's not at all clear to me why this requirement=0A=
is considered specific to WebRTC. I would think the access network=0A=
changes should be transparent to WebRTC. Certainly, the device needs to=0A=
know what's happening, but I think whether this works is entirely based=0A=
on the internals of the device and the specific access network=0A=
technology, and not WebRTC application.=0A=
=0A=
It is not WebRTC specific per se, but is important for the use-case.=0A=
What parts that solve it is of less importance IMO.=0A=
=0A=
> 4) Section 3.3.10.1. Why is F24 not considered an additional=0A=
requirement here? Also, do you not need to have a statement as to what=0A=
other use case is the basis for this one such that the core requirements=0A=
are reference?=0A=
=0A=
I added F24 (it belongs here). I don't think we need a statement=0A=
regarding the basis, it is covered in section 3.2=0A=
=0A=
> 5) Section 3.3.11.1, 2nd paragraph, 1st sentence. I don't understand=0A=
what this is trying to say. What is meant by "enhance intelligibility"?=0A=
And, what is meant by "pans the audio from different participants=0A=
differently when rendering the audio".=0A=
> [As an aside, I will note that some of the CLUE use cases likely=0A=
encompass what you are trying to communicate here in this requirements=0A=
(including subsequent paragraphs and the last "Note:"), so you may want=0A=
to look at those and use similar terminology and concepts, that CLUE=0A=
spent a lot of time developing. ]=0A=
=0A=
I updated (using similar terminology to Clue).=0A=
=0A=
> 6) Section 3.4. I would expect F27 to be referenced by at least one of=0A=
these use cases.=0A=
=0A=
I added it to 3.4.1=0A=
=0A=
> 7) Section 4.2.=0A=
> - General: I am a bit confused as there are requirements in this=0A=
section that aren't referenced in section 3, including F19, F23, and F27=0A=
. Perhaps, that's because there are some missing references in section 3=0A=
(see item 7)? If not, then why are they there. At a minimum you should=0A=
add a sentence to section 4.1 indicating that not all the requirements=0A=
are derived from the use cases (contrary to what is currently stated).=0A=
=0A=
F19 has been removed. I added ref to F23 (and F24!) from ues-case=0A=
"multi-party game with voice communication". Ref to F27 added to 3.4.1.=0A=
=0A=
> - What's the difference between F24 and F34?=0A=
=0A=
I don't know, and I removed F34=0A=
=0A=
> - F30. I had to read this several times to understand it due to=0A=
structure. I would suggest changing as follows:=0A=
> OLD:=0A=
> F30 The browser must be able to use the screen (or=0A=
> a specific area of the screen) or what a certain=0A=
> application displays on the screen to generate=0A=
> streams.=0A=
> NEW:=0A=
> F30 The browser must be able to generate streams using the entire user=0A=
display, a specific area of the user's display or the information being=0A=
displayed by a specific application.=0A=
=0A=
Updated.=0A=
=0A=
> On this one, I also think it would be good to clarify what type of=0A=
stream - are you talking about using protocol to share content or or is=0A=
this just a video stream? Or would you have two separate requirement to=0A=
cover both of these?=0A=
=0A=
I would like to avoid going into detail; the idea is that it is some=0A=
kind of "stream" that allows sharing what is rendered on your screen,=0A=
the requirement does not need to go into what kind it is.=0A=
=0A=
> - F32. I can't quite grok this one. Maybe you are trying to say=0A=
something like the following?=0A=
> OLD:=0A=
> F32 There browser must support that STUN and TURN=0A=
> servers to use are supplied by other entities=0A=
> than via the web application (i.e. the network=0A=
> provider).=0A=
> NEW:=0A=
> F32 The browser must support the use of STUN and TURN=0A=
> servers that are supplied by entities=0A=
> other than the web application (i.e. the network=0A=
> provider).=0A=
=0A=
Updated.=0A=
=0A=
> 8) Section 7. I have mixed feelings about leaving this list with URLs=0A=
in the document. I think it's good to highlight the use cases that=0A=
weren't incorporated and why they weren't. I think it would add a lot=0A=
more value to provide a concise summary of the reasons they weren't=0A=
added than just including links, in particular, since we usually don't=0A=
like to publish RFCs with links.=0A=
=0A=
I think the entire list should go before publication. The list has been=0A=
kept there as a reminder of what we included and what we did not=0A=
included. I have removed it in this version.=0A=
=0A=
> Nits:=0A=
> ------=0A=
> 1) Section 1, 1st paragraph, last sentence, "at least one of the=0A=
end-user client" -> "at least one of the end-user clients"=0A=
> 2) Section 3.2, 1st paragraph, 1st sentence:=0A=
> - "retrives" -> "retrieves"=0A=
> - add a section reference for "Simple Video Communication Service"=0A=
> 3) Section 3.2. , next to last sentence. "retrieved from" -> "derived=0A=
from"=0A=
> 4) Section 3.3.5.1, 3rd paragraph,=0A=
> - 1st sentence. "session" - "session"=0A=
> - 2nd sentence. "straddle the boundary between the internal network=0A=
and external." -> "straddles the boundary between the internal and=0A=
external networks.=0A=
> 5) Section 3.3.5.1, 4th paragraph. "they still want to have the=0A=
traffic to stay" -> "they still want the traffic to say"=0A=
> 6) Section 3.3.61. 1st paragraph. I'm not sure why this ends with ":"=0A=
> 7) Section 3.3.6.1, 2nd paragraph. "device used by one of the users=0A=
have several" -> "device used by one of the users has several"=0A=
> 8) Section 3.3.11.1, 1st para, 1st sentence. "In this use case is the=0A=
Simple..." -> "In this use case, the Simple...."=0A=
> 9) Section 3.3.11.1 3rd from last paragraph. "use experience" -> "user=0A=
experience"=0A=
> 10) Section 3.4.3.1, 2nd paragraph. "participant send" -> "participant=0A=
sends"=0A=
> 11) Section 4.2:=0A=
> - F35. "of that streams" -> "that streams"=0A=
=0A=
Most Nit's fixed=0A=
=0A=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
=0A=
=0A=
On 2014-01-29 10:29, Magnus Westerlund wrote:=0A=
> Hi Partha and WG,=0A=
>=0A=
> I don't see any support for the changes you proposes in this discussion.=
=0A=
> What I see some support for is to add a statement making clear that=0A=
> there might be additional NAT/Firewall traversal mechanisms than=0A=
> STUN/TURN. Simon proposed:=0A=
>=0A=
> "Note that TURN support being mandatory does not preclude a WebRTC=0A=
> endpoint from supporting additional traversal mechanisms."=0A=
=0A=
I added this to section 3.3.4.1=0A=
=0A=
>=0A=
>=0A=
> However, looking at the document as it is currently written, I am=0A=
> uncertain where this would be added. The first mention of TURN is in=0A=
> Section 3.3.4.1, and that section is focused on the global service=0A=
> provider perspective and the need for location based provisioning of=0A=
> NAT/Firewall traversal server resources.=0A=
>=0A=
> I think it can be added to Section 3.3.5.1 without being misplaced, but=
=0A=
> it would be given a slightly narrower scope.=0A=
>=0A=
> I any of you want to be more explicit where this should be included,=0A=
> please be. If you are not forthcoming I will request the authors to add=
=0A=
> this in what they consider sensible position.=0A=
>=0A=
> Cheers=0A=
>=0A=
> Magnus=0A=
>=0A=
>=0A=
> On 2014-01-25 17:46, Parthasarathi R wrote:=0A=
>> Hi Simon,=0A=
>>=0A=
>>=0A=
>>=0A=
>> Thanks for your understanding about my firewall/NAT related problem=0A=
>> statement in this draft.=0A=
>>=0A=
>>=0A=
>>=0A=
>> I have proposed the firewall/NAT related text by which the specific=0A=
>> mechanism is not highlighted in the requirement document as there is no=
=0A=
>> WG consensus for any of the mechanism including TURN. It is possible to=
=0A=
>> argue hypothetically in PNTAW alias that PCP is the only mechanism=0A=
>> required in WebRTC endpoint. Also, I=92m more interested in WebRTC=0A=
>> gateway/server (Sec 4.3. of=0A=
>> draft-ietf-rtcweb-use-cases-and-requirements-12) requirements wherein it=
=0A=
>> is not required to support TURN and the related mail thread is=0A=
>> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.=0A=
>>=0A=
>>=0A=
>>=0A=
>> IMO, my proposed text without mentioning any firewall/NAT mechanism in=
=0A=
>> the requirement document helps to move forward without depend on the=0A=
>> solution discussion in PNTAW alias.=0A=
>>=0A=
>>=0A=
>>=0A=
>> Thanks=0A=
>>=0A=
>> Partha=0A=
>>=0A=
>>=0A=
>>=0A=
>> *From:*Cb B [mailto:cb.list6@gmail.com]=0A=
>> *Sent:* Saturday, January 25, 2014 6:22 AM=0A=
>> *To:* Simon Perreault=0A=
>> *Cc:* rtcweb@ietf.org; Parthasarathi R=0A=
>> *Subject:* Re: [rtcweb] Query/Comment on=0A=
>> draft-ietf-rtcweb-use-cases-and-requirements-12=0A=
>>=0A=
>>=0A=
>>=0A=
>>=0A=
>> On Jan 24, 2014 10:17 AM, "Simon Perreault" <simon.perreault@viagenie.ca=
=0A=
>> <mailto:simon.perreault@viagenie.ca>> wrote:=0A=
>>>=0A=
>>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :=0A=
>>>=0A=
>>>> Please note that when non-IETFers read this requirement document,=0A=
>> they come=0A=
>>>> to the conclusion that IETF RTCWeb WG recommends TURN and not other=0A=
>>>> mechanisms. I'm saying that requirement document should not be used=0A=
>> as the=0A=
>>>> mechanism to eliminate the other alternatives when there is a=0A=
discussion=0A=
>>>> going-on in PNTAW alias. So, I'm asking for the change.=0A=
>>>=0A=
>>>=0A=
>>> I would totally agree with that sentiment, although I don't see your=0A=
>> proposed text change reflecting it accurately. How about simply:=0A=
>>>=0A=
>>> "Note that TURN support being mandatory does not preclude a WebRTC=0A=
>> endpoint from supporting additional traversal mechanisms."=0A=
>>>=0A=
>>>=0A=
>>=0A=
>> +1 for the above text.=0A=
>>=0A=
>> CB=0A=
>>=0A=
>>> Simon=0A=
>>> --=0A=
>>> DTN made easy, lean, and smart --> http://postellation.viagenie.ca=0A=
>>> NAT64/DNS64 open-source --> http://ecdysis.viagenie.ca=0A=
>>> STUN/TURN server --> http://numb.viagenie.ca=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=
>> rtcweb mailing list=0A=
>> rtcweb@ietf.org=0A=
>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>>=0A=
>=0A=
>=0A=
=0A=
=0A=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
=0A=
On 2014-01-16 15:18, Hutton, Andrew wrote:=0A=
> Some comments in the text below.=0A=
>=0A=
> As a general comment I suggest changing all the "FW" abbreviations to=0A=
"Firewall" are have at least a first use definition.=0A=
=0A=
Done=0A=
=0A=
>=0A=
>=0A=
> Andy=0A=
>=0A=
>=0A=
>> -----Original Message-----=0A=
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=0A=
>> Sent: 15 January 2014 10:20=0A=
>> To: Chenxin (Xin); Hutton, Andrew; Christer Holmberg; Parthasarathi R;=
=0A=
>> rtcweb@ietf.org=0A=
>> Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-and-=
=0A=
>> requirements-12=0A=
>>=0A=
>> WG,=0A=
>>=0A=
>> It has been quite some time since the WG last call ended and a new=0A=
>> revision was submitted. As Document Shepherd I want to push this=0A=
>> document to publication request.=0A=
>>=0A=
>> Chenxin proposed below three different sets of changes to the document.=
=0A=
>> Does the WG support making these changes? Please indicate within the=0A=
>> next week if you support or want to reject these changes.=0A=
>>=0A=
>> Thanks=0A=
>>=0A=
>> Magnus=0A=
>>=0A=
>>=0A=
>> On 2013-10-17 12:23, Chenxin (Xin) wrote:=0A=
>>> Hi Andy,=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> I think you means F29 not F27:). When I read it , I realize that=0A=
>> there=0A=
>>> is cross and ambiguous between 3.3.2 and 3.3.3=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> More details:=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> The topic of 3.3.2 is "Simple Video Communication Service, *NAT/FW*=0A=
>>> that blocks UDP". But in the description and requirement, only *NAT*=0A=
>> is=0A=
>>> considered.=0A=
>>>=0A=
>>> The topic of 3.3.3 is "Simple Video Communication Service, FW that=0A=
>>> only allows http", But only *http proxy* deployed scenarios is=0A=
>> considered.=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> There are other usecases " FW block UDP, incoming TCP, Http=0A=
>> allowing=0A=
>>> FW without http proxy deplolyed under the permission of FW policy" ,=0A=
>>> which is lost in the description. If we need consider these usecases=0A=
>> , I=0A=
>>> suggest to make some change to the description.=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> Proposal 1 :=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> add FW related words to section 3.3.2=0A=
>>>=0A=
>>> -------------------------------------------------=0A=
>>>=0A=
>>> 3.3.2. Simple Video Communication Service, NAT/FW that blocks UDP=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> 3.3.2.1. Description=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> This use-case is almost identical to the Simple Video=0A=
>> Communication=0A=
>>>=0A=
>>> Service use-case (Section 3.3.1). The difference is that one of=0A=
>> the=0A=
>>>=0A=
>>> users is behind a NAT*/FW* that blocks UDP traffic.=0A=
>>>=0A=
>>> .=0A=
>=0A=
> [AndyH] - I support this change.=0A=
=0A=
Changed=0A=
=0A=
>=0A=
>=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> 3.3.2.2. Additional Requirements=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> F29 The browser must be able to send streams and=0A=
>>>=0A=
>>> data to a peer in the presence of NATs *and FWs* that=0A=
>>>=0A=
>>> block UDP traffic ,* when FW policy allows WebRTC=0A=
>> traffic*.=0A=
>=0A=
> [AndyH] - I support this change.=0A=
=0A=
Changed=0A=
=0A=
>=0A=
>=0A=
>>>=0A=
>>> -------------------------------------------------------=0A=
>>>=0A=
>>> Proposal 2: If the" Http allowing FW without http proxy deployed"=0A=
>>> case is impliedly included in F29. I suggest to change the topics of=0A=
>>> 3.3.3 to "Simple Video Communication Service, FW that only allows=0A=
>>> traffic via a http proxy". So the 3.3.3 is a specific case.=0A=
>>>=0A=
>=0A=
> [AndyH] Yes the heading of 3.3.3 should have been changed during the=0A=
last edit to "Simple Video Communication Service, FW that only allows=0A=
traffic via a HTTP Proxy". I support this change.=0A=
=0A=
Changed.=0A=
=0A=
>=0A=
>=0A=
>>>=0A=
>>>=0A=
>>> Proposal 3: If " Http allowing FW without http proxy deployed"=0A=
>> case=0A=
>>> need to be explicitly mentioned. I suggest to add some descriptions=0A=
>> to 3.3.3=0A=
>>>=0A=
>>> -----------------------------------------------------------------=0A=
>>>=0A=
>>> 3.3.3. Simple Video Communication Service, FW that only allows http=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> 3.3.3.1. Description=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> This use-case is almost identical to the Simple Video=0A=
>> Communication=0A=
>>>=0A=
>>> Service use-case (Section 3.3.1). The difference is that one of=0A=
>> the=0A=
>>>=0A=
>>> users is behind a http allowing FW or a FW that only allows=0A=
>> traffic=0A=
>>> via a HTTP Proxy.=0A=
>>>=0A=
>>>=0A=
>=0A=
> [AndyH] I don't believe the intention here was to state a requirement=0A=
that WebRTC media should be able to flow through a FW only allowing=0A=
HTTP. Therefore I think the original description is ok it is just the=0A=
heading that needs to change.=0A=
>=0A=
>=0A=
>>>=0A=
>>> 3.3.3.2. Additional Requirements=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> F37 The browser must be able to send streams and=0A=
>>>=0A=
>>> data to a peer in the presence of http allowing FWs or FWs=0A=
>>> that only=0A=
>>>=0A=
>>> allows traffic via a HTTP Proxy, when FW policy=0A=
>>>=0A=
>>> allows WebRTC traffic.=0A=
>>>=0A=
>=0A=
> [AndyH] The existing text is in the 012 draft is ok and I don't think=0A=
this change is needed as again I don't believe the intention here was to=0A=
state a requirement that WebRTC media should be able to flow through a=0A=
FW only allowing HTTP.=0A=
>=0A=
>=0A=
>>> -------------------------------------------------------------------=0A=
>>>=0A=
>>>=0A=
>=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>> Best Regards,=0A=
>>>=0A=
>>> Xin=0A=
>>>=0A=
>>>=0A=
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>>=0A=
>>=0A=
>> --=0A=
>>=0A=
>> Magnus Westerlund=0A=
>>=0A=
>> ----------------------------------------------------------------------=
=0A=
>> Services, Media and Network features, Ericsson Research EAB/TXM=0A=
>> ----------------------------------------------------------------------=
=0A=
>> Ericsson AB | Phone +46 10 7148287=0A=
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From christer.holmberg@ericsson.com  Thu Feb  6 03:15:13 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Data channel: JavaScript usage of data channel without the Data Channel Protocol?
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Subject: [rtcweb] Data channel: JavaScript usage of data channel without the Data Channel Protocol?
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Hi,

As was indicated earlier, the usage of the Data Channel Protocol (DCP) is o=
ptional with the Data Channel.

However, if I understand the API correctly, only the DCP "open"/"ack" messa=
ges will trigger a JavaScript "open" event.

If so, how will my JS App be able to use the data channel if I do NOT use t=
he Data Channel Protocol? Can I use it without the "open" event, and/or is =
there something else that will trigger it?

Regards,

Christer

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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micr=
osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As was indicated earlier, the usage of the Data Chan=
nel Protocol (DCP) is optional with the Data Channel.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">However, if I understand the API correctly, only the=
 DCP &quot;open&quot;/&quot;ack&quot; messages will trigger a JavaScript &q=
uot;open&quot; event.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">If so, how will my JS App be able to use the data ch=
annel if I do NOT use the Data Channel Protocol? Can I use it without the &=
#8220;open&#8221; event, and/or is there something else that will trigger i=
t?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D15F1ACESESSMB209erics_--

From Raju.Makaraju@alcatel-lucent.com  Thu Feb  6 03:46:23 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Date: Thu, 6 Feb 2014 11:45:46 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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> >>> As discussed in the last IETF meeting, this PPID based fragmentation =
and
> reassembly will be removed from the ID.
> >>
> >> I assume the associated PPID values will also be un-registered with IA=
NA?
> > I guess they will stay... They are supported by Firefox and PPIDs are a
> pretty cheap resource. They are from a pool of 32-bit entities. That is w=
hy
> we registered the values before the RFCs get published.
>=20
> So, does that mean my WebRTC entity has to support them, in case the remo=
te
> peer is on Firefox?
>=20
> Also, if they are removed from the data channel draft, what reference wil=
l
> be used in the IANA table?

[Raju] Per my understanding some of the PPIDs are still needed.
The send() API for data channel comes in 4 flavors:
1.    void send (DOMString data);
2.    void send (Blob data);
3.    void send (ArrayBuffer data);
4.    void send (ArrayBufferView data);

1 maps to text data
2,3,4 maps to binary data

When one end sends this data one would expect that the other end gets an on=
message() callback with same (if other end is also javascript) or similar(i=
f other end is a native client) data structure. I believe this is achieved =
by using PPIDs.
We need at least 2 PPIDs: text and binary. So, existing "DOM String last" (=
=3D51) and "Binary Data Last"(=3D53) can be used.
To facilitate interworking with native clients I don't think there is a nee=
d to represent all sub-data types of send() in PPIDs as native client on th=
e other end using C/C++ won't know what "ArrayBufferView" is?!
Also, I think "DOMString" PPID is too specific to Javascript, instead it sh=
ould probably have a generic name like "UTF-16 String". The send API can st=
ill use DOMString as this is Javascript API.

-Raju

From christer.holmberg@ericsson.com  Thu Feb  6 03:50:46 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Michael Tuexen" <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Hi,

I agree that PPID values are still needed, and my question was not about re=
moving all of them - only the ones used with application-level fragmentatio=
n.

Regards,

Christer

-----Original Message-----
From: Makaraju, Maridi Raju (Raju) [mailto:Raju.Makaraju@alcatel-lucent.com=
]=20
Sent: 6. helmikuuta 2014 13:46
To: Christer Holmberg; Michael Tuexen
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multi=
plexing

> >>> As discussed in the last IETF meeting, this PPID based=20
> >>> fragmentation and
> reassembly will be removed from the ID.
> >>
> >> I assume the associated PPID values will also be un-registered with IA=
NA?
> > I guess they will stay... They are supported by Firefox and PPIDs=20
> > are a
> pretty cheap resource. They are from a pool of 32-bit entities. That=20
> is why we registered the values before the RFCs get published.
>=20
> So, does that mean my WebRTC entity has to support them, in case the=20
> remote peer is on Firefox?
>=20
> Also, if they are removed from the data channel draft, what reference=20
> will be used in the IANA table?

[Raju] Per my understanding some of the PPIDs are still needed.
The send() API for data channel comes in 4 flavors:
1.    void send (DOMString data);
2.    void send (Blob data);
3.    void send (ArrayBuffer data);
4.    void send (ArrayBufferView data);

1 maps to text data
2,3,4 maps to binary data

When one end sends this data one would expect that the other end gets an on=
message() callback with same (if other end is also javascript) or similar(i=
f other end is a native client) data structure. I believe this is achieved =
by using PPIDs.
We need at least 2 PPIDs: text and binary. So, existing "DOM String last" (=
=3D51) and "Binary Data Last"(=3D53) can be used.
To facilitate interworking with native clients I don't think there is a nee=
d to represent all sub-data types of send() in PPIDs as native client on th=
e other end using C/C++ won't know what "ArrayBufferView" is?!
Also, I think "DOMString" PPID is too specific to Javascript, instead it sh=
ould probably have a generic name like "UTF-16 String". The send API can st=
ill use DOMString as this is Javascript API.

-Raju

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To: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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>>>> Q1: Assuming that both endpoints want to use the data channel, do both
>endpoints need to send DATA_CHANNEL_OPEN
>>>> (on separate SCTP streams)? Or, can one of the endpoints, if it has re=
ceived
>DCO, start using the data channel (using a SCTP stream of its choice)?

[Raju] As explained by others already, DC is bidirectional, however both di=
rections have to use the same SCTP stream that was assigned for the data ch=
annel (which is done as part of DC OPEN an DC ACK). In fact the PeerConnect=
ion send() API does not allow stream # to be specified by the app, stream #=
 is internally tied to DC.

-Raju

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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Hi,

>>>>> Q1: Assuming that both endpoints want to use the data channel, do bot=
h
>>>>>endpoints need to send DATA_CHANNEL_OPEN
>>>>> (on separate SCTP streams)? Or, can one of the endpoints, if it has r=
eceived
>>>>>>DCO, start using the data channel (using a SCTP stream of its choice)=
?
>
> [Raju] As explained by others already, DC is bidirectional, however both =
directions have to use the same SCTP stream that was assigned for the data =
channel (which is done as part of DC OPEN an DC ACK).

When I send DC OPEN, do I know on which stream the DC ACK will arrive? If n=
ot, if there are DC OPEN/ACKs for multiple data channels, how is a DC ACK m=
apped to the associated DC OPEN? There is no "transaction id", or similar, =
in the messages.

> In fact the PeerConnection send() API does not allow stream # to be speci=
fied by the app, stream # is internally tied to DC.

My understanding is that it is possible to "manually" assign a stream id wh=
en creating the data channel. But, I may be wrong.

Regards,

Christer


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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] RTCWEB Data Channel: message interleaving
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>So you can
> limit the message size if NDATA is not supported by both peers. So that i=
s
> the plan:
> * If NDATA is supported by both peers, message size are only limited by
> reassembly
>   memory and any limit is signalled via SDP. The corresponding support wi=
ll
> be in the
>   next revision of the SDP ID.

[Raju] The key is "next revision of the SDP ID". It is understood that NDAT=
A support indication is only avilable after both the peers talk. But then, =
the initial message sizes were already conveyed (to make them talk). Now, i=
f the new message size per "no NDATA support"/"NDATA support" is lower/high=
er then another ound of end-to-end signaling is needed to update the peers.=
 Not optimal, but I do not see any other better way other than "starting co=
nservatively with lower values initially assuming no NDATA support; later w=
hen NDATA support becomes widespread default values can be higher".

Best Regards
Raju

From Raju.Makaraju@alcatel-lucent.com  Thu Feb  6 04:29:41 2014
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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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> > [Raju] As explained by others already, DC is bidirectional, however bot=
h
> directions have to use the same SCTP stream that was assigned for the dat=
a
> channel (which is done as part of DC OPEN an DC ACK).
>=20
> When I send DC OPEN, do I know on which stream the DC ACK will arrive? If
> not, if there are DC OPEN/ACKs for multiple data channels, how is a DC AC=
K
> mapped to the associated DC OPEN? There is no "transaction id", or simila=
r,
> in the messages.

[Raju]=20
http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section-6 say=
s stream id in ACK is same as the one in DC OPEN:
   If a DATA_CHANNEL_OPEN message is received on an unused stream, the
   stream identifier corresponds to the role of the peer and all
   parameters in the DATA_CHANNEL_OPEN message are valid, then a
   corresponding DATA_CHANNEL_ACK message is sent on the stream with the
   same stream identifier as the one the DATA_CHANNEL_OPEN message was
   received on.


> > In fact the PeerConnection send() API does not allow stream # to be
> specified by the app, stream # is internally tied to DC.
>=20
> My understanding is that it is possible to "manually" assign a stream id
> when creating the data channel. But, I may be wrong.
[Raju] Yes, it is possible.
For createDataChannel() API (http://dev.w3.org/2011/webrtc/editor/webrtc.ht=
ml#dom-peerconnection-createdatachannel) set 'negotiated' to 'true' and 'id=
' to the stread id you want. Then it is the responsibility of the applicati=
on to convey this info to the other end and the internal DC protcol (DC OPE=
N/ACK etc.) won't be used to manage the data channel/streams.
However, I am not sure if "negotiated=3Dfalse" and specifying "id" is a val=
id combination. For id the above spec says the following, so may be it is a=
llowed, but then user have to manage stream ids to avoid collisions:
	id of type unsigned short
		Overrides the default selection of id for this channel.

It is also interesting to note that the spec allows within the same SCTP li=
nk, some DCs can be negotiated using inband internal protocol while other D=
Cs can be negotiated using external out-of-band protocol! However, I think =
applications should try to avoid mixing these methods to avoid conflicts in=
 using stream ids.

-Raju

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>
Thread-Topic: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Hi,

>>> [Raju] As explained by others already, DC is bidirectional, however=20
>>> both directions have to use the same SCTP stream that was assigned for =
the=20
>>> data channel (which is done as part of DC OPEN an DC ACK).
>>>=20
>> When I send DC OPEN, do I know on which stream the DC ACK will arrive?=20
>> If not, if there are DC OPEN/ACKs for multiple data channels, how is a=20
>> DC ACK mapped to the associated DC OPEN? There is no "transaction id",=20
>> or similar, in the messages.
>
> [Raju]
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section-6 s=
ays stream id in ACK is same as the one in DC OPEN:
>   If a DATA_CHANNEL_OPEN message is received on an unused stream, the
>   stream identifier corresponds to the role of the peer and all
>   parameters in the DATA_CHANNEL_OPEN message are valid, then a
>   corresponding DATA_CHANNEL_ACK message is sent on the stream with the
>   same stream identifier as the one the DATA_CHANNEL_OPEN message was
>   received on.

And if the stream id is already used?=20

The "core" data channel draft does not mandate the usage of the same stream=
 id values in both directions.=20

Are you saying that, when using the DCP, one must use the same stream id va=
lue in both directions?

Regards,

Christer



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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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> > [Raju]
> > http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section-6
> says stream id in ACK is same as the one in DC OPEN:
> >   If a DATA_CHANNEL_OPEN message is received on an unused stream, the
> >   stream identifier corresponds to the role of the peer and all
> >   parameters in the DATA_CHANNEL_OPEN message are valid, then a
> >   corresponding DATA_CHANNEL_ACK message is sent on the stream with the
> >   same stream identifier as the one the DATA_CHANNEL_OPEN message was
> >   received on.
>=20
> And if the stream id is already used?
[Raju] DC protocol says:

   If a DATA_CHANNEL_OPEN message is received on an already used SCTP
   stream or there are any problems with parameters within the
   DATA_CHANNEL_OPEN message or the DATA_CHANNEL_OPEN message itself is
   not well-formed, the receiver MUST reset the corresponding outgoing
   SCTP stream using [RFC6525] and MUST NOT send a DATA_CHANNEL_ACK
   message in response to the received message.

>=20
> The "core" data channel draft does not mandate the usage of the same stre=
am
> id values in both directions.
>=20
> Are you saying that, when using the DCP, one must use the same stream id
> value in both directions?

[Raju] Correct.
When external DC negotiation is used, I expect most implementations to use =
same stream ids, but not a MUST, in both directions. This not only simplifi=
es the procedures but also aligns with internal DCP allocation of streams.

-Raju

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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Hi,

>>> [Raju]
>>> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#sectio
>>> n-6 says stream id in ACK is same as the one in DC OPEN:
>>>   If a DATA_CHANNEL_OPEN message is received on an unused stream, the
>>>   stream identifier corresponds to the role of the peer and all
>>>   parameters in the DATA_CHANNEL_OPEN message are valid, then a
>>>   corresponding DATA_CHANNEL_ACK message is sent on the stream with the
>>>   same stream identifier as the one the DATA_CHANNEL_OPEN message was
>>>   received on.
>>=20
>> And if the stream id is already used?
> [Raju] DC protocol says:
>
>   If a DATA_CHANNEL_OPEN message is received on an already used SCTP
>   stream or there are any problems with parameters within the
>   DATA_CHANNEL_OPEN message or the DATA_CHANNEL_OPEN message itself is
>   not well-formed, the receiver MUST reset the corresponding outgoing
>   SCTP stream using [RFC6525] and MUST NOT send a DATA_CHANNEL_ACK
>   message in response to the received message.
>=20
>> The "core" data channel draft does not mandate the usage of the same=20
>> stream id values in both directions.
>>=20
>> Are you saying that, when using the DCP, one must use the same stream=20
>> id value in both directions?
>
> [Raju] Correct.
> When external DC negotiation is used, I expect most implementations to us=
e same stream ids, but not a MUST, in both directions. This not only simpli=
fies the procedures but also aligns with internal DCP allocation of streams=
.

Is there a reason why we couldn't mandate usage of the same id in the core =
spec? Then the correlation would not be dependent on the negotiation mechan=
ism(s).

Regards,

Christer


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Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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On 2/6/14 3:52 AM, Michael Tuexen wrote:
> On Feb 6, 2014, at 9:38 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>> If NDATA is not supported, the message size is limited to avoid the monopolization. This limit will be signalled through SDP.
>>
>> What do you mean by "signaled through SDP"?
>>
>> Will the browser inform the JS app, using SDP, if the local and/or remote peer doesn't support NDATA?
> What will be signalled in SDP is the maximum message size the end-point can handle. This is needed
> anyway... If you don't support NDATA, you can signal a smaller value of, let's say 16KB or so.
>
> NDATA is negotiated via the SCTP setup and there is no need for another signalling channel.
>
> After the association is up, both sides know if NDATA is supported and, if I remember correctly,
> there is a JS way of making message size limits available to the application. So you can
> limit the message size if NDATA is not supported by both peers. So that is the plan:
> * If NDATA is supported by both peers, message size are only limited by reassembly
>    memory and any limit is signalled via SDP. The corresponding support will be in the
>    next revision of the SDP ID.
> * If NDATA is not supported by both peers, a message size limit applies to avoid monopolisation.

I assume that in the RFC this will just be a MUST NOT. Will that apply 
to the SCTP implementation? Or to the SCTP user?

(In a practical sense, will this be implemented in the SCTP stack, in 
the DataChannel "stack", in the browser, or in javascript?)

	Thanks,
	Paul


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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On 2/6/14 3:45 AM, Michael Tuexen wrote:
> On Feb 6, 2014, at 9:36 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>>>> As discussed in the last IETF meeting, this PPID based fragmentation and reassembly will be removed from the ID.
>>>>
>>>> I assume the associated PPID values will also be un-registered with IANA?
>>> I guess they will stay... They are supported by Firefox and PPIDs are a pretty cheap resource. They are from a pool of 32-bit entities. That is why we registered the values before the RFCs get published.
>>
>> So, does that mean my WebRTC entity has to support them, in case the remote peer is on Firefox?
> I don't think so. Not sure what the plan for Firefox is, but I expect Firefox to change the
> code as soon as the final solution is specified.
>>
>> Also, if they are removed from the data channel draft, what reference will be used in the IANA table?
> I think it will stay. For PPIDs you even don't have to have an RFC. You can even register PPIDs
> as a person.

I hope that if values are retained simply for backward compatibility 
they are explicitly marked as deprecated.

	Thanks,
	Paul


From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 11:17:45 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 6, 2014, at 12:45 PM, "Makaraju, Maridi Raju (Raju)" =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>>>>> As discussed in the last IETF meeting, this PPID based =
fragmentation and
>> reassembly will be removed from the ID.
>>>>=20
>>>> I assume the associated PPID values will also be un-registered with =
IANA?
>>> I guess they will stay... They are supported by Firefox and PPIDs =
are a
>> pretty cheap resource. They are from a pool of 32-bit entities. That =
is why
>> we registered the values before the RFCs get published.
>>=20
>> So, does that mean my WebRTC entity has to support them, in case the =
remote
>> peer is on Firefox?
>>=20
>> Also, if they are removed from the data channel draft, what reference =
will
>> be used in the IANA table?
>=20
> [Raju] Per my understanding some of the PPIDs are still needed.
> The send() API for data channel comes in 4 flavors:
> 1.    void send (DOMString data);
> 2.    void send (Blob data);
> 3.    void send (ArrayBuffer data);
> 4.    void send (ArrayBufferView data);
>=20
> 1 maps to text data
> 2,3,4 maps to binary data
What is still be needed:
WebRTC Control: Used for the DATA_CHANNEL_OPEN and DATA_CHANNEL_ACK =
message
DOMString Last: Used for 1.
Binary Data Last: Used for 2, 3, 4.

The PPIDs 'Binary Data Partial' and 'DOMString Partial' won't be used =
anymore.
These were specific to the PPID based fragmentation and reassembly.
>=20
> When one end sends this data one would expect that the other end gets =
an onmessage() callback with same (if other end is also javascript) or =
similar(if other end is a native client) data structure. I believe this =
is achieved by using PPIDs.
> We need at least 2 PPIDs: text and binary. So, existing "DOM String =
last" (=3D51) and "Binary Data Last"(=3D53) can be used.
> To facilitate interworking with native clients I don't think there is =
a need to represent all sub-data types of send() in PPIDs as native =
client on the other end using C/C++ won't know what "ArrayBufferView" =
is?!
At least there are no PPIDs for them right now.
> Also, I think "DOMString" PPID is too specific to Javascript, instead =
it should probably have a generic name like "UTF-16 String". The send =
API can still use DOMString as this is Javascript API.
Any comments from others?

Best regards
Michael
>=20
> -Raju
>=20


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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 6, 2014, at 5:26 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 2/6/14 3:45 AM, Michael Tuexen wrote:
>> On Feb 6, 2014, at 9:36 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>=20
>>> Hi,
>>>=20
>>>>>> As discussed in the last IETF meeting, this PPID based =
fragmentation and reassembly will be removed from the ID.
>>>>>=20
>>>>> I assume the associated PPID values will also be un-registered =
with IANA?
>>>> I guess they will stay... They are supported by Firefox and PPIDs =
are a pretty cheap resource. They are from a pool of 32-bit entities. =
That is why we registered the values before the RFCs get published.
>>>=20
>>> So, does that mean my WebRTC entity has to support them, in case the =
remote peer is on Firefox?
>> I don't think so. Not sure what the plan for Firefox is, but I expect =
Firefox to change the
>> code as soon as the final solution is specified.
>>>=20
>>> Also, if they are removed from the data channel draft, what =
reference will be used in the IANA table?
>> I think it will stay. For PPIDs you even don't have to have an RFC. =
You can even register PPIDs
>> as a person.
>=20
> I hope that if values are retained simply for backward compatibility =
they are explicitly marked as deprecated.
My plan was just not to defined what they are used for...

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From christer.holmberg@ericsson.com  Thu Feb  6 11:35:28 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Hi,

>> Also, I think "DOMString" PPID is too specific to Javascript, instead it=
 should probably have a generic name like "UTF-16 String". The send API can=
 still use DOMString as this is Javascript API.
> Any comments from others?

I agree that it would be good to use something more generic (e.g. UTF-16 St=
ring), as not all entities using this PPID value will be JavaScript based.

Regards,

Christer



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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol	multiplexing
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Hi,

>> I hope that if values are retained simply for backward compatibility the=
y are explicitly marked as deprecated.
> My plan was just not to defined what they are used for...

I would like to have some explanation text. Having registered values withou=
t any kind of explanation is not a good approach, in my opinion.

Regards,

Christer

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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>
Thread-Topic: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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> Is there a reason why we couldn't mandate usage of the same id in the cor=
e
> spec? Then the correlation would not be dependent on the negotiation
> mechanism(s).

[Raju] WebRTC does not provide any rules for external negotiation of data c=
hannels, so it should leave all the procedures, including stream id managem=
ent, outside the scope. Having "symmetric stream id" restriction puts unnec=
essary hurdle on external negotiation procedures, which adds no value overa=
ll. Like an example app may not care which data channel/stream-id is used f=
or send or receive, it just wants to know data is received and sent (the da=
ta itself may have some identification info).
=20
Btw, the SDP based external negotiation draft http://tools.ietf.org/id/draf=
t-ejzak-dispatch-webrtc-data-channel-sdpneg-00.txt is already proposing to =
use symmetric stream ids. Other external negotiation protocols may put such=
 restriction as they fit.

-Raju


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB Data Channel: message interleaving
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Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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> > * If NDATA is not supported by both peers, a message size limit applies=
 to
> avoid monopolisation.
>=20
> I assume that in the RFC this will just be a MUST NOT. Will that apply
> to the SCTP implementation? Or to the SCTP user?
>=20
> (In a practical sense, will this be implemented in the SCTP stack, in
> the DataChannel "stack", in the browser, or in javascript?)

[Raju] It has to be done exclusively by the application (Javascript for bro=
wser or some other language for native apps).
This is because per recent comments the browser/data-channel level fragment=
ation/reassembly, based on PPID, will be removed from data channel core spe=
c.

-Raju

From christer.holmberg@ericsson.com  Thu Feb  6 12:26:32 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>
Thread-Topic: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Subject: Re: [rtcweb] RTCWEB data channel protocol: Do both endpoints need to send DATA_CHANNEL_OPEN?
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Hi,

>> Is there a reason why we couldn't mandate usage of the same id in the co=
re
>> spec? Then the correlation would not be dependent on the negotiation
>> mechanism(s).
>
> [Raju] WebRTC does not provide any rules for external negotiation of data=
 channels, so it should leave all the procedures, including stream id=20
> management, outside the scope. Having "symmetric stream id" restriction p=
uts unnecessary hurdle on external negotiation procedures, which=20
> adds no value overall. Like an example app may not care which data channe=
l/stream-id is used for send or receive, it just wants to know data=20
> is received and sent (the data itself may have some identification info).
>
> Btw, the SDP based external negotiation draft http://tools.ietf.org/id/dr=
aft-ejzak-dispatch-webrtc-data-channel-sdpneg-00.txt is already proposing t=
o use symmetric stream ids. Other external negotiation protocols may put su=
ch restriction as they fit.

So, as far as the data channel mechanism itself is concerned, we do define =
that it consists of two uni-directional SCTP streams, but we don't define h=
ow those streams are associated with the data channel. For that some extern=
al mechanism is needed.

Regards,

Christer


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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 6, 2014, at 8:37 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
> Hi,
>=20
>>> I hope that if values are retained simply for backward compatibility =
they are explicitly marked as deprecated.
>> My plan was just not to defined what they are used for...
>=20
> I would like to have some explanation text. Having registered values =
without any kind of explanation is not a good approach, in my opinion.
The reason why I registered them is that they are used by Firefox and =
there was
the possibility that the PPID based fragmentation and reassembly got
standardised. I wanted to avoid the case that Firefox uses some values
and other protocols register these values. That would have been pretty
bad. Since they are a very cheap resource, I just went ahead and "saved"
the values.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20


From christer.holmberg@ericsson.com  Thu Feb  6 12:35:17 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Hi,

>>>> I hope that if values are retained simply for backward compatibility t=
hey are explicitly marked as deprecated.
>>> My plan was just not to defined what they are used for...
>>
>> I would like to have some explanation text. Having registered values wit=
hout any kind of explanation is not a good approach, in my opinion.
> The reason why I registered them is that they are used by Firefox and the=
re was
> the possibility that the PPID based fragmentation and reassembly got
> standardised. I wanted to avoid the case that Firefox uses some values
> and other protocols register these values. That would have been pretty
> bad. Since they are a very cheap resource, I just went ahead and "saved"
> the values.

That is a all good. But, when people wonder 5 years from now, I don't want =
them to have to look for this e-mail in oder to get an explanation :)

Regards,

Christer


From juberti@gmail.com  Thu Feb  6 11:56:16 2014
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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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--047d7b670311f6a51804f1c243cd
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+rtcweb

WebRTC endpoints must implement full ICE. Legacy (non-WebRTC) endpoints can
use ICE Lite, but as mentioned in RFC5245, S 2.7, full ICE is preferred.


On Tue, Jan 21, 2014 at 10:50 AM, Rachel Dvori <rachel.dvori@gmail.com>wrote:

> draft-ietf-rtcweb-jsep@tools.ietf.org
>
>
>
>
>
> Dear Person,
>
>
>
> I have seen the below in the draft of:
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05
>
>
>
> Attributes other than the ones specified above MAY be included,
>
> except for the following attributes which are specifically
>
> incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
>
> and *MUST NOT* be included:
>
>
>
> o  "a=crypto"
>
> o  "a=key-mgmt"
>
> o  "*a=ice-lite*"
>
>
>
>
>
> But in draft of :
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-11
>
> There is no phrase that prevents a WebRTC user to be ice-lite
>
>
>
>
>
> Can you pls clarify:
>
> -          Where is it mentioned that ice-lite cannot be one of the
> participants of WebRTC
>
> -          What is the reason that no side can be ice-lite  (as if one
> side is full-ice, there is still STUN and connectivity checks etc.
>
>
>
> Thanks
>
> RD
>
>
>
> p.s. - what is the correct email address for this type of questions ?
>  (Thanks and sorry)
>
>
>

--047d7b670311f6a51804f1c243cd
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>+rtcweb</div><div><br></div>WebRTC endpoints must imp=
lement full ICE. Legacy (non-WebRTC) endpoints can use ICE Lite, but as men=
tioned in RFC5245, S 2.7, full ICE is preferred.</div><div class=3D"gmail_e=
xtra">

<br><br><div class=3D"gmail_quote">On Tue, Jan 21, 2014 at 10:50 AM, Rachel=
 Dvori <span dir=3D"ltr">&lt;<a href=3D"mailto:rachel.dvori@gmail.com" targ=
et=3D"_blank">rachel.dvori@gmail.com</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

<div dir=3D"rtl"><p class=3D"MsoNormal" style=3D"direction:ltr"><a href=3D"=
mailto:draft-ietf-rtcweb-jsep@tools.ietf.org" target=3D"_blank">draft-ietf-=
rtcweb-jsep@tools.ietf.org</a></p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">Dear Person,</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">I have seen the below in the=
 draft of:=A0 =A0<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-js=
ep-05" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-=
05</a></p>



<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">Attributes other than the on=
es specified above MAY be
included,</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">except for the following att=
ributes which are specifically</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">incompatible with the requir=
ements of
[I-D.ietf-rtcweb-rtp-usage],</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">and <b>MUST NOT</b> be inclu=
ded:</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">o=A0
&quot;a=3Dcrypto&quot;</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">o=A0
&quot;a=3Dkey-mgmt&quot;</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">o=A0
&quot;<b>a=3Dice-lite</b>&quot;</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">But in draft of : =A0<a href=
=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-11" target=3D"_b=
lank">http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-11</a>=A0 </p>

<p class=3D"MsoNormal" style=3D"direction:ltr">There is no phrase that prev=
ents a WebRTC user to be
ice-lite</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">Can you pls clarify:</p>

<p style=3D"direction:ltr">-<span style=3D"font-size:7pt;font-family:&#39;T=
imes New Roman&#39;">=A0=A0=A0=A0=A0=A0=A0=A0=A0
</span><span dir=3D"LTR"></span>Where is it mentioned that
ice-lite cannot be one of the participants of WebRTC</p>

<p style=3D"direction:ltr">-<span style=3D"font-size:7pt;font-family:&#39;T=
imes New Roman&#39;">=A0=A0=A0=A0=A0=A0=A0=A0=A0
</span><span dir=3D"LTR"></span>What is the reason that no
side can be ice-lite=A0 (as if one side is
full-ice, there is still STUN and connectivity checks etc.</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p>

<p class=3D"MsoNormal" style=3D"direction:ltr">Thanks</p><span class=3D"HOE=
nZb"><font color=3D"#888888">

<p class=3D"MsoNormal" style=3D"direction:ltr">RD</p>

</font></span><p class=3D"MsoNormal" style=3D"direction:ltr">=A0</p><p clas=
s=3D"MsoNormal" dir=3D"ltr">p.s. - what is the correct email address for th=
is type of questions ? =A0(Thanks and sorry)</p><p class=3D"MsoNormal" dir=
=3D"ltr"><br>

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

--047d7b670311f6a51804f1c243cd--

From Michael.Tuexen@lurchi.franken.de  Thu Feb  6 12:47:15 2014
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References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <4CE22DF8-DE34-400A-95AE-2E011828DE88@lurchi.franken.de> <52F3B7A5.6070105@alum.mit.edu>, <3B7B334C-868A-4128-B7BF-07C66B649443@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D160930@ESESSMB209.ericsson.se>, <D5AE9FDB-77D9-422A-B995-C4BF299D9DE0@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D160A98@ESESSMB209.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On Feb 6, 2014, at 9:35 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
> Hi,
>=20
>>>>> I hope that if values are retained simply for backward =
compatibility they are explicitly marked as deprecated.
>>>> My plan was just not to defined what they are used for...
>>>=20
>>> I would like to have some explanation text. Having registered values =
without any kind of explanation is not a good approach, in my opinion.
>> The reason why I registered them is that they are used by Firefox and =
there was
>> the possibility that the PPID based fragmentation and reassembly got
>> standardised. I wanted to avoid the case that Firefox uses some =
values
>> and other protocols register these values. That would have been =
pretty
>> bad. Since they are a very cheap resource, I just went ahead and =
"saved"
>> the values.
>=20
> That is a all good. But, when people wonder 5 years from now, I don't =
want them to have to look for this e-mail in oder to get an explanation =
:)
Understood. But do you think they walk through the IANA registry? I =
would expect them
to read RFCs and follow the procedures described there.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


From harald@alvestrand.no  Thu Feb  6 15:18:14 2014
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Subject: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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On 02/06/2014 08:17 PM, Michael Tuexen wrote:
>> Also, I think "DOMString" PPID is too specific to Javascript, instead it should probably have a generic name like "UTF-16 String". The send API can still use DOMString as this is Javascript API.
> Any comments from others?

Note: The WebSockets protocol defines the transferred strings as UTF-8.
http://tools.ietf.org/html/rfc6455#section-5.6

As far as I can tell, we've always intended to follow that example.

The fact that Javascript implementations currently choose to represent
text strings as UTF-16 at their API is sad, but not an argument for
sending that particular text representation over the wire, or reflecting
the name in the API.


-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On 02/06/2014 09:47 PM, Michael Tuexen wrote:
> On Feb 6, 2014, at 9:35 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>>>>> I hope that if values are retained simply for backward compatibility they are explicitly marked as deprecated.
>>>>> My plan was just not to defined what they are used for...
>>>> I would like to have some explanation text. Having registered values without any kind of explanation is not a good approach, in my opinion.
>>> The reason why I registered them is that they are used by Firefox and there was
>>> the possibility that the PPID based fragmentation and reassembly got
>>> standardised. I wanted to avoid the case that Firefox uses some values
>>> and other protocols register these values. That would have been pretty
>>> bad. Since they are a very cheap resource, I just went ahead and "saved"
>>> the values.
>> That is a all good. But, when people wonder 5 years from now, I don't want them to have to look for this e-mail in oder to get an explanation :)
> Understood. But do you think they walk through the IANA registry? I would expect them
> to read RFCs and follow the procedures described there.

Updating the registry to read "Obsolete; was used by <draft name>"
should satisfy all comers.

People do read registries.


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From: "Wilson Chen (weichen2)" <weichen2@cisco.com>
To: Justin Uberti <justin@uberti.name>, Rachel Dvori <rachel.dvori@gmail.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
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From harald@alvestrand.no  Thu Feb  6 17:59:21 2014
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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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On 02/07/2014 02:56 AM, Wilson Chen (weichen2) wrote:
> In case of negotiating with server (may be a gateway), IMHO, the server would like not to do connectivity check, it is simpler and faster to work in ice-lite and passive mode.

The server doesn't have to claim WebRTC conformance, it just has to
claim to interoperate with WebRTC endpoints.

>
> Thanks,
> Wilson Chen
>
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin Uberti
> Sent: 2014å¹´2æœˆ7æ—¥ 3:56
> To: Rachel Dvori; rtcweb@ietf.org
> Subject: Re: [rtcweb] ice-lite in WebRTC ???
>
> +rtcweb
>
> WebRTC endpoints must implement full ICE. Legacy (non-WebRTC) endpoints can use ICE Lite, but as mentioned in RFC5245, S 2.7, full ICE is preferred.
>
> On Tue, Jan 21, 2014 at 10:50 AM, Rachel Dvori <rachel.dvori@gmail.com> wrote:
> draft-ietf-rtcweb-jsep@tools.ietf.org
>  
>  
> Dear Person,
>  
> I have seen the below in the draft of:   http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05
>  
> Attributes other than the ones specified above MAY be included,
> except for the following attributes which are specifically
> incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
> and MUST NOT be included:
>  
> o  "a=crypto"
> o  "a=key-mgmt"
> o  "a=ice-lite"
>  
>  
> But in draft of :  http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-11  
> There is no phrase that prevents a WebRTC user to be ice-lite
>  
>  
> Can you pls clarify:
> -          Where is it mentioned that ice-lite cannot be one of the participants of WebRTC
> -          What is the reason that no side can be ice-lite  (as if one side is full-ice, there is still STUN and connectivity checks etc.
>  
> Thanks
> RD
>  
> p.s. - what is the correct email address for this type of questions ?  (Thanks and sorry)
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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From: Justin Uberti <juberti@google.com>
Date: Thu, 6 Feb 2014 22:40:35 -0800
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWEB Data Channel: message interleaving
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--089e01183a009d495704f1cb45e0
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The browser will enforce the message limit to ensure predictable behavior.


On Thu, Feb 6, 2014 at 12:15 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> > > * If NDATA is not supported by both peers, a message size limit
> applies to
> > avoid monopolisation.
> >
> > I assume that in the RFC this will just be a MUST NOT. Will that apply
> > to the SCTP implementation? Or to the SCTP user?
> >
> > (In a practical sense, will this be implemented in the SCTP stack, in
> > the DataChannel "stack", in the browser, or in javascript?)
>
> [Raju] It has to be done exclusively by the application (Javascript for
> browser or some other language for native apps).
> This is because per recent comments the browser/data-channel level
> fragmentation/reassembly, based on PPID, will be removed from data channel
> core spec.
>
> -Raju
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e01183a009d495704f1cb45e0
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<div dir=3D"ltr">The browser will enforce the message limit to ensure predi=
ctable behavior.</div><div class=3D"gmail_extra"><br><br><div class=3D"gmai=
l_quote">On Thu, Feb 6, 2014 at 12:15 PM, Makaraju, Maridi Raju (Raju) <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" targe=
t=3D"_blank">Raju.Makaraju@alcatel-lucent.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">&gt; &gt; * If NDATA is no=
t supported by both peers, a message size limit applies to<br>
&gt; avoid monopolisation.<br>
&gt;<br>
&gt; I assume that in the RFC this will just be a MUST NOT. Will that apply=
<br>
&gt; to the SCTP implementation? Or to the SCTP user?<br>
&gt;<br>
&gt; (In a practical sense, will this be implemented in the SCTP stack, in<=
br>
&gt; the DataChannel &quot;stack&quot;, in the browser, or in javascript?)<=
br>
<br>
</div>[Raju] It has to be done exclusively by the application (Javascript f=
or browser or some other language for native apps).<br>
This is because per recent comments the browser/data-channel level fragment=
ation/reassembly, based on PPID, will be removed from data channel core spe=
c.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
-Raju<br>
</font></span><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></div>

--089e01183a009d495704f1cb45e0--

From christer.holmberg@ericsson.com  Thu Feb  6 23:27:52 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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Date: Fri, 7 Feb 2014 07:27:47 +0000
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol	multiplexing
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>>>>>>> I hope that if values are retained simply for backward compatibilit=
y they are explicitly marked as deprecated.
>>>>>> My plan was just not to defined what they are used for...
>>>>> I would like to have some explanation text. Having registered values =
without any kind of explanation is not a good approach, in my opinion.
>>>> The reason why I registered them is that they are used by Firefox=20
>>>> and there was the possibility that the PPID based fragmentation and=20
>>>> reassembly got standardised. I wanted to avoid the case that Firefox=20
>>>> uses some values and other protocols register these values. That=20
>>>> would have been pretty bad. Since they are a very cheap resource, I ju=
st went ahead and "saved"
>>>> the values.
>>> That is a all good. But, when people wonder 5 years from now, I don't=20
>>> want them to have to look for this e-mail in oder to get an=20
>>> explanation :)
>> Understood. But do you think they walk through the IANA registry? I=20
>> would expect them to read RFCs and follow the procedures described there=
.
>
> Updating the registry to read "Obsolete; was used by <draft name>"
> should satisfy all comers.

+ 1

Regards,

Christer


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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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<snip> > The server doesn't have to claim WebRTC conformance, it just =
has to
> claim to interoperate with WebRTC endpoints. </snip>

The above assumption leads to lot of confusion in WebRTC Server =
implementation which starts from RTCWeb requirement itself. IMO, it is =
better to provide the minimum IETF standard guidelines for WebRTC =
servers like ice-lite/ICE, BUNDLE required or not, ICE-TCP/TURN.=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
> Sent: Friday, February 07, 2014 7:29 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] ice-lite in WebRTC ???
>=20
> On 02/07/2014 02:56 AM, Wilson Chen (weichen2) wrote:
> > In case of negotiating with server (may be a gateway), IMHO, the
> server would like not to do connectivity check, it is simpler and
> faster to work in ice-lite and passive mode.
>=20
> The server doesn't have to claim WebRTC conformance, it just has to
> claim to interoperate with WebRTC endpoints.
>=20
> >
> > Thanks,
> > Wilson Chen
> >
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin
> Uberti
> > Sent: 2014=E5=B9=B42=E6=9C=887=E6=97=A5 3:56
> > To: Rachel Dvori; rtcweb@ietf.org
> > Subject: Re: [rtcweb] ice-lite in WebRTC ???
> >
> > +rtcweb
> >
> > WebRTC endpoints must implement full ICE. Legacy (non-WebRTC)
> endpoints can use ICE Lite, but as mentioned in RFC5245, S 2.7, full
> ICE is preferred.
> >
> > On Tue, Jan 21, 2014 at 10:50 AM, Rachel Dvori
> <rachel.dvori@gmail.com> wrote:
> > draft-ietf-rtcweb-jsep@tools.ietf.org
> >
> >
> > Dear Person,
> >
> > I have seen the below in the draft of:
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05
> >
> > Attributes other than the ones specified above MAY be included,
> > except for the following attributes which are specifically
> > incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
> > and MUST NOT be included:
> >
> > o  "a=3Dcrypto"
> > o  "a=3Dkey-mgmt"
> > o  "a=3Dice-lite"
> >
> >
> > But in draft of :  http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-
> usage-11
> > There is no phrase that prevents a WebRTC user to be ice-lite
> >
> >
> > Can you pls clarify:
> > -          Where is it mentioned that ice-lite cannot be one of the
> participants of WebRTC
> > -          What is the reason that no side can be ice-lite  (as if
> one side is full-ice, there is still STUN and connectivity checks etc.
> >
> > Thanks
> > RD
> >
> > p.s. - what is the correct email address for this type of questions =
?
> (Thanks and sorry)
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --
> Surveillance is pervasive. Go Dark.
>=20
> _______________________________________________
> 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: =?iso-8859-1?Q?'Stefan_H=E5kansson_LK'?= <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Stefan,

Simon proposed the following text after the lengthy mailing discussion:

> "Note that TURN support being mandatory does not preclude a WebRTC=20
> endpoint from supporting additional traversal mechanisms."

Whereas Sec 3.3.4.1 updated as=20

"Note that ICE support being mandatory does not preclude a WebRTC
   endpoint from supporting additional traversal mechanisms."

Apart from this, Tiru
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg11246.html) and
myself =
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg11241.html)
mentioned in the different mail thread that Sec 3.3.4.1 is not only the
right place to put the above text. Could you please correct the F31, F32
requirement text in the next revision of the draft.

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan
> H=E5kansson LK
> Sent: Thursday, February 06, 2014 4:23 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] New version of use-case draft uploaded
>=20
> Hi,
>=20
> I've uploaded a new version -13 [1]. I have made changes based on =
input
> from Mary, Magnus and Chenxing+Andy, for details read further down.
>=20
> I hope I have caught most things; what remains to my knowledge is:
> * Include the requirement text the first time a requirement is derived
> * Group (and re-number) the claims
>=20
> I am hesitating to do the last thing 'cause of the risk of getting all
> internal references into a mess, but one day soon I will feel brave
> enough :-)
>=20
> Stefan
>=20
> [1]
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-
> requirements/
>=20
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> On 2014-01-15 21:57, Mary Barnes wrote:
> > I had thought I had previously done a detailed review of this doc,
> but
> I can't find it to know whether changes suggested have been
> incorporated. So, I have re-reviewed the document and I think it's
> almost ready to progress. I think it needs some editorial
> clarifications
> and nits to be fixed as summarized below. Note, I did not review the
> appendix.
> > Regards,
> > Mary.
> > In general, I still find the style of this document very difficult =
to
> grok since the requirements are not grouped in categories and one has
> to
> keep switching back and forth in the document to match requirements to
> use cases.
> > Questions/Comments for clarification:
> > ----------------------------------------------------
> > 1) Section 1, 1st paragraph, last sentence. It's not clear to me =
that
> "e.g., a telephone" is meaningful. I don't think you're intending to
> interworking with a legacy PSTN connected black phone. So, it might be
> more accurate to say " e.g., a mobile phone or a SIP UA".
>=20
> Changed.
>=20
> > 2) Section 3.3.1.1. Next to last paragraph. I'm not sure what you
> mean
> by different "makes". I think you mean different types of devices
> (e.g.,
> mobile, SIP UA, etc. ). That all said, I don't think that's not so
> relevant. I think simply stating different OSs and different browsers
> is
> sufficient.
>=20
> Changed.
>=20
> > 3) Section 3.3.6.1. It's not at all clear to me why this requirement
> is considered specific to WebRTC. I would think the access network
> changes should be transparent to WebRTC. Certainly, the device needs =
to
> know what's happening, but I think whether this works is entirely =
based
> on the internals of the device and the specific access network
> technology, and not WebRTC application.
>=20
> It is not WebRTC specific per se, but is important for the use-case.
> What parts that solve it is of less importance IMO.
>=20
> > 4) Section 3.3.10.1. Why is F24 not considered an additional
> requirement here? Also, do you not need to have a statement as to what
> other use case is the basis for this one such that the core
> requirements
> are reference?
>=20
> I added F24 (it belongs here). I don't think we need a statement
> regarding the basis, it is covered in section 3.2
>=20
> > 5) Section 3.3.11.1, 2nd paragraph, 1st sentence. I don't understand
> what this is trying to say. What is meant by "enhance =
intelligibility"?
> And, what is meant by "pans the audio from different participants
> differently when rendering the audio".
> > [As an aside, I will note that some of the CLUE use cases likely
> encompass what you are trying to communicate here in this requirements
> (including subsequent paragraphs and the last "Note:"), so you may =
want
> to look at those and use similar terminology and concepts, that CLUE
> spent a lot of time developing. ]
>=20
> I updated (using similar terminology to Clue).
>=20
> > 6) Section 3.4. I would expect F27 to be referenced by at least one
> of
> these use cases.
>=20
> I added it to 3.4.1
>=20
> > 7) Section 4.2.
> > - General: I am a bit confused as there are requirements in this
> section that aren't referenced in section 3, including F19, F23, and
> F27
> . Perhaps, that's because there are some missing references in section
> 3
> (see item 7)? If not, then why are they there. At a minimum you should
> add a sentence to section 4.1 indicating that not all the requirements
> are derived from the use cases (contrary to what is currently stated).
>=20
> F19 has been removed. I added ref to F23 (and F24!) from ues-case
> "multi-party game with voice communication". Ref to F27 added to =
3.4.1.
>=20
> > - What's the difference between F24 and F34?
>=20
> I don't know, and I removed F34
>=20
> > - F30. I had to read this several times to understand it due to
> structure. I would suggest changing as follows:
> > OLD:
> > F30 The browser must be able to use the screen (or
> > a specific area of the screen) or what a certain
> > application displays on the screen to generate
> > streams.
> > NEW:
> > F30 The browser must be able to generate streams using the entire
> user
> display, a specific area of the user's display or the information =
being
> displayed by a specific application.
>=20
> Updated.
>=20
> > On this one, I also think it would be good to clarify what type of
> stream - are you talking about using protocol to share content or or =
is
> this just a video stream? Or would you have two separate requirement =
to
> cover both of these?
>=20
> I would like to avoid going into detail; the idea is that it is some
> kind of "stream" that allows sharing what is rendered on your screen,
> the requirement does not need to go into what kind it is.
>=20
> > - F32. I can't quite grok this one. Maybe you are trying to say
> something like the following?
> > OLD:
> > F32 There browser must support that STUN and TURN
> > servers to use are supplied by other entities
> > than via the web application (i.e. the network
> > provider).
> > NEW:
> > F32 The browser must support the use of STUN and TURN
> > servers that are supplied by entities
> > other than the web application (i.e. the network
> > provider).
>=20
> Updated.
>=20
> > 8) Section 7. I have mixed feelings about leaving this list with =
URLs
> in the document. I think it's good to highlight the use cases that
> weren't incorporated and why they weren't. I think it would add a lot
> more value to provide a concise summary of the reasons they weren't
> added than just including links, in particular, since we usually don't
> like to publish RFCs with links.
>=20
> I think the entire list should go before publication. The list has =
been
> kept there as a reminder of what we included and what we did not
> included. I have removed it in this version.
>=20
> > Nits:
> > ------
> > 1) Section 1, 1st paragraph, last sentence, "at least one of the
> end-user client" -> "at least one of the end-user clients"
> > 2) Section 3.2, 1st paragraph, 1st sentence:
> > - "retrives" -> "retrieves"
> > - add a section reference for "Simple Video Communication Service"
> > 3) Section 3.2. , next to last sentence. "retrieved from" -> =
"derived
> from"
> > 4) Section 3.3.5.1, 3rd paragraph,
> > - 1st sentence. "session" - "session"
> > - 2nd sentence. "straddle the boundary between the internal network
> and external." -> "straddles the boundary between the internal and
> external networks.
> > 5) Section 3.3.5.1, 4th paragraph. "they still want to have the
> traffic to stay" -> "they still want the traffic to say"
> > 6) Section 3.3.61. 1st paragraph. I'm not sure why this ends with =
":"
> > 7) Section 3.3.6.1, 2nd paragraph. "device used by one of the users
> have several" -> "device used by one of the users has several"
> > 8) Section 3.3.11.1, 1st para, 1st sentence. "In this use case is =
the
> Simple..." -> "In this use case, the Simple...."
> > 9) Section 3.3.11.1 3rd from last paragraph. "use experience" ->
> "user
> experience"
> > 10) Section 3.4.3.1, 2nd paragraph. "participant send" ->
> "participant
> sends"
> > 11) Section 4.2:
> > - F35. "of that streams" -> "that streams"
>=20
> Most Nit's fixed
>=20
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
>=20
> On 2014-01-29 10:29, Magnus Westerlund wrote:
> > Hi Partha and WG,
> >
> > I don't see any support for the changes you proposes in this
> discussion.
> > What I see some support for is to add a statement making clear that
> > there might be additional NAT/Firewall traversal mechanisms than
> > STUN/TURN. Simon proposed:
> >
> > "Note that TURN support being mandatory does not preclude a WebRTC
> > endpoint from supporting additional traversal mechanisms."
>=20
> I added this to section 3.3.4.1
>=20
> >
> >
> > However, looking at the document as it is currently written, I am
> > uncertain where this would be added. The first mention of TURN is in
> > Section 3.3.4.1, and that section is focused on the global service
> > provider perspective and the need for location based provisioning of
> > NAT/Firewall traversal server resources.
> >
> > I think it can be added to Section 3.3.5.1 without being misplaced,
> but
> > it would be given a slightly narrower scope.
> >
> > I any of you want to be more explicit where this should be included,
> > please be. If you are not forthcoming I will request the authors to
> add
> > this in what they consider sensible position.
> >
> > Cheers
> >
> > Magnus
> >
> >
> > On 2014-01-25 17:46, Parthasarathi R wrote:
> >> Hi Simon,
> >>
> >>
> >>
> >> Thanks for your understanding about my firewall/NAT related problem
> >> statement in this draft.
> >>
> >>
> >>
> >> I have proposed the firewall/NAT related text by which the specific
> >> mechanism is not highlighted in the requirement document as there =
is
> no
> >> WG consensus for any of the mechanism including TURN. It is =
possible
> to
> >> argue hypothetically in PNTAW alias that PCP is the only mechanism
> >> required in WebRTC endpoint. Also, I=92m more interested in WebRTC
> >> gateway/server (Sec 4.3. of
> >> draft-ietf-rtcweb-use-cases-and-requirements-12) requirements
> wherein it
> >> is not required to support TURN and the related mail thread is
> >> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.
> >>
> >>
> >>
> >> IMO, my proposed text without mentioning any firewall/NAT mechanism
> in
> >> the requirement document helps to move forward without depend on =
the
> >> solution discussion in PNTAW alias.
> >>
> >>
> >>
> >> Thanks
> >>
> >> Partha
> >>
> >>
> >>
> >> *From:*Cb B [mailto:cb.list6@gmail.com]
> >> *Sent:* Saturday, January 25, 2014 6:22 AM
> >> *To:* Simon Perreault
> >> *Cc:* rtcweb@ietf.org; Parthasarathi R
> >> *Subject:* Re: [rtcweb] Query/Comment on
> >> draft-ietf-rtcweb-use-cases-and-requirements-12
> >>
> >>
> >>
> >>
> >> On Jan 24, 2014 10:17 AM, "Simon Perreault"
> <simon.perreault@viagenie.ca
> >> <mailto:simon.perreault@viagenie.ca>> wrote:
> >>>
> >>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :
> >>>
> >>>> Please note that when non-IETFers read this requirement document,
> >> they come
> >>>> to the conclusion that IETF RTCWeb WG recommends TURN and not
> other
> >>>> mechanisms. I'm saying that requirement document should not be
> used
> >> as the
> >>>> mechanism to eliminate the other alternatives when there is a
> discussion
> >>>> going-on in PNTAW alias. So, I'm asking for the change.
> >>>
> >>>
> >>> I would totally agree with that sentiment, although I don't see
> your
> >> proposed text change reflecting it accurately. How about simply:
> >>>
> >>> "Note that TURN support being mandatory does not preclude a WebRTC
> >> endpoint from supporting additional traversal mechanisms."
> >>>
> >>>
> >>
> >> +1 for the above text.
> >>
> >> CB
> >>
> >>> Simon
> >>> --
> >>> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> >>> NAT64/DNS64 open-source --> http://ecdysis.viagenie.ca
> >>> STUN/TURN server --> http://numb.viagenie.ca
> >>> _______________________________________________
> >>> 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
> >>
> >
> >
>=20
>=20
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> On 2014-01-16 15:18, Hutton, Andrew wrote:
> > Some comments in the text below.
> >
> > As a general comment I suggest changing all the "FW" abbreviations =
to
> "Firewall" are have at least a first use definition.
>=20
> Done
>=20
> >
> >
> > Andy
> >
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: 15 January 2014 10:20
> >> To: Chenxin (Xin); Hutton, Andrew; Christer Holmberg; Parthasarathi
> R;
> >> rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-
> and-
> >> requirements-12
> >>
> >> WG,
> >>
> >> It has been quite some time since the WG last call ended and a new
> >> revision was submitted. As Document Shepherd I want to push this
> >> document to publication request.
> >>
> >> Chenxin proposed below three different sets of changes to the
> document.
> >> Does the WG support making these changes? Please indicate within =
the
> >> next week if you support or want to reject these changes.
> >>
> >> Thanks
> >>
> >> Magnus
> >>
> >>
> >> On 2013-10-17 12:23, Chenxin (Xin) wrote:
> >>> Hi Andy,
> >>>
> >>>
> >>>
> >>> I think you means F29 not F27:). When I read it , I realize that
> >> there
> >>> is cross and ambiguous between 3.3.2 and 3.3.3
> >>>
> >>>
> >>>
> >>> More details:
> >>>
> >>>
> >>>
> >>> The topic of 3.3.2 is "Simple Video Communication Service, =
*NAT/FW*
> >>> that blocks UDP". But in the description and requirement, only
> *NAT*
> >> is
> >>> considered.
> >>>
> >>> The topic of 3.3.3 is "Simple Video Communication Service, FW that
> >>> only allows http", But only *http proxy* deployed scenarios is
> >> considered.
> >>>
> >>>
> >>>
> >>> There are other usecases " FW block UDP, incoming TCP, Http
> >> allowing
> >>> FW without http proxy deplolyed under the permission of FW policy"
> ,
> >>> which is lost in the description. If we need consider these
> usecases
> >> , I
> >>> suggest to make some change to the description.
> >>>
> >>>
> >>>
> >>> Proposal 1 :
> >>>
> >>>
> >>>
> >>> add FW related words to section 3.3.2
> >>>
> >>> -------------------------------------------------
> >>>
> >>> 3.3.2. Simple Video Communication Service, NAT/FW that blocks UDP
> >>>
> >>>
> >>>
> >>> 3.3.2.1. Description
> >>>
> >>>
> >>>
> >>> This use-case is almost identical to the Simple Video
> >> Communication
> >>>
> >>> Service use-case (Section 3.3.1). The difference is that one of
> >> the
> >>>
> >>> users is behind a NAT*/FW* that blocks UDP traffic.
> >>>
> >>> .
> >
> > [AndyH] - I support this change.
>=20
> Changed
>=20
> >
> >
> >>>
> >>>
> >>>
> >>> 3.3.2.2. Additional Requirements
> >>>
> >>>
> >>>
> >>> F29 The browser must be able to send streams and
> >>>
> >>> data to a peer in the presence of NATs *and FWs* that
> >>>
> >>> block UDP traffic ,* when FW policy allows WebRTC
> >> traffic*.
> >
> > [AndyH] - I support this change.
>=20
> Changed
>=20
> >
> >
> >>>
> >>> -------------------------------------------------------
> >>>
> >>> Proposal 2: If the" Http allowing FW without http proxy deployed"
> >>> case is impliedly included in F29. I suggest to change the topics
> of
> >>> 3.3.3 to "Simple Video Communication Service, FW that only allows
> >>> traffic via a http proxy". So the 3.3.3 is a specific case.
> >>>
> >
> > [AndyH] Yes the heading of 3.3.3 should have been changed during the
> last edit to "Simple Video Communication Service, FW that only allows
> traffic via a HTTP Proxy". I support this change.
>=20
> Changed.
>=20
> >
> >
> >>>
> >>>
> >>> Proposal 3: If " Http allowing FW without http proxy deployed"
> >> case
> >>> need to be explicitly mentioned. I suggest to add some =
descriptions
> >> to 3.3.3
> >>>
> >>> -----------------------------------------------------------------
> >>>
> >>> 3.3.3. Simple Video Communication Service, FW that only allows =
http
> >>>
> >>>
> >>>
> >>> 3.3.3.1. Description
> >>>
> >>>
> >>>
> >>> This use-case is almost identical to the Simple Video
> >> Communication
> >>>
> >>> Service use-case (Section 3.3.1). The difference is that one of
> >> the
> >>>
> >>> users is behind a http allowing FW or a FW that only allows
> >> traffic
> >>> via a HTTP Proxy.
> >>>
> >>>
> >
> > [AndyH] I don't believe the intention here was to state a =
requirement
> that WebRTC media should be able to flow through a FW only allowing
> HTTP. Therefore I think the original description is ok it is just the
> heading that needs to change.
> >
> >
> >>>
> >>> 3.3.3.2. Additional Requirements
> >>>
> >>>
> >>>
> >>> F37 The browser must be able to send streams and
> >>>
> >>> data to a peer in the presence of http allowing FWs or FWs
> >>> that only
> >>>
> >>> allows traffic via a HTTP Proxy, when FW policy
> >>>
> >>> allows WebRTC traffic.
> >>>
> >
> > [AndyH] The existing text is in the 012 draft is ok and I don't =
think
> this change is needed as again I don't believe the intention here was
> to
> state a requirement that WebRTC media should be able to flow through a
> FW only allowing HTTP.
> >
> >
> >>> =
-------------------------------------------------------------------
> >>>
> >>>
> >
> >>>
> >>>
> >>>
> >>> Best Regards,
> >>>
> >>> Xin
> >>>
> >>>
> >>>
> >>
> >>
> >> --
> >>
> >> Magnus Westerlund
> >>
> >> =
--------------------------------------------------------------------
> --
> >> Services, Media and Network features, Ericsson Research EAB/TXM
> >> =
--------------------------------------------------------------------
> --
> >> Ericsson AB | Phone +46 10 7148287
> >> F=E4r=F6gatan 6 | Mobile +46 73 0949079
> >> SE-164 80 Stockholm, Sweden | mailto: =
magnus.westerlund@ericsson.com
> >> =
--------------------------------------------------------------------
> --
> >
> > _______________________________________________
> > 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: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] "SCTP New Data Chunk" review requested
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--14dae93405f9c2572a04f1d3f176
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I'm not sure how large the cross-over is between tsvwg and rtcweb, so I'd
like to highlight that the tsvwg version of this draft has now been
uploaded:

http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/

Reviews of the document for tsvwg would be great, as they have an
aggressive deadline for pushing the document to the IESG (early Summer).

thanks,

Ted (for the Chairs)

--14dae93405f9c2572a04f1d3f176
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">I&#39;m not sure how large the cross-over is between tsvwg and rtcwe=
b, so I&#39;d like to highlight that the tsvwg version of this draft has no=
w been uploaded:<br>
<br><a href=3D"http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/=
">http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/</a><br><br><=
/div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">Revie=
ws of the document for tsvwg would be great, as they have an aggressive dea=
dline for pushing the document to the IESG (early Summer).<br>
<br>thanks,<br><br>Ted (for the Chairs)<br></div></div>

--14dae93405f9c2572a04f1d3f176--

From harald@alvestrand.no  Fri Feb  7 10:04:11 2014
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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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On 02/07/2014 05:26 PM, Parthasarathi R wrote:
> <snip> > The server doesn't have to claim WebRTC conformance, it just has to
>> claim to interoperate with WebRTC endpoints. </snip>
> The above assumption leads to lot of confusion in WebRTC Server implementation which starts from RTCWeb requirement itself. IMO, it is better to provide the minimum IETF standard guidelines for WebRTC servers like ice-lite/ICE, BUNDLE required or not, ICE-TCP/TURN. 

It seems to me like a good document to write, but should probably be
done after we finish the requirements for WebRTC endpoints.

Unless someone wishes to start one, of course.


From stefan.lk.hakansson@ericsson.com  Fri Feb  7 10:29:06 2014
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Hi Partha!=0A=
=0A=
On 2014-02-07 17:39, Parthasarathi R wrote:=0A=
> Hi Stefan,=0A=
>=0A=
> Simon proposed the following text after the lengthy mailing discussion:=
=0A=
>=0A=
>> "Note that TURN support being mandatory does not preclude a WebRTC =0A=
>> endpoint from supporting additional traversal mechanisms."=0A=
> Whereas Sec 3.3.4.1 updated as =0A=
>=0A=
> "Note that ICE support being mandatory does not preclude a WebRTC=0A=
>    endpoint from supporting additional traversal mechanisms."=0A=
=0A=
Yes, I did that change (TURN -> ICE). My understanding was that ICE is=0A=
what is used/mandated (and it in turn makes use of STUN and TURN).=0A=
=0A=
>=0A=
> Apart from this, Tiru=0A=
> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg11246.html) and=
=0A=
> myself (http://www.ietf.org/mail-archive/web/rtcweb/current/msg11241.html=
)=0A=
> mentioned in the different mail thread that Sec 3.3.4.1 is not only the=
=0A=
> right place to put the above text. Could you please correct the F31, F32=
=0A=
> requirement text in the next revision of the draft.=0A=
=0A=
I can update them to what you propose. But I think it is a little bit=0A=
silly. If we add that note, then we should probably add to F28 "The=0A=
browser must support a baseline audio and video codec" a note saying=0A=
"this does not preclude the browser from supporting additional codecs"=0A=
and so on. After all, the requirements are meant to be what must be=0A=
supported; they do not say that additions are not allowed.=0A=
=0A=
>=0A=
> Thanks=0A=
> Partha=0A=
>=0A=
>> -----Original Message-----=0A=
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan=0A=
>> H=E5kansson LK=0A=
>> Sent: Thursday, February 06, 2014 4:23 PM=0A=
>> To: rtcweb@ietf.org=0A=
>> Subject: [rtcweb] New version of use-case draft uploaded=0A=
>>=0A=
>> Hi,=0A=
>>=0A=
>> I've uploaded a new version -13 [1]. I have made changes based on input=
=0A=
>> from Mary, Magnus and Chenxing+Andy, for details read further down.=0A=
>>=0A=
>> I hope I have caught most things; what remains to my knowledge is:=0A=
>> * Include the requirement text the first time a requirement is derived=
=0A=
>> * Group (and re-number) the claims=0A=
>>=0A=
>> I am hesitating to do the last thing 'cause of the risk of getting all=
=0A=
>> internal references into a mess, but one day soon I will feel brave=0A=
>> enough :-)=0A=
>>=0A=
>> Stefan=0A=
>>=0A=
>> [1]=0A=
>> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-=0A=
>> requirements/=0A=
>>=0A=
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
>> On 2014-01-15 21:57, Mary Barnes wrote:=0A=
>>> I had thought I had previously done a detailed review of this doc,=0A=
>> but=0A=
>> I can't find it to know whether changes suggested have been=0A=
>> incorporated. So, I have re-reviewed the document and I think it's=0A=
>> almost ready to progress. I think it needs some editorial=0A=
>> clarifications=0A=
>> and nits to be fixed as summarized below. Note, I did not review the=0A=
>> appendix.=0A=
>>> Regards,=0A=
>>> Mary.=0A=
>>> In general, I still find the style of this document very difficult to=
=0A=
>> grok since the requirements are not grouped in categories and one has=0A=
>> to=0A=
>> keep switching back and forth in the document to match requirements to=
=0A=
>> use cases.=0A=
>>> Questions/Comments for clarification:=0A=
>>> ----------------------------------------------------=0A=
>>> 1) Section 1, 1st paragraph, last sentence. It's not clear to me that=
=0A=
>> "e.g., a telephone" is meaningful. I don't think you're intending to=0A=
>> interworking with a legacy PSTN connected black phone. So, it might be=
=0A=
>> more accurate to say " e.g., a mobile phone or a SIP UA".=0A=
>>=0A=
>> Changed.=0A=
>>=0A=
>>> 2) Section 3.3.1.1. Next to last paragraph. I'm not sure what you=0A=
>> mean=0A=
>> by different "makes". I think you mean different types of devices=0A=
>> (e.g.,=0A=
>> mobile, SIP UA, etc. ). That all said, I don't think that's not so=0A=
>> relevant. I think simply stating different OSs and different browsers=0A=
>> is=0A=
>> sufficient.=0A=
>>=0A=
>> Changed.=0A=
>>=0A=
>>> 3) Section 3.3.6.1. It's not at all clear to me why this requirement=0A=
>> is considered specific to WebRTC. I would think the access network=0A=
>> changes should be transparent to WebRTC. Certainly, the device needs to=
=0A=
>> know what's happening, but I think whether this works is entirely based=
=0A=
>> on the internals of the device and the specific access network=0A=
>> technology, and not WebRTC application.=0A=
>>=0A=
>> It is not WebRTC specific per se, but is important for the use-case.=0A=
>> What parts that solve it is of less importance IMO.=0A=
>>=0A=
>>> 4) Section 3.3.10.1. Why is F24 not considered an additional=0A=
>> requirement here? Also, do you not need to have a statement as to what=
=0A=
>> other use case is the basis for this one such that the core=0A=
>> requirements=0A=
>> are reference?=0A=
>>=0A=
>> I added F24 (it belongs here). I don't think we need a statement=0A=
>> regarding the basis, it is covered in section 3.2=0A=
>>=0A=
>>> 5) Section 3.3.11.1, 2nd paragraph, 1st sentence. I don't understand=0A=
>> what this is trying to say. What is meant by "enhance intelligibility"?=
=0A=
>> And, what is meant by "pans the audio from different participants=0A=
>> differently when rendering the audio".=0A=
>>> [As an aside, I will note that some of the CLUE use cases likely=0A=
>> encompass what you are trying to communicate here in this requirements=
=0A=
>> (including subsequent paragraphs and the last "Note:"), so you may want=
=0A=
>> to look at those and use similar terminology and concepts, that CLUE=0A=
>> spent a lot of time developing. ]=0A=
>>=0A=
>> I updated (using similar terminology to Clue).=0A=
>>=0A=
>>> 6) Section 3.4. I would expect F27 to be referenced by at least one=0A=
>> of=0A=
>> these use cases.=0A=
>>=0A=
>> I added it to 3.4.1=0A=
>>=0A=
>>> 7) Section 4.2.=0A=
>>> - General: I am a bit confused as there are requirements in this=0A=
>> section that aren't referenced in section 3, including F19, F23, and=0A=
>> F27=0A=
>> . Perhaps, that's because there are some missing references in section=
=0A=
>> 3=0A=
>> (see item 7)? If not, then why are they there. At a minimum you should=
=0A=
>> add a sentence to section 4.1 indicating that not all the requirements=
=0A=
>> are derived from the use cases (contrary to what is currently stated).=
=0A=
>>=0A=
>> F19 has been removed. I added ref to F23 (and F24!) from ues-case=0A=
>> "multi-party game with voice communication". Ref to F27 added to 3.4.1.=
=0A=
>>=0A=
>>> - What's the difference between F24 and F34?=0A=
>> I don't know, and I removed F34=0A=
>>=0A=
>>> - F30. I had to read this several times to understand it due to=0A=
>> structure. I would suggest changing as follows:=0A=
>>> OLD:=0A=
>>> F30 The browser must be able to use the screen (or=0A=
>>> a specific area of the screen) or what a certain=0A=
>>> application displays on the screen to generate=0A=
>>> streams.=0A=
>>> NEW:=0A=
>>> F30 The browser must be able to generate streams using the entire=0A=
>> user=0A=
>> display, a specific area of the user's display or the information being=
=0A=
>> displayed by a specific application.=0A=
>>=0A=
>> Updated.=0A=
>>=0A=
>>> On this one, I also think it would be good to clarify what type of=0A=
>> stream - are you talking about using protocol to share content or or is=
=0A=
>> this just a video stream? Or would you have two separate requirement to=
=0A=
>> cover both of these?=0A=
>>=0A=
>> I would like to avoid going into detail; the idea is that it is some=0A=
>> kind of "stream" that allows sharing what is rendered on your screen,=0A=
>> the requirement does not need to go into what kind it is.=0A=
>>=0A=
>>> - F32. I can't quite grok this one. Maybe you are trying to say=0A=
>> something like the following?=0A=
>>> OLD:=0A=
>>> F32 There browser must support that STUN and TURN=0A=
>>> servers to use are supplied by other entities=0A=
>>> than via the web application (i.e. the network=0A=
>>> provider).=0A=
>>> NEW:=0A=
>>> F32 The browser must support the use of STUN and TURN=0A=
>>> servers that are supplied by entities=0A=
>>> other than the web application (i.e. the network=0A=
>>> provider).=0A=
>> Updated.=0A=
>>=0A=
>>> 8) Section 7. I have mixed feelings about leaving this list with URLs=
=0A=
>> in the document. I think it's good to highlight the use cases that=0A=
>> weren't incorporated and why they weren't. I think it would add a lot=0A=
>> more value to provide a concise summary of the reasons they weren't=0A=
>> added than just including links, in particular, since we usually don't=
=0A=
>> like to publish RFCs with links.=0A=
>>=0A=
>> I think the entire list should go before publication. The list has been=
=0A=
>> kept there as a reminder of what we included and what we did not=0A=
>> included. I have removed it in this version.=0A=
>>=0A=
>>> Nits:=0A=
>>> ------=0A=
>>> 1) Section 1, 1st paragraph, last sentence, "at least one of the=0A=
>> end-user client" -> "at least one of the end-user clients"=0A=
>>> 2) Section 3.2, 1st paragraph, 1st sentence:=0A=
>>> - "retrives" -> "retrieves"=0A=
>>> - add a section reference for "Simple Video Communication Service"=0A=
>>> 3) Section 3.2. , next to last sentence. "retrieved from" -> "derived=
=0A=
>> from"=0A=
>>> 4) Section 3.3.5.1, 3rd paragraph,=0A=
>>> - 1st sentence. "session" - "session"=0A=
>>> - 2nd sentence. "straddle the boundary between the internal network=0A=
>> and external." -> "straddles the boundary between the internal and=0A=
>> external networks.=0A=
>>> 5) Section 3.3.5.1, 4th paragraph. "they still want to have the=0A=
>> traffic to stay" -> "they still want the traffic to say"=0A=
>>> 6) Section 3.3.61. 1st paragraph. I'm not sure why this ends with ":"=
=0A=
>>> 7) Section 3.3.6.1, 2nd paragraph. "device used by one of the users=0A=
>> have several" -> "device used by one of the users has several"=0A=
>>> 8) Section 3.3.11.1, 1st para, 1st sentence. "In this use case is the=
=0A=
>> Simple..." -> "In this use case, the Simple...."=0A=
>>> 9) Section 3.3.11.1 3rd from last paragraph. "use experience" ->=0A=
>> "user=0A=
>> experience"=0A=
>>> 10) Section 3.4.3.1, 2nd paragraph. "participant send" ->=0A=
>> "participant=0A=
>> sends"=0A=
>>> 11) Section 4.2:=0A=
>>> - F35. "of that streams" -> "that streams"=0A=
>> Most Nit's fixed=0A=
>>=0A=
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
>>=0A=
>>=0A=
>> On 2014-01-29 10:29, Magnus Westerlund wrote:=0A=
>>> Hi Partha and WG,=0A=
>>>=0A=
>>> I don't see any support for the changes you proposes in this=0A=
>> discussion.=0A=
>>> What I see some support for is to add a statement making clear that=0A=
>>> there might be additional NAT/Firewall traversal mechanisms than=0A=
>>> STUN/TURN. Simon proposed:=0A=
>>>=0A=
>>> "Note that TURN support being mandatory does not preclude a WebRTC=0A=
>>> endpoint from supporting additional traversal mechanisms."=0A=
>> I added this to section 3.3.4.1=0A=
>>=0A=
>>>=0A=
>>> However, looking at the document as it is currently written, I am=0A=
>>> uncertain where this would be added. The first mention of TURN is in=0A=
>>> Section 3.3.4.1, and that section is focused on the global service=0A=
>>> provider perspective and the need for location based provisioning of=0A=
>>> NAT/Firewall traversal server resources.=0A=
>>>=0A=
>>> I think it can be added to Section 3.3.5.1 without being misplaced,=0A=
>> but=0A=
>>> it would be given a slightly narrower scope.=0A=
>>>=0A=
>>> I any of you want to be more explicit where this should be included,=0A=
>>> please be. If you are not forthcoming I will request the authors to=0A=
>> add=0A=
>>> this in what they consider sensible position.=0A=
>>>=0A=
>>> Cheers=0A=
>>>=0A=
>>> Magnus=0A=
>>>=0A=
>>>=0A=
>>> On 2014-01-25 17:46, Parthasarathi R wrote:=0A=
>>>> Hi Simon,=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> Thanks for your understanding about my firewall/NAT related problem=0A=
>>>> statement in this draft.=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> I have proposed the firewall/NAT related text by which the specific=0A=
>>>> mechanism is not highlighted in the requirement document as there is=
=0A=
>> no=0A=
>>>> WG consensus for any of the mechanism including TURN. It is possible=
=0A=
>> to=0A=
>>>> argue hypothetically in PNTAW alias that PCP is the only mechanism=0A=
>>>> required in WebRTC endpoint. Also, I=92m more interested in WebRTC=0A=
>>>> gateway/server (Sec 4.3. of=0A=
>>>> draft-ietf-rtcweb-use-cases-and-requirements-12) requirements=0A=
>> wherein it=0A=
>>>> is not required to support TURN and the related mail thread is=0A=
>>>> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> IMO, my proposed text without mentioning any firewall/NAT mechanism=0A=
>> in=0A=
>>>> the requirement document helps to move forward without depend on the=
=0A=
>>>> solution discussion in PNTAW alias.=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> Thanks=0A=
>>>>=0A=
>>>> Partha=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> *From:*Cb B [mailto:cb.list6@gmail.com]=0A=
>>>> *Sent:* Saturday, January 25, 2014 6:22 AM=0A=
>>>> *To:* Simon Perreault=0A=
>>>> *Cc:* rtcweb@ietf.org; Parthasarathi R=0A=
>>>> *Subject:* Re: [rtcweb] Query/Comment on=0A=
>>>> draft-ietf-rtcweb-use-cases-and-requirements-12=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> On Jan 24, 2014 10:17 AM, "Simon Perreault"=0A=
>> <simon.perreault@viagenie.ca=0A=
>>>> <mailto:simon.perreault@viagenie.ca>> wrote:=0A=
>>>>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :=0A=
>>>>>=0A=
>>>>>> Please note that when non-IETFers read this requirement document,=0A=
>>>> they come=0A=
>>>>>> to the conclusion that IETF RTCWeb WG recommends TURN and not=0A=
>> other=0A=
>>>>>> mechanisms. I'm saying that requirement document should not be=0A=
>> used=0A=
>>>> as the=0A=
>>>>>> mechanism to eliminate the other alternatives when there is a=0A=
>> discussion=0A=
>>>>>> going-on in PNTAW alias. So, I'm asking for the change.=0A=
>>>>>=0A=
>>>>> I would totally agree with that sentiment, although I don't see=0A=
>> your=0A=
>>>> proposed text change reflecting it accurately. How about simply:=0A=
>>>>> "Note that TURN support being mandatory does not preclude a WebRTC=0A=
>>>> endpoint from supporting additional traversal mechanisms."=0A=
>>>>>=0A=
>>>> +1 for the above text.=0A=
>>>>=0A=
>>>> CB=0A=
>>>>=0A=
>>>>> Simon=0A=
>>>>> --=0A=
>>>>> DTN made easy, lean, and smart --> http://postellation.viagenie.ca=0A=
>>>>> NAT64/DNS64 open-source --> http://ecdysis.viagenie.ca=0A=
>>>>> STUN/TURN server --> http://numb.viagenie.ca=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=0A=
>>>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>>>>=0A=
>>>=0A=
>>=0A=
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0A=
>>=0A=
>> On 2014-01-16 15:18, Hutton, Andrew wrote:=0A=
>>> Some comments in the text below.=0A=
>>>=0A=
>>> As a general comment I suggest changing all the "FW" abbreviations to=
=0A=
>> "Firewall" are have at least a first use definition.=0A=
>>=0A=
>> Done=0A=
>>=0A=
>>>=0A=
>>> Andy=0A=
>>>=0A=
>>>=0A=
>>>> -----Original Message-----=0A=
>>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=0A=
>>>> Sent: 15 January 2014 10:20=0A=
>>>> To: Chenxin (Xin); Hutton, Andrew; Christer Holmberg; Parthasarathi=0A=
>> R;=0A=
>>>> rtcweb@ietf.org=0A=
>>>> Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-=0A=
>> and-=0A=
>>>> requirements-12=0A=
>>>>=0A=
>>>> WG,=0A=
>>>>=0A=
>>>> It has been quite some time since the WG last call ended and a new=0A=
>>>> revision was submitted. As Document Shepherd I want to push this=0A=
>>>> document to publication request.=0A=
>>>>=0A=
>>>> Chenxin proposed below three different sets of changes to the=0A=
>> document.=0A=
>>>> Does the WG support making these changes? Please indicate within the=
=0A=
>>>> next week if you support or want to reject these changes.=0A=
>>>>=0A=
>>>> Thanks=0A=
>>>>=0A=
>>>> Magnus=0A=
>>>>=0A=
>>>>=0A=
>>>> On 2013-10-17 12:23, Chenxin (Xin) wrote:=0A=
>>>>> Hi Andy,=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> I think you means F29 not F27:). When I read it , I realize that=0A=
>>>> there=0A=
>>>>> is cross and ambiguous between 3.3.2 and 3.3.3=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> More details:=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> The topic of 3.3.2 is "Simple Video Communication Service, *NAT/FW*=
=0A=
>>>>> that blocks UDP". But in the description and requirement, only=0A=
>> *NAT*=0A=
>>>> is=0A=
>>>>> considered.=0A=
>>>>>=0A=
>>>>> The topic of 3.3.3 is "Simple Video Communication Service, FW that=0A=
>>>>> only allows http", But only *http proxy* deployed scenarios is=0A=
>>>> considered.=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> There are other usecases " FW block UDP, incoming TCP, Http=0A=
>>>> allowing=0A=
>>>>> FW without http proxy deplolyed under the permission of FW policy"=0A=
>> ,=0A=
>>>>> which is lost in the description. If we need consider these=0A=
>> usecases=0A=
>>>> , I=0A=
>>>>> suggest to make some change to the description.=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> Proposal 1 :=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> add FW related words to section 3.3.2=0A=
>>>>>=0A=
>>>>> -------------------------------------------------=0A=
>>>>>=0A=
>>>>> 3.3.2. Simple Video Communication Service, NAT/FW that blocks UDP=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> 3.3.2.1. Description=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> This use-case is almost identical to the Simple Video=0A=
>>>> Communication=0A=
>>>>> Service use-case (Section 3.3.1). The difference is that one of=0A=
>>>> the=0A=
>>>>> users is behind a NAT*/FW* that blocks UDP traffic.=0A=
>>>>>=0A=
>>>>> .=0A=
>>> [AndyH] - I support this change.=0A=
>> Changed=0A=
>>=0A=
>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> 3.3.2.2. Additional Requirements=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> F29 The browser must be able to send streams and=0A=
>>>>>=0A=
>>>>> data to a peer in the presence of NATs *and FWs* that=0A=
>>>>>=0A=
>>>>> block UDP traffic ,* when FW policy allows WebRTC=0A=
>>>> traffic*.=0A=
>>> [AndyH] - I support this change.=0A=
>> Changed=0A=
>>=0A=
>>>=0A=
>>>>> -------------------------------------------------------=0A=
>>>>>=0A=
>>>>> Proposal 2: If the" Http allowing FW without http proxy deployed"=0A=
>>>>> case is impliedly included in F29. I suggest to change the topics=0A=
>> of=0A=
>>>>> 3.3.3 to "Simple Video Communication Service, FW that only allows=0A=
>>>>> traffic via a http proxy". So the 3.3.3 is a specific case.=0A=
>>>>>=0A=
>>> [AndyH] Yes the heading of 3.3.3 should have been changed during the=0A=
>> last edit to "Simple Video Communication Service, FW that only allows=0A=
>> traffic via a HTTP Proxy". I support this change.=0A=
>>=0A=
>> Changed.=0A=
>>=0A=
>>>=0A=
>>>>>=0A=
>>>>> Proposal 3: If " Http allowing FW without http proxy deployed"=0A=
>>>> case=0A=
>>>>> need to be explicitly mentioned. I suggest to add some descriptions=
=0A=
>>>> to 3.3.3=0A=
>>>>> -----------------------------------------------------------------=0A=
>>>>>=0A=
>>>>> 3.3.3. Simple Video Communication Service, FW that only allows http=
=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> 3.3.3.1. Description=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> This use-case is almost identical to the Simple Video=0A=
>>>> Communication=0A=
>>>>> Service use-case (Section 3.3.1). The difference is that one of=0A=
>>>> the=0A=
>>>>> users is behind a http allowing FW or a FW that only allows=0A=
>>>> traffic=0A=
>>>>> via a HTTP Proxy.=0A=
>>>>>=0A=
>>>>>=0A=
>>> [AndyH] I don't believe the intention here was to state a requirement=
=0A=
>> that WebRTC media should be able to flow through a FW only allowing=0A=
>> HTTP. Therefore I think the original description is ok it is just the=0A=
>> heading that needs to change.=0A=
>>>=0A=
>>>>> 3.3.3.2. Additional Requirements=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> F37 The browser must be able to send streams and=0A=
>>>>>=0A=
>>>>> data to a peer in the presence of http allowing FWs or FWs=0A=
>>>>> that only=0A=
>>>>>=0A=
>>>>> allows traffic via a HTTP Proxy, when FW policy=0A=
>>>>>=0A=
>>>>> allows WebRTC traffic.=0A=
>>>>>=0A=
>>> [AndyH] The existing text is in the 012 draft is ok and I don't think=
=0A=
>> this change is needed as again I don't believe the intention here was=0A=
>> to=0A=
>> state a requirement that WebRTC media should be able to flow through a=
=0A=
>> FW only allowing HTTP.=0A=
>>>=0A=
>>>>> -------------------------------------------------------------------=
=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> Best Regards,=0A=
>>>>>=0A=
>>>>> Xin=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>=0A=
>>>> --=0A=
>>>>=0A=
>>>> Magnus Westerlund=0A=
>>>>=0A=
>>>> --------------------------------------------------------------------=
=0A=
>> --=0A=
>>>> Services, Media and Network features, Ericsson Research EAB/TXM=0A=
>>>> --------------------------------------------------------------------=
=0A=
>> --=0A=
>>>> Ericsson AB | Phone +46 10 7148287=0A=
>>>> F=E4r=F6gatan 6 | Mobile +46 73 0949079=0A=
>>>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com=
=0A=
>>>> --------------------------------------------------------------------=
=0A=
>> --=0A=
>>> _______________________________________________=0A=
>>> rtcweb mailing list=0A=
>>> rtcweb@ietf.org=0A=
>>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>>>=0A=
>> _______________________________________________=0A=
>> rtcweb mailing list=0A=
>> rtcweb@ietf.org=0A=
>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

From Michael.Tuexen@lurchi.franken.de  Fri Feb  7 10:58:49 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Date: Fri, 7 Feb 2014 19:58:22 +0100
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Fwd: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-dtls-encaps: To End 28th February 2014
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Dear all,

any comments would be very welcome!

Best regards
Michael

Begin forwarded message:

> From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
> Subject: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-dtls-encaps: To End =
28th February 2014
> Date: February 7, 2014 6:18:33 PM GMT+01:00
> To: tsvwg WG <tsvwg@ietf.org>
> Cc: tsvwg chair <tsvwg-chairs@tools.ietf.org>
> Reply-To: gorry@erg.abdn.ac.uk
>=20
> This email announces the start of a working group last call
> of draft-ietf-tsvwg-sctp-dtls-encaps-03,
> "DTLS Encapsulation of SCTP Packets". This document was
> discussed at the WG meeting in Vancouver and was thought at that
> meeting to be ready for WG review, this email starts this process by =
requesting review comments.
>=20
> Please send any comments, notes of support, or concerns to the TSVWG =
list.
>=20
> The draft is available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps
>=20
> The last call will run for TWO weeks, ending 28th February 2014.
>=20
> James, David, and Gorry
> (TSVWG Chairs)
> tsvwg-chairs@ietf.org
>=20


From harald@alvestrand.no  Fri Feb  7 11:56:23 2014
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Subject: Re: [rtcweb] "SCTP New Data Chunk" review requested
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Based on my understanding of WG consensus, I have added this as a "MUST
support" in draft-ietf-rtcweb-transport. (Not uploaded yet; I'll try to
get any other comments that come in up to the deadline.)

On 02/07/2014 06:01 PM, Ted Hardie wrote:
> I'm not sure how large the cross-over is between tsvwg and rtcweb, so
> I'd like to highlight that the tsvwg version of this draft has now
> been uploaded:
>
> http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/
>
> Reviews of the document for tsvwg would be great, as they have an
> aggressive deadline for pushing the document to the IESG (early Summer).
>
> thanks,
>
> Ted (for the Chairs)
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


--------------000809070604010108090803
Content-Type: text/html; charset=ISO-8859-1
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">Based on my understanding of WG
      consensus, I have added this as a "MUST support" in
      draft-ietf-rtcweb-transport. (Not uploaded yet; I'll try to get
      any other comments that come in up to the deadline.)<br>
      <br>
      On 02/07/2014 06:01 PM, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMCHmxvJ6OcgPqnSev0qj6wOaW73sz-sqh3pDMDFz4h7xQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default" style="font-family:georgia,serif">I'm
          not sure how large the cross-over is between tsvwg and rtcweb,
          so I'd like to highlight that the tsvwg version of this draft
          has now been uploaded:<br>
          <br>
          <a moz-do-not-send="true"
            href="http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/">http://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/</a><br>
          <br>
        </div>
        <div class="gmail_default" style="font-family:georgia,serif">Reviews
          of the document for tsvwg would be great, as they have an
          aggressive deadline for pushing the document to the IESG
          (early Summer).<br>
          <br>
          thanks,<br>
          <br>
          Ted (for the Chairs)<br>
        </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>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------000809070604010108090803--

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Date: Fri, 7 Feb 2014 22:01:44 +0000
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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> >> Also, I think "DOMString" PPID is too specific to Javascript, instead =
it
> should probably have a generic name like "UTF-16 String". The send API ca=
n
> still use DOMString as this is Javascript API.
> > Any comments from others?
>=20
> Note: The WebSockets protocol defines the transferred strings as UTF-8.
> http://tools.ietf.org/html/rfc6455#section-5.6
>=20
> As far as I can tell, we've always intended to follow that example.
>=20
> The fact that Javascript implementations currently choose to represent
> text strings as UTF-16 at their API is sad, but not an argument for
> sending that particular text representation over the wire, or reflecting
> the name in the API.

[Raju] I agree that using UTF-8 is desired and more appropriate! Then, shou=
ld the PPID be changed from "DOMString" to "UTF-8"? Javascript based apps h=
ave to use some library to do the conversion of DOMString/UTF-16 to UTF-8. =
Alternatively, browsers can do this conversion under the APIs (send and onm=
essage) before sending and after receiving UTF-8 PPID data.
Without such conversion webrtc interworking between browsers and native cli=
ents will be problematic (basically, will not work).

Another option, which is more flexible, is to define PPIDs for different en=
codings like "UTF-8", "UTF-16" (or even "base64" for binary to text instead=
 of using "binary" directly); then pass this encoding information to send()=
 and onmessage() calls, which will use these PPIDs. Passing encoding inform=
ation might be implicit (like Javavscript DOMString) in most languages by t=
he type of the argument to send. This way it is upto the application to dea=
l with the encoding conversions as it see fits.=20

The latter approach is best suitable for interworking between clients of si=
milar type (web or native); but it is bit painful (to do conversions) for c=
lients of different types.

I am wondering what kind of API calls native WebRTC API stacks (e.g. Google=
 Chrome's http://www.webrtc.org/webrtc-native-code-package) provide for dat=
a channel send/onmessage ()? UTF-16 strings? Or UTF-8?

-Raju


From martin.thomson@gmail.com  Fri Feb  7 15:45:27 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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I don't think that we need any complication here.  It's a string.

Strings are UTF-8 on the wire.

Strings are UTF-16 (mostly) in JavaScript.

Anything else would generate pain.

On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com> wrote:
>> >> Also, I think "DOMString" PPID is too specific to Javascript, instead=
 it
>> should probably have a generic name like "UTF-16 String". The send API c=
an
>> still use DOMString as this is Javascript API.
>> > Any comments from others?
>>
>> Note: The WebSockets protocol defines the transferred strings as UTF-8.
>> http://tools.ietf.org/html/rfc6455#section-5.6
>>
>> As far as I can tell, we've always intended to follow that example.
>>
>> The fact that Javascript implementations currently choose to represent
>> text strings as UTF-16 at their API is sad, but not an argument for
>> sending that particular text representation over the wire, or reflecting
>> the name in the API.
>
> [Raju] I agree that using UTF-8 is desired and more appropriate! Then, sh=
ould the PPID be changed from "DOMString" to "UTF-8"? Javascript based apps=
 have to use some library to do the conversion of DOMString/UTF-16 to UTF-8=
. Alternatively, browsers can do this conversion under the APIs (send and o=
nmessage) before sending and after receiving UTF-8 PPID data.
> Without such conversion webrtc interworking between browsers and native c=
lients will be problematic (basically, will not work).
>
> Another option, which is more flexible, is to define PPIDs for different =
encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text inste=
ad of using "binary" directly); then pass this encoding information to send=
() and onmessage() calls, which will use these PPIDs. Passing encoding info=
rmation might be implicit (like Javavscript DOMString) in most languages by=
 the type of the argument to send. This way it is upto the application to d=
eal with the encoding conversions as it see fits.
>
> The latter approach is best suitable for interworking between clients of =
similar type (web or native); but it is bit painful (to do conversions) for=
 clients of different types.
>
> I am wondering what kind of API calls native WebRTC API stacks (e.g. Goog=
le Chrome's http://www.webrtc.org/webrtc-native-code-package) provide for d=
ata channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>
> -Raju
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From christer.holmberg@ericsson.com  Fri Feb  7 16:01:18 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFCF6C8@US70UWXCHMBA02.zam.alcatel-lucent.com> <B304F67A-9EA2-44A0-86DD-9DD0E21CB86F@lurchi.franken.de> <52F4182F.60404@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17826DFD40C3@US70UWXCHMBA02.zam.alcatel-lucent.com>, <CABkgnnXMYt7teSpxm7chTvQPR8ThKzKQ_bq3Po_FNFv2tdBFGQ@mail.gmail.com>
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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+ 1

Regards,

Christer

________________________________________
From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson [martin.=
thomson@gmail.com]
Sent: Saturday, 08 February 2014 1:45 AM
To: Makaraju, Maridi Raju (Raju)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: =
Usage of PPID for protocol multiplexing)

I don't think that we need any complication here.  It's a string.

Strings are UTF-8 on the wire.

Strings are UTF-16 (mostly) in JavaScript.

Anything else would generate pain.

On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com> wrote:
>> >> Also, I think "DOMString" PPID is too specific to Javascript, instead=
 it
>> should probably have a generic name like "UTF-16 String". The send API c=
an
>> still use DOMString as this is Javascript API.
>> > Any comments from others?
>>
>> Note: The WebSockets protocol defines the transferred strings as UTF-8.
>> http://tools.ietf.org/html/rfc6455#section-5.6
>>
>> As far as I can tell, we've always intended to follow that example.
>>
>> The fact that Javascript implementations currently choose to represent
>> text strings as UTF-16 at their API is sad, but not an argument for
>> sending that particular text representation over the wire, or reflecting
>> the name in the API.
>
> [Raju] I agree that using UTF-8 is desired and more appropriate! Then, sh=
ould the PPID be changed from "DOMString" to "UTF-8"? Javascript based apps=
 have to use some library to do the conversion of DOMString/UTF-16 to UTF-8=
. Alternatively, browsers can do this conversion under the APIs (send and o=
nmessage) before sending and after receiving UTF-8 PPID data.
> Without such conversion webrtc interworking between browsers and native c=
lients will be problematic (basically, will not work).
>
> Another option, which is more flexible, is to define PPIDs for different =
encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text inste=
ad of using "binary" directly); then pass this encoding information to send=
() and onmessage() calls, which will use these PPIDs. Passing encoding info=
rmation might be implicit (like Javavscript DOMString) in most languages by=
 the type of the argument to send. This way it is upto the application to d=
eal with the encoding conversions as it see fits.
>
> The latter approach is best suitable for interworking between clients of =
similar type (web or native); but it is bit painful (to do conversions) for=
 clients of different types.
>
> I am wondering what kind of API calls native WebRTC API stacks (e.g. Goog=
le Chrome's http://www.webrtc.org/webrtc-native-code-package) provide for d=
ata channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>
> -Raju
>
> _______________________________________________
> 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: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Date: Sat, 8 Feb 2014 01:14:52 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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That's good. Glad to see that no one liked the complex alternate option... =
yet! :-)
I just put it there for completeness and some discussion.
With "utf-8 on wire", which I too think is the right approach, there is som=
e spec work todo:
1. Define IANA utf-8 ppid. Creating a new one is better instead of replacin=
g existing DOMString as DOMString may be used by browsers for sometime to c=
ome and also to have backward compatibility/interworking.
2. Data channel core spec (http://tools.ietf.org/html/draft-ietf-rtcweb-dat=
a-channel-06#section-6.6) need to specify explictly the wire format is utf-=
8.

Then browsers need to do the conversion from utf-16 DOMstring to utf-8 befo=
re writing on wire also in the reverse direction.=20

-Raju

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, February 07, 2014 6:01 PM
> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel=
:
> Usage of PPID for protocol multiplexing)
>=20
>=20
> + 1
>=20
> Regards,
>=20
> Christer
>=20
> ________________________________________
> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
> [martin.thomson@gmail.com]
> Sent: Saturday, 08 February 2014 1:45 AM
> To: Makaraju, Maridi Raju (Raju)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel=
:
> Usage of PPID for protocol multiplexing)
>=20
> I don't think that we need any complication here.  It's a string.
>=20
> Strings are UTF-8 on the wire.
>=20
> Strings are UTF-16 (mostly) in JavaScript.
>=20
> Anything else would generate pain.
>=20
> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
> <Raju.Makaraju@alcatel-lucent.com> wrote:
> >> >> Also, I think "DOMString" PPID is too specific to Javascript, inste=
ad
> it
> >> should probably have a generic name like "UTF-16 String". The send API
> can
> >> still use DOMString as this is Javascript API.
> >> > Any comments from others?
> >>
> >> Note: The WebSockets protocol defines the transferred strings as UTF-8=
.
> >> http://tools.ietf.org/html/rfc6455#section-5.6
> >>
> >> As far as I can tell, we've always intended to follow that example.
> >>
> >> The fact that Javascript implementations currently choose to represent
> >> text strings as UTF-16 at their API is sad, but not an argument for
> >> sending that particular text representation over the wire, or reflecti=
ng
> >> the name in the API.
> >
> > [Raju] I agree that using UTF-8 is desired and more appropriate! Then,
> should the PPID be changed from "DOMString" to "UTF-8"? Javascript based
> apps have to use some library to do the conversion of DOMString/UTF-16 to
> UTF-8. Alternatively, browsers can do this conversion under the APIs (sen=
d
> and onmessage) before sending and after receiving UTF-8 PPID data.
> > Without such conversion webrtc interworking between browsers and native
> clients will be problematic (basically, will not work).
> >
> > Another option, which is more flexible, is to define PPIDs for differen=
t
> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text
> instead of using "binary" directly); then pass this encoding information =
to
> send() and onmessage() calls, which will use these PPIDs. Passing encodin=
g
> information might be implicit (like Javavscript DOMString) in most langua=
ges
> by the type of the argument to send. This way it is upto the application =
to
> deal with the encoding conversions as it see fits.
> >
> > The latter approach is best suitable for interworking between clients o=
f
> similar type (web or native); but it is bit painful (to do conversions) f=
or
> clients of different types.
> >
> > I am wondering what kind of API calls native WebRTC API stacks (e.g.
> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) provide
> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
> >
> > -Raju
> >
> > _______________________________________________
> > 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 partha@parthasarathi.co.in  Sat Feb  8 01:13:07 2014
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Stefan,

<snip>
> Yes, I did that change (TURN -> ICE). My understanding was that ICE is
> what is used/mandated (and it in turn makes use of STUN and TURN).
>
</snip>

ICE is mandated in RTCWeb but I disagree to your assumption that ICE in =
turn
makes use of TURN. My concern is that The requirement document is =
misleading
about TURN requirements. ICE-TCP avoids TURN server as the middle-box in =
the
network. ICE-TCP is host candidate whereas TURN is relay candidate and =
as
per ICE candidate selection, host candidate is preferred over relay
candidates.=20

In case of WebRTC servers (like conference)/gateway implementation, =
WebRTC
server is never going to be behind firewall, ICE-TCP serve the purpose =
and
there is no need of TURN server as a middle-box in the network. So, =
Could
you please correct the F31, F32 requirement text in the next revision of =
the
draft.
 =20
Thanks
Partha

> -----Original Message-----
> From: Stefan H=E5kansson LK [mailto:stefan.lk.hakansson@ericsson.com]
> Sent: Friday, February 07, 2014 11:59 PM
> To: Parthasarathi R; rtcweb@ietf.org
> Subject: Re: [rtcweb] New version of use-case draft uploaded
>=20
> Hi Partha!
>=20
> On 2014-02-07 17:39, Parthasarathi R wrote:
> > Hi Stefan,
> >
> > Simon proposed the following text after the lengthy mailing
> discussion:
> >
> >> "Note that TURN support being mandatory does not preclude a WebRTC
> >> endpoint from supporting additional traversal mechanisms."
> > Whereas Sec 3.3.4.1 updated as
> >
> > "Note that ICE support being mandatory does not preclude a WebRTC
> >    endpoint from supporting additional traversal mechanisms."
>=20
> Yes, I did that change (TURN -> ICE). My understanding was that ICE is
> what is used/mandated (and it in turn makes use of STUN and TURN).
>=20
> >
> > Apart from this, Tiru
> > (http://www.ietf.org/mail-archive/web/rtcweb/current/msg11246.html)
> and
> > myself (http://www.ietf.org/mail-
> archive/web/rtcweb/current/msg11241.html)
> > mentioned in the different mail thread that Sec 3.3.4.1 is not only
> the
> > right place to put the above text. Could you please correct the F31,
> F32
> > requirement text in the next revision of the draft.
>=20
> I can update them to what you propose. But I think it is a little bit
> silly. If we add that note, then we should probably add to F28 "The
> browser must support a baseline audio and video codec" a note saying
> "this does not preclude the browser from supporting additional codecs"
> and so on. After all, the requirements are meant to be what must be
> supported; they do not say that additions are not allowed.
>=20
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan
> >> H=E5kansson LK
> >> Sent: Thursday, February 06, 2014 4:23 PM
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] New version of use-case draft uploaded
> >>
> >> Hi,
> >>
> >> I've uploaded a new version -13 [1]. I have made changes based on
> input
> >> from Mary, Magnus and Chenxing+Andy, for details read further down.
> >>
> >> I hope I have caught most things; what remains to my knowledge is:
> >> * Include the requirement text the first time a requirement is
> derived
> >> * Group (and re-number) the claims
> >>
> >> I am hesitating to do the last thing 'cause of the risk of getting
> all
> >> internal references into a mess, but one day soon I will feel brave
> >> enough :-)
> >>
> >> Stefan
> >>
> >> [1]
> >> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-
> >> requirements/
> >>
> >> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >> On 2014-01-15 21:57, Mary Barnes wrote:
> >>> I had thought I had previously done a detailed review of this doc,
> >> but
> >> I can't find it to know whether changes suggested have been
> >> incorporated. So, I have re-reviewed the document and I think it's
> >> almost ready to progress. I think it needs some editorial
> >> clarifications
> >> and nits to be fixed as summarized below. Note, I did not review =
the
> >> appendix.
> >>> Regards,
> >>> Mary.
> >>> In general, I still find the style of this document very difficult
> to
> >> grok since the requirements are not grouped in categories and one
> has
> >> to
> >> keep switching back and forth in the document to match requirements
> to
> >> use cases.
> >>> Questions/Comments for clarification:
> >>> ----------------------------------------------------
> >>> 1) Section 1, 1st paragraph, last sentence. It's not clear to me
> that
> >> "e.g., a telephone" is meaningful. I don't think you're intending =
to
> >> interworking with a legacy PSTN connected black phone. So, it might
> be
> >> more accurate to say " e.g., a mobile phone or a SIP UA".
> >>
> >> Changed.
> >>
> >>> 2) Section 3.3.1.1. Next to last paragraph. I'm not sure what you
> >> mean
> >> by different "makes". I think you mean different types of devices
> >> (e.g.,
> >> mobile, SIP UA, etc. ). That all said, I don't think that's not so
> >> relevant. I think simply stating different OSs and different
> browsers
> >> is
> >> sufficient.
> >>
> >> Changed.
> >>
> >>> 3) Section 3.3.6.1. It's not at all clear to me why this
> requirement
> >> is considered specific to WebRTC. I would think the access network
> >> changes should be transparent to WebRTC. Certainly, the device =
needs
> to
> >> know what's happening, but I think whether this works is entirely
> based
> >> on the internals of the device and the specific access network
> >> technology, and not WebRTC application.
> >>
> >> It is not WebRTC specific per se, but is important for the =
use-case.
> >> What parts that solve it is of less importance IMO.
> >>
> >>> 4) Section 3.3.10.1. Why is F24 not considered an additional
> >> requirement here? Also, do you not need to have a statement as to
> what
> >> other use case is the basis for this one such that the core
> >> requirements
> >> are reference?
> >>
> >> I added F24 (it belongs here). I don't think we need a statement
> >> regarding the basis, it is covered in section 3.2
> >>
> >>> 5) Section 3.3.11.1, 2nd paragraph, 1st sentence. I don't
> understand
> >> what this is trying to say. What is meant by "enhance
> intelligibility"?
> >> And, what is meant by "pans the audio from different participants
> >> differently when rendering the audio".
> >>> [As an aside, I will note that some of the CLUE use cases likely
> >> encompass what you are trying to communicate here in this
> requirements
> >> (including subsequent paragraphs and the last "Note:"), so you may
> want
> >> to look at those and use similar terminology and concepts, that =
CLUE
> >> spent a lot of time developing. ]
> >>
> >> I updated (using similar terminology to Clue).
> >>
> >>> 6) Section 3.4. I would expect F27 to be referenced by at least =
one
> >> of
> >> these use cases.
> >>
> >> I added it to 3.4.1
> >>
> >>> 7) Section 4.2.
> >>> - General: I am a bit confused as there are requirements in this
> >> section that aren't referenced in section 3, including F19, F23, =
and
> >> F27
> >> . Perhaps, that's because there are some missing references in
> section
> >> 3
> >> (see item 7)? If not, then why are they there. At a minimum you
> should
> >> add a sentence to section 4.1 indicating that not all the
> requirements
> >> are derived from the use cases (contrary to what is currently
> stated).
> >>
> >> F19 has been removed. I added ref to F23 (and F24!) from ues-case
> >> "multi-party game with voice communication". Ref to F27 added to
> 3.4.1.
> >>
> >>> - What's the difference between F24 and F34?
> >> I don't know, and I removed F34
> >>
> >>> - F30. I had to read this several times to understand it due to
> >> structure. I would suggest changing as follows:
> >>> OLD:
> >>> F30 The browser must be able to use the screen (or
> >>> a specific area of the screen) or what a certain
> >>> application displays on the screen to generate
> >>> streams.
> >>> NEW:
> >>> F30 The browser must be able to generate streams using the entire
> >> user
> >> display, a specific area of the user's display or the information
> being
> >> displayed by a specific application.
> >>
> >> Updated.
> >>
> >>> On this one, I also think it would be good to clarify what type of
> >> stream - are you talking about using protocol to share content or =
or
> is
> >> this just a video stream? Or would you have two separate =
requirement
> to
> >> cover both of these?
> >>
> >> I would like to avoid going into detail; the idea is that it is =
some
> >> kind of "stream" that allows sharing what is rendered on your
> screen,
> >> the requirement does not need to go into what kind it is.
> >>
> >>> - F32. I can't quite grok this one. Maybe you are trying to say
> >> something like the following?
> >>> OLD:
> >>> F32 There browser must support that STUN and TURN
> >>> servers to use are supplied by other entities
> >>> than via the web application (i.e. the network
> >>> provider).
> >>> NEW:
> >>> F32 The browser must support the use of STUN and TURN
> >>> servers that are supplied by entities
> >>> other than the web application (i.e. the network
> >>> provider).
> >> Updated.
> >>
> >>> 8) Section 7. I have mixed feelings about leaving this list with
> URLs
> >> in the document. I think it's good to highlight the use cases that
> >> weren't incorporated and why they weren't. I think it would add a
> lot
> >> more value to provide a concise summary of the reasons they weren't
> >> added than just including links, in particular, since we usually
> don't
> >> like to publish RFCs with links.
> >>
> >> I think the entire list should go before publication. The list has
> been
> >> kept there as a reminder of what we included and what we did not
> >> included. I have removed it in this version.
> >>
> >>> Nits:
> >>> ------
> >>> 1) Section 1, 1st paragraph, last sentence, "at least one of the
> >> end-user client" -> "at least one of the end-user clients"
> >>> 2) Section 3.2, 1st paragraph, 1st sentence:
> >>> - "retrives" -> "retrieves"
> >>> - add a section reference for "Simple Video Communication Service"
> >>> 3) Section 3.2. , next to last sentence. "retrieved from" ->
> "derived
> >> from"
> >>> 4) Section 3.3.5.1, 3rd paragraph,
> >>> - 1st sentence. "session" - "session"
> >>> - 2nd sentence. "straddle the boundary between the internal =
network
> >> and external." -> "straddles the boundary between the internal and
> >> external networks.
> >>> 5) Section 3.3.5.1, 4th paragraph. "they still want to have the
> >> traffic to stay" -> "they still want the traffic to say"
> >>> 6) Section 3.3.61. 1st paragraph. I'm not sure why this ends with
> ":"
> >>> 7) Section 3.3.6.1, 2nd paragraph. "device used by one of the =
users
> >> have several" -> "device used by one of the users has several"
> >>> 8) Section 3.3.11.1, 1st para, 1st sentence. "In this use case is
> the
> >> Simple..." -> "In this use case, the Simple...."
> >>> 9) Section 3.3.11.1 3rd from last paragraph. "use experience" ->
> >> "user
> >> experience"
> >>> 10) Section 3.4.3.1, 2nd paragraph. "participant send" ->
> >> "participant
> >> sends"
> >>> 11) Section 4.2:
> >>> - F35. "of that streams" -> "that streams"
> >> Most Nit's fixed
> >>
> >> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>
> >>
> >> On 2014-01-29 10:29, Magnus Westerlund wrote:
> >>> Hi Partha and WG,
> >>>
> >>> I don't see any support for the changes you proposes in this
> >> discussion.
> >>> What I see some support for is to add a statement making clear =
that
> >>> there might be additional NAT/Firewall traversal mechanisms than
> >>> STUN/TURN. Simon proposed:
> >>>
> >>> "Note that TURN support being mandatory does not preclude a WebRTC
> >>> endpoint from supporting additional traversal mechanisms."
> >> I added this to section 3.3.4.1
> >>
> >>>
> >>> However, looking at the document as it is currently written, I am
> >>> uncertain where this would be added. The first mention of TURN is
> in
> >>> Section 3.3.4.1, and that section is focused on the global service
> >>> provider perspective and the need for location based provisioning
> of
> >>> NAT/Firewall traversal server resources.
> >>>
> >>> I think it can be added to Section 3.3.5.1 without being =
misplaced,
> >> but
> >>> it would be given a slightly narrower scope.
> >>>
> >>> I any of you want to be more explicit where this should be
> included,
> >>> please be. If you are not forthcoming I will request the authors =
to
> >> add
> >>> this in what they consider sensible position.
> >>>
> >>> Cheers
> >>>
> >>> Magnus
> >>>
> >>>
> >>> On 2014-01-25 17:46, Parthasarathi R wrote:
> >>>> Hi Simon,
> >>>>
> >>>>
> >>>>
> >>>> Thanks for your understanding about my firewall/NAT related
> problem
> >>>> statement in this draft.
> >>>>
> >>>>
> >>>>
> >>>> I have proposed the firewall/NAT related text by which the
> specific
> >>>> mechanism is not highlighted in the requirement document as there
> is
> >> no
> >>>> WG consensus for any of the mechanism including TURN. It is
> possible
> >> to
> >>>> argue hypothetically in PNTAW alias that PCP is the only =
mechanism
> >>>> required in WebRTC endpoint. Also, I=92m more interested in =
WebRTC
> >>>> gateway/server (Sec 4.3. of
> >>>> draft-ietf-rtcweb-use-cases-and-requirements-12) requirements
> >> wherein it
> >>>> is not required to support TURN and the related mail thread is
> >>>> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.
> >>>>
> >>>>
> >>>>
> >>>> IMO, my proposed text without mentioning any firewall/NAT
> mechanism
> >> in
> >>>> the requirement document helps to move forward without depend on
> the
> >>>> solution discussion in PNTAW alias.
> >>>>
> >>>>
> >>>>
> >>>> Thanks
> >>>>
> >>>> Partha
> >>>>
> >>>>
> >>>>
> >>>> *From:*Cb B [mailto:cb.list6@gmail.com]
> >>>> *Sent:* Saturday, January 25, 2014 6:22 AM
> >>>> *To:* Simon Perreault
> >>>> *Cc:* rtcweb@ietf.org; Parthasarathi R
> >>>> *Subject:* Re: [rtcweb] Query/Comment on
> >>>> draft-ietf-rtcweb-use-cases-and-requirements-12
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Jan 24, 2014 10:17 AM, "Simon Perreault"
> >> <simon.perreault@viagenie.ca
> >>>> <mailto:simon.perreault@viagenie.ca>> wrote:
> >>>>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :
> >>>>>
> >>>>>> Please note that when non-IETFers read this requirement
> document,
> >>>> they come
> >>>>>> to the conclusion that IETF RTCWeb WG recommends TURN and not
> >> other
> >>>>>> mechanisms. I'm saying that requirement document should not be
> >> used
> >>>> as the
> >>>>>> mechanism to eliminate the other alternatives when there is a
> >> discussion
> >>>>>> going-on in PNTAW alias. So, I'm asking for the change.
> >>>>>
> >>>>> I would totally agree with that sentiment, although I don't see
> >> your
> >>>> proposed text change reflecting it accurately. How about simply:
> >>>>> "Note that TURN support being mandatory does not preclude a
> WebRTC
> >>>> endpoint from supporting additional traversal mechanisms."
> >>>>>
> >>>> +1 for the above text.
> >>>>
> >>>> CB
> >>>>
> >>>>> Simon
> >>>>> --
> >>>>> DTN made easy, lean, and smart -->
> http://postellation.viagenie.ca
> >>>>> NAT64/DNS64 open-source --> http://ecdysis.viagenie.ca
> >>>>> STUN/TURN server --> http://numb.viagenie.ca
> >>>>> _______________________________________________
> >>>>> 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
> >>>>
> >>>
> >>
> >> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>
> >> On 2014-01-16 15:18, Hutton, Andrew wrote:
> >>> Some comments in the text below.
> >>>
> >>> As a general comment I suggest changing all the "FW" abbreviations
> to
> >> "Firewall" are have at least a first use definition.
> >>
> >> Done
> >>
> >>>
> >>> Andy
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >>>> Sent: 15 January 2014 10:20
> >>>> To: Chenxin (Xin); Hutton, Andrew; Christer Holmberg;
> Parthasarathi
> >> R;
> >>>> rtcweb@ietf.org
> >>>> Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-
> cases-
> >> and-
> >>>> requirements-12
> >>>>
> >>>> WG,
> >>>>
> >>>> It has been quite some time since the WG last call ended and a =
new
> >>>> revision was submitted. As Document Shepherd I want to push this
> >>>> document to publication request.
> >>>>
> >>>> Chenxin proposed below three different sets of changes to the
> >> document.
> >>>> Does the WG support making these changes? Please indicate within
> the
> >>>> next week if you support or want to reject these changes.
> >>>>
> >>>> Thanks
> >>>>
> >>>> Magnus
> >>>>
> >>>>
> >>>> On 2013-10-17 12:23, Chenxin (Xin) wrote:
> >>>>> Hi Andy,
> >>>>>
> >>>>>
> >>>>>
> >>>>> I think you means F29 not F27:). When I read it , I realize that
> >>>> there
> >>>>> is cross and ambiguous between 3.3.2 and 3.3.3
> >>>>>
> >>>>>
> >>>>>
> >>>>> More details:
> >>>>>
> >>>>>
> >>>>>
> >>>>> The topic of 3.3.2 is "Simple Video Communication Service,
> *NAT/FW*
> >>>>> that blocks UDP". But in the description and requirement, only
> >> *NAT*
> >>>> is
> >>>>> considered.
> >>>>>
> >>>>> The topic of 3.3.3 is "Simple Video Communication Service, FW
> that
> >>>>> only allows http", But only *http proxy* deployed scenarios is
> >>>> considered.
> >>>>>
> >>>>>
> >>>>> There are other usecases " FW block UDP, incoming TCP, Http
> >>>> allowing
> >>>>> FW without http proxy deplolyed under the permission of FW
> policy"
> >> ,
> >>>>> which is lost in the description. If we need consider these
> >> usecases
> >>>> , I
> >>>>> suggest to make some change to the description.
> >>>>>
> >>>>>
> >>>>>
> >>>>> Proposal 1 :
> >>>>>
> >>>>>
> >>>>>
> >>>>> add FW related words to section 3.3.2
> >>>>>
> >>>>> -------------------------------------------------
> >>>>>
> >>>>> 3.3.2. Simple Video Communication Service, NAT/FW that blocks =
UDP
> >>>>>
> >>>>>
> >>>>>
> >>>>> 3.3.2.1. Description
> >>>>>
> >>>>>
> >>>>>
> >>>>> This use-case is almost identical to the Simple Video
> >>>> Communication
> >>>>> Service use-case (Section 3.3.1). The difference is that one of
> >>>> the
> >>>>> users is behind a NAT*/FW* that blocks UDP traffic.
> >>>>>
> >>>>> .
> >>> [AndyH] - I support this change.
> >> Changed
> >>
> >>>
> >>>>>
> >>>>>
> >>>>> 3.3.2.2. Additional Requirements
> >>>>>
> >>>>>
> >>>>>
> >>>>> F29 The browser must be able to send streams and
> >>>>>
> >>>>> data to a peer in the presence of NATs *and FWs* that
> >>>>>
> >>>>> block UDP traffic ,* when FW policy allows WebRTC
> >>>> traffic*.
> >>> [AndyH] - I support this change.
> >> Changed
> >>
> >>>
> >>>>> -------------------------------------------------------
> >>>>>
> >>>>> Proposal 2: If the" Http allowing FW without http proxy =
deployed"
> >>>>> case is impliedly included in F29. I suggest to change the =
topics
> >> of
> >>>>> 3.3.3 to "Simple Video Communication Service, FW that only =
allows
> >>>>> traffic via a http proxy". So the 3.3.3 is a specific case.
> >>>>>
> >>> [AndyH] Yes the heading of 3.3.3 should have been changed during
> the
> >> last edit to "Simple Video Communication Service, FW that only
> allows
> >> traffic via a HTTP Proxy". I support this change.
> >>
> >> Changed.
> >>
> >>>
> >>>>>
> >>>>> Proposal 3: If " Http allowing FW without http proxy deployed"
> >>>> case
> >>>>> need to be explicitly mentioned. I suggest to add some
> descriptions
> >>>> to 3.3.3
> >>>>> =
-----------------------------------------------------------------
> >>>>>
> >>>>> 3.3.3. Simple Video Communication Service, FW that only allows
> http
> >>>>>
> >>>>>
> >>>>>
> >>>>> 3.3.3.1. Description
> >>>>>
> >>>>>
> >>>>>
> >>>>> This use-case is almost identical to the Simple Video
> >>>> Communication
> >>>>> Service use-case (Section 3.3.1). The difference is that one of
> >>>> the
> >>>>> users is behind a http allowing FW or a FW that only allows
> >>>> traffic
> >>>>> via a HTTP Proxy.
> >>>>>
> >>>>>
> >>> [AndyH] I don't believe the intention here was to state a
> requirement
> >> that WebRTC media should be able to flow through a FW only allowing
> >> HTTP. Therefore I think the original description is ok it is just
> the
> >> heading that needs to change.
> >>>
> >>>>> 3.3.3.2. Additional Requirements
> >>>>>
> >>>>>
> >>>>>
> >>>>> F37 The browser must be able to send streams and
> >>>>>
> >>>>> data to a peer in the presence of http allowing FWs or FWs
> >>>>> that only
> >>>>>
> >>>>> allows traffic via a HTTP Proxy, when FW policy
> >>>>>
> >>>>> allows WebRTC traffic.
> >>>>>
> >>> [AndyH] The existing text is in the 012 draft is ok and I don't
> think
> >> this change is needed as again I don't believe the intention here
> was
> >> to
> >> state a requirement that WebRTC media should be able to flow =
through
> a
> >> FW only allowing HTTP.
> >>>
> >>>>> =
-----------------------------------------------------------------
> --
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> Best Regards,
> >>>>>
> >>>>> Xin
> >>>>>
> >>>>>
> >>>>>
> >>>>
> >>>> --
> >>>>
> >>>> Magnus Westerlund
> >>>>
> >>>> =
------------------------------------------------------------------
> --
> >> --
> >>>> Services, Media and Network features, Ericsson Research EAB/TXM
> >>>> =
------------------------------------------------------------------
> --
> >> --
> >>>> Ericsson AB | Phone +46 10 7148287
> >>>> F=E4r=F6gatan 6 | Mobile +46 73 0949079
> >>>> SE-164 80 Stockholm, Sweden | mailto:
> magnus.westerlund@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
> >
>=20
> =3D


From Michael.Tuexen@lurchi.franken.de  Sat Feb  8 01:16:57 2014
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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On Feb 8, 2014, at 2:14 AM, "Makaraju, Maridi Raju (Raju)" =
<Raju.Makaraju@alcatel-lucent.com> wrote:

> That's good. Glad to see that no one liked the complex alternate =
option... yet! :-)
> I just put it there for completeness and some discussion.
> With "utf-8 on wire", which I too think is the right approach, there =
is some spec work todo:
> 1. Define IANA utf-8 ppid. Creating a new one is better instead of =
replacing existing DOMString as DOMString may be used by browsers for =
sometime to come and also to have backward compatibility/interworking.
> 2. Data channel core spec =
(http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-6.6)=
 need to specify explictly the wire format is utf-8.
Anu objections?

If not, I'll put corresponding text into the spec.

Best regards
Michael
>=20
> Then browsers need to do the conversion from utf-16 DOMstring to utf-8 =
before writing on wire also in the reverse direction.=20
>=20
> -Raju
>=20
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Friday, February 07, 2014 6:01 PM
>> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data =
Channel:
>> Usage of PPID for protocol multiplexing)
>>=20
>>=20
>> + 1
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>> ________________________________________
>> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
>> [martin.thomson@gmail.com]
>> Sent: Saturday, 08 February 2014 1:45 AM
>> To: Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data =
Channel:
>> Usage of PPID for protocol multiplexing)
>>=20
>> I don't think that we need any complication here.  It's a string.
>>=20
>> Strings are UTF-8 on the wire.
>>=20
>> Strings are UTF-16 (mostly) in JavaScript.
>>=20
>> Anything else would generate pain.
>>=20
>> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
>> <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>>> Also, I think "DOMString" PPID is too specific to Javascript, =
instead
>> it
>>>> should probably have a generic name like "UTF-16 String". The send =
API
>> can
>>>> still use DOMString as this is Javascript API.
>>>>> Any comments from others?
>>>>=20
>>>> Note: The WebSockets protocol defines the transferred strings as =
UTF-8.
>>>> http://tools.ietf.org/html/rfc6455#section-5.6
>>>>=20
>>>> As far as I can tell, we've always intended to follow that example.
>>>>=20
>>>> The fact that Javascript implementations currently choose to =
represent
>>>> text strings as UTF-16 at their API is sad, but not an argument for
>>>> sending that particular text representation over the wire, or =
reflecting
>>>> the name in the API.
>>>=20
>>> [Raju] I agree that using UTF-8 is desired and more appropriate! =
Then,
>> should the PPID be changed from "DOMString" to "UTF-8"? Javascript =
based
>> apps have to use some library to do the conversion of =
DOMString/UTF-16 to
>> UTF-8. Alternatively, browsers can do this conversion under the APIs =
(send
>> and onmessage) before sending and after receiving UTF-8 PPID data.
>>> Without such conversion webrtc interworking between browsers and =
native
>> clients will be problematic (basically, will not work).
>>>=20
>>> Another option, which is more flexible, is to define PPIDs for =
different
>> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text
>> instead of using "binary" directly); then pass this encoding =
information to
>> send() and onmessage() calls, which will use these PPIDs. Passing =
encoding
>> information might be implicit (like Javavscript DOMString) in most =
languages
>> by the type of the argument to send. This way it is upto the =
application to
>> deal with the encoding conversions as it see fits.
>>>=20
>>> The latter approach is best suitable for interworking between =
clients of
>> similar type (web or native); but it is bit painful (to do =
conversions) for
>> clients of different types.
>>>=20
>>> I am wondering what kind of API calls native WebRTC API stacks (e.g.
>> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) =
provide
>> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>>>=20
>>> -Raju
>>>=20
>>> _______________________________________________
>>> 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
>=20


From christer.holmberg@ericsson.com  Sat Feb  8 01:23:18 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Martin Thomson" <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
Thread-Index: AQHPI5G4gedSwhADR0Wr7uzX17tRu5qqSHkAgAAc9wCAABUk/4AAA9sAgACZNEg=
Date: Sat, 8 Feb 2014 09:23:12 +0000
Message-ID: <3au9wgasyo329a7x4xu2o08l.1391851390844@email.android.com>
References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFCF6C8@US70UWXCHMBA02.zam.alcatel-lucent.com> <B304F67A-9EA2-44A0-86DD-9DD0E21CB86F@lurchi.franken.de> <52F4182F.60404@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17826DFD40C3@US70UWXCHMBA02.zam.alcatel-lucent.com>, <CABkgnnXMYt7teSpxm7chTvQPR8ThKzKQ_bq3Po_FNFv2tdBFGQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D163B26@ESESSMB209.ericsson.se>, <E1FE4C082A89A246A11D7F32A95A17826DFD4581@US70UWXCHMBA02.zam.alcatel-lucent.com>
In-Reply-To: <E1FE4C082A89A246A11D7F32A95A17826DFD4581@US70UWXCHMBA02.zam.alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Hi,

I guess there is a good reason why the current PPID values have been regist=
ered to begin with, because normally any IANA registrations take place prio=
r to RFC publication.

But, I have no problem with defining a new value for utf-8, and keep the cu=
rrent values in the registry - as long as it is clear that they are depreca=
ted, and kept in the registry only for "historical" reasons (according to t=
he suggestion by Harald).

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

"Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com> wrote:


That's good. Glad to see that no one liked the complex alternate option... =
yet! :-)
I just put it there for completeness and some discussion.
With "utf-8 on wire", which I too think is the right approach, there is som=
e spec work todo:
1. Define IANA utf-8 ppid. Creating a new one is better instead of replacin=
g existing DOMString as DOMString may be used by browsers for sometime to c=
ome and also to have backward compatibility/interworking.
2. Data channel core spec (http://tools.ietf.org/html/draft-ietf-rtcweb-dat=
a-channel-06#section-6.6) need to specify explictly the wire format is utf-=
8.

Then browsers need to do the conversion from utf-16 DOMstring to utf-8 befo=
re writing on wire also in the reverse direction.

-Raju

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, February 07, 2014 6:01 PM
> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel=
:
> Usage of PPID for protocol multiplexing)
>
>
> + 1
>
> Regards,
>
> Christer
>
> ________________________________________
> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
> [martin.thomson@gmail.com]
> Sent: Saturday, 08 February 2014 1:45 AM
> To: Makaraju, Maridi Raju (Raju)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel=
:
> Usage of PPID for protocol multiplexing)
>
> I don't think that we need any complication here.  It's a string.
>
> Strings are UTF-8 on the wire.
>
> Strings are UTF-16 (mostly) in JavaScript.
>
> Anything else would generate pain.
>
> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
> <Raju.Makaraju@alcatel-lucent.com> wrote:
> >> >> Also, I think "DOMString" PPID is too specific to Javascript, inste=
ad
> it
> >> should probably have a generic name like "UTF-16 String". The send API
> can
> >> still use DOMString as this is Javascript API.
> >> > Any comments from others?
> >>
> >> Note: The WebSockets protocol defines the transferred strings as UTF-8=
.
> >> http://tools.ietf.org/html/rfc6455#section-5.6
> >>
> >> As far as I can tell, we've always intended to follow that example.
> >>
> >> The fact that Javascript implementations currently choose to represent
> >> text strings as UTF-16 at their API is sad, but not an argument for
> >> sending that particular text representation over the wire, or reflecti=
ng
> >> the name in the API.
> >
> > [Raju] I agree that using UTF-8 is desired and more appropriate! Then,
> should the PPID be changed from "DOMString" to "UTF-8"? Javascript based
> apps have to use some library to do the conversion of DOMString/UTF-16 to
> UTF-8. Alternatively, browsers can do this conversion under the APIs (sen=
d
> and onmessage) before sending and after receiving UTF-8 PPID data.
> > Without such conversion webrtc interworking between browsers and native
> clients will be problematic (basically, will not work).
> >
> > Another option, which is more flexible, is to define PPIDs for differen=
t
> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text
> instead of using "binary" directly); then pass this encoding information =
to
> send() and onmessage() calls, which will use these PPIDs. Passing encodin=
g
> information might be implicit (like Javavscript DOMString) in most langua=
ges
> by the type of the argument to send. This way it is upto the application =
to
> deal with the encoding conversions as it see fits.
> >
> > The latter approach is best suitable for interworking between clients o=
f
> similar type (web or native); but it is bit painful (to do conversions) f=
or
> clients of different types.
> >
> > I am wondering what kind of API calls native WebRTC API stacks (e.g.
> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) provide
> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
> >
> > -Raju
> >
> > _______________________________________________
> > 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 christer.holmberg@ericsson.com  Sat Feb  8 01:26:09 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
Thread-Index: AQHPI5G4gedSwhADR0Wr7uzX17tRu5qqSHkAgAAc9wCAABUk/4AAA9sAgACGp4CAABNZEA==
Date: Sat, 8 Feb 2014 09:26:03 +0000
Message-ID: <s2ej79huj4akcjl2tbtekvx8.1391851562242@email.android.com>
References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFCF6C8@US70UWXCHMBA02.zam.alcatel-lucent.com> <B304F67A-9EA2-44A0-86DD-9DD0E21CB86F@lurchi.franken.de> <52F4182F.60404@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17826DFD40C3@US70UWXCHMBA02.zam.alcatel-lucent.com>, <CABkgnnXMYt7teSpxm7chTvQPR8ThKzKQ_bq3Po_FNFv2tdBFGQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D163B26@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFD4581@US70UWXCHMBA02.zam.alcatel-lucent.com>, <23A98499-FA93-472A-B762-92BF72EAC2F1@lurchi.franken.de>
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Hi,

We could have an informative note somewhere.in the draft, pointing out the =
format used by JavaScript, if people think it would be useful.

Regards,

Christer


Sent from my Sony Ericsson Xperia arc S

Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:


On Feb 8, 2014, at 2:14 AM, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@a=
lcatel-lucent.com> wrote:

> That's good. Glad to see that no one liked the complex alternate option..=
. yet! :-)
> I just put it there for completeness and some discussion.
> With "utf-8 on wire", which I too think is the right approach, there is s=
ome spec work todo:
> 1. Define IANA utf-8 ppid. Creating a new one is better instead of replac=
ing existing DOMString as DOMString may be used by browsers for sometime to=
 come and also to have backward compatibility/interworking.
> 2. Data channel core spec (http://tools.ietf.org/html/draft-ietf-rtcweb-d=
ata-channel-06#section-6.6) need to specify explictly the wire format is ut=
f-8.
Anu objections?

If not, I'll put corresponding text into the spec.

Best regards
Michael
>
> Then browsers need to do the conversion from utf-16 DOMstring to utf-8 be=
fore writing on wire also in the reverse direction.
>
> -Raju
>
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Friday, February 07, 2014 6:01 PM
>> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channe=
l:
>> Usage of PPID for protocol multiplexing)
>>
>>
>> + 1
>>
>> Regards,
>>
>> Christer
>>
>> ________________________________________
>> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
>> [martin.thomson@gmail.com]
>> Sent: Saturday, 08 February 2014 1:45 AM
>> To: Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channe=
l:
>> Usage of PPID for protocol multiplexing)
>>
>> I don't think that we need any complication here.  It's a string.
>>
>> Strings are UTF-8 on the wire.
>>
>> Strings are UTF-16 (mostly) in JavaScript.
>>
>> Anything else would generate pain.
>>
>> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
>> <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>>> Also, I think "DOMString" PPID is too specific to Javascript, instea=
d
>> it
>>>> should probably have a generic name like "UTF-16 String". The send API
>> can
>>>> still use DOMString as this is Javascript API.
>>>>> Any comments from others?
>>>>
>>>> Note: The WebSockets protocol defines the transferred strings as UTF-8=
.
>>>> http://tools.ietf.org/html/rfc6455#section-5.6
>>>>
>>>> As far as I can tell, we've always intended to follow that example.
>>>>
>>>> The fact that Javascript implementations currently choose to represent
>>>> text strings as UTF-16 at their API is sad, but not an argument for
>>>> sending that particular text representation over the wire, or reflecti=
ng
>>>> the name in the API.
>>>
>>> [Raju] I agree that using UTF-8 is desired and more appropriate! Then,
>> should the PPID be changed from "DOMString" to "UTF-8"? Javascript based
>> apps have to use some library to do the conversion of DOMString/UTF-16 t=
o
>> UTF-8. Alternatively, browsers can do this conversion under the APIs (se=
nd
>> and onmessage) before sending and after receiving UTF-8 PPID data.
>>> Without such conversion webrtc interworking between browsers and native
>> clients will be problematic (basically, will not work).
>>>
>>> Another option, which is more flexible, is to define PPIDs for differen=
t
>> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text
>> instead of using "binary" directly); then pass this encoding information=
 to
>> send() and onmessage() calls, which will use these PPIDs. Passing encodi=
ng
>> information might be implicit (like Javavscript DOMString) in most langu=
ages
>> by the type of the argument to send. This way it is upto the application=
 to
>> deal with the encoding conversions as it see fits.
>>>
>>> The latter approach is best suitable for interworking between clients o=
f
>> similar type (web or native); but it is bit painful (to do conversions) =
for
>> clients of different types.
>>>
>>> I am wondering what kind of API calls native WebRTC API stacks (e.g.
>> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) provid=
e
>> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>>>
>>> -Raju
>>>
>>> _______________________________________________
>>> 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 Michael.Tuexen@lurchi.franken.de  Sat Feb  8 02:01:00 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFCF6C8@US70UWXCHMBA02.zam.alcatel-lucent.com> <B304F67A-9EA2-44A0-86DD-9DD0E21CB86F@lurchi.franken.de> <52F4182F.60404@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17826DFD40C3@US70UWXCHMBA02.zam.alcatel-lucent.com>, <CABkgnnXMYt7teSpxm7chTvQPR8ThKzKQ_bq3Po_FNFv2tdBFGQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D163B26@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFD4581@US70UWXCHMBA02.zam.alcatel-lucent.com>, <23A98499-FA93-472A-B762-92BF72EAC2F1@lurchi.franken.de> <s2ej79huj4akcjl2tbtekvx8.1391851562242@email.android.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
X-Mailer: Apple Mail (2.1510)
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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On Feb 8, 2014, at 10:26 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> We could have an informative note somewhere.in the draft, pointing out =
the format used by JavaScript, if people think it would be useful.
Hmm. I don't want to repeat what is in
http://heycam.github.io/webidl/#idl-DOMString
I like the idea to specify UTF-8 using a new PPID and deprecate the =
usage of
the current PPID for DOMString.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20
> Sent from my Sony Ericsson Xperia arc S
>=20
> Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>=20
>=20
> On Feb 8, 2014, at 2:14 AM, "Makaraju, Maridi Raju (Raju)" =
<Raju.Makaraju@alcatel-lucent.com> wrote:
>=20
>> That's good. Glad to see that no one liked the complex alternate =
option... yet! :-)
>> I just put it there for completeness and some discussion.
>> With "utf-8 on wire", which I too think is the right approach, there =
is some spec work todo:
>> 1. Define IANA utf-8 ppid. Creating a new one is better instead of =
replacing existing DOMString as DOMString may be used by browsers for =
sometime to come and also to have backward compatibility/interworking.
>> 2. Data channel core spec =
(http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-6.6)=
 need to specify explictly the wire format is utf-8.
> Anu objections?
>=20
> If not, I'll put corresponding text into the spec.
>=20
> Best regards
> Michael
>>=20
>> Then browsers need to do the conversion from utf-16 DOMstring to =
utf-8 before writing on wire also in the reverse direction.
>>=20
>> -Raju
>>=20
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: Friday, February 07, 2014 6:01 PM
>>> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
>>> Cc: rtcweb@ietf.org
>>> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data =
Channel:
>>> Usage of PPID for protocol multiplexing)
>>>=20
>>>=20
>>> + 1
>>>=20
>>> Regards,
>>>=20
>>> Christer
>>>=20
>>> ________________________________________
>>> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
>>> [martin.thomson@gmail.com]
>>> Sent: Saturday, 08 February 2014 1:45 AM
>>> To: Makaraju, Maridi Raju (Raju)
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data =
Channel:
>>> Usage of PPID for protocol multiplexing)
>>>=20
>>> I don't think that we need any complication here.  It's a string.
>>>=20
>>> Strings are UTF-8 on the wire.
>>>=20
>>> Strings are UTF-16 (mostly) in JavaScript.
>>>=20
>>> Anything else would generate pain.
>>>=20
>>> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
>>> <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>>>> Also, I think "DOMString" PPID is too specific to Javascript, =
instead
>>> it
>>>>> should probably have a generic name like "UTF-16 String". The send =
API
>>> can
>>>>> still use DOMString as this is Javascript API.
>>>>>> Any comments from others?
>>>>>=20
>>>>> Note: The WebSockets protocol defines the transferred strings as =
UTF-8.
>>>>> http://tools.ietf.org/html/rfc6455#section-5.6
>>>>>=20
>>>>> As far as I can tell, we've always intended to follow that =
example.
>>>>>=20
>>>>> The fact that Javascript implementations currently choose to =
represent
>>>>> text strings as UTF-16 at their API is sad, but not an argument =
for
>>>>> sending that particular text representation over the wire, or =
reflecting
>>>>> the name in the API.
>>>>=20
>>>> [Raju] I agree that using UTF-8 is desired and more appropriate! =
Then,
>>> should the PPID be changed from "DOMString" to "UTF-8"? Javascript =
based
>>> apps have to use some library to do the conversion of =
DOMString/UTF-16 to
>>> UTF-8. Alternatively, browsers can do this conversion under the APIs =
(send
>>> and onmessage) before sending and after receiving UTF-8 PPID data.
>>>> Without such conversion webrtc interworking between browsers and =
native
>>> clients will be problematic (basically, will not work).
>>>>=20
>>>> Another option, which is more flexible, is to define PPIDs for =
different
>>> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to =
text
>>> instead of using "binary" directly); then pass this encoding =
information to
>>> send() and onmessage() calls, which will use these PPIDs. Passing =
encoding
>>> information might be implicit (like Javavscript DOMString) in most =
languages
>>> by the type of the argument to send. This way it is upto the =
application to
>>> deal with the encoding conversions as it see fits.
>>>>=20
>>>> The latter approach is best suitable for interworking between =
clients of
>>> similar type (web or native); but it is bit painful (to do =
conversions) for
>>> clients of different types.
>>>>=20
>>>> I am wondering what kind of API calls native WebRTC API stacks =
(e.g.
>>> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) =
provide
>>> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>>>>=20
>>>> -Raju
>>>>=20
>>>> _______________________________________________
>>>> 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
>>=20
>=20
>=20


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To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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Hi Harald & all,

I agree with Harald that WebRTC Endpoints requirements has to be frozen =
before starting WebRTC Server/Gateway guidelines documents. I guess that =
draft-ietf-rtcweb-use-cases-and-requirements document open items will be =
closed during IETF-89 based on the current open items.

I'll write the strawman proposal after IETF-89 to start the discussion. =
I'm interested in hearing anybody else interested in contributing for =
the same.

Thanks
Partha

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Friday, February 07, 2014 11:34 PM
> To: Parthasarathi R; rtcweb@ietf.org
> Subject: Re: [rtcweb] ice-lite in WebRTC ???
>=20
> On 02/07/2014 05:26 PM, Parthasarathi R wrote:
> > <snip> > The server doesn't have to claim WebRTC conformance, it =
just
> has to
> >> claim to interoperate with WebRTC endpoints. </snip>
> > The above assumption leads to lot of confusion in WebRTC Server
> implementation which starts from RTCWeb requirement itself. IMO, it is
> better to provide the minimum IETF standard guidelines for WebRTC
> servers like ice-lite/ICE, BUNDLE required or not, ICE-TCP/TURN.
>=20
> It seems to me like a good document to write, but should probably be
> done after we finish the requirements for WebRTC endpoints.
>=20
> Unless someone wishes to start one, of course.


From sergio.garcia.murillo@gmail.com  Sat Feb  8 03:23:39 2014
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Subject: Re: [rtcweb] ice-lite in WebRTC ???
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Hi Partha,

I have quite some real life experience on webrtc server side 
implementation, so count on me.

Best regards
Sergio

El 08/02/2014 11:07, Parthasarathi R escribió:
> Hi Harald & all,
>
> I agree with Harald that WebRTC Endpoints requirements has to be frozen before starting WebRTC Server/Gateway guidelines documents. I guess that draft-ietf-rtcweb-use-cases-and-requirements document open items will be closed during IETF-89 based on the current open items.
>
> I'll write the strawman proposal after IETF-89 to start the discussion. I'm interested in hearing anybody else interested in contributing for the same.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Sent: Friday, February 07, 2014 11:34 PM
>> To: Parthasarathi R; rtcweb@ietf.org
>> Subject: Re: [rtcweb] ice-lite in WebRTC ???
>>
>> On 02/07/2014 05:26 PM, Parthasarathi R wrote:
>>> <snip> > The server doesn't have to claim WebRTC conformance, it just
>> has to
>>>> claim to interoperate with WebRTC endpoints. </snip>
>>> The above assumption leads to lot of confusion in WebRTC Server
>> implementation which starts from RTCWeb requirement itself. IMO, it is
>> better to provide the minimum IETF standard guidelines for WebRTC
>> servers like ice-lite/ICE, BUNDLE required or not, ICE-TCP/TURN.
>>
>> It seems to me like a good document to write, but should probably be
>> done after we finish the requirements for WebRTC endpoints.
>>
>> Unless someone wishes to start one, of course.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
Thread-Index: AQHPI5G5uqx5hsIOAEq9u24GTLgZVJqqTeJQgAB8IwCAAARsAP//vsSQgADcd4CAAAKUgP//7a+g
Date: Sat, 8 Feb 2014 13:45:05 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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> We could have an informative note somewhere.in the draft, pointing out th=
e
> format used by JavaScript, if people think it would be useful.

[Raju] The format used by Javascript is already documented in http://dev.w3=
.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescription
via various API calls:
    void send (DOMString data);
    void send (Blob data);
    void send (ArrayBuffer data);
    void send (ArrayBufferView data);

As long as browser+javascript is used by both webrtc clients the receiving =
gets these formats in onmessage () callback (NOTE: please see exceptions be=
low).
So, what goes/comes on wire is controlled by browser to give the applicatio=
n suitable format.=20
If one side is a non-browser/javascript webrtc client then the wire format =
really matters, for which the Data Channel Core spec will have a note (per =
discussion and agreement).

The spec is already assuming, though not explictly, onwire UTF-8 format whi=
le doing "bufferedAmount" calculations. This is evident from the note:

"bufferedAmount of type unsigned long, readonly
The bufferedAmount attribute must return the number of bytes of application=
 data (UTF-8 text and binary data) that have been queued using send() but t=
hat, as of the last time the event loop started executing a task, had not y=
et been transmitted to the network."

May be an additional explicit note in the above text about "onwire format i=
s UTF-8" makes it very clear!?

Coming back to http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTC=
SessionDescription :
As shown above the API has send() API which takes Blob or ArrayBuffer or Ar=
rayBufferView as arguments. Now since there will only be one "binary" PPID =
defined, all these maps to it. Assuming browser on the other end, how does =
it knows which type of data type to give in the onmessage() callback? Is th=
ere any expectation on the other end it gets same data type? I guess not! O=
ther end may always default to using "Blob". In either case, we need a note=
 here about the data type to be expected in onmessage(), which may always b=
e "Blob"/binary or "DOMString"/text?!


-Raju

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Michael Tuexen" <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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References: <7594FB04B1934943A5C02806D1A2204B1D15E955@ESESSMB209.ericsson.se> <74072016-DA11-41E8-9944-779428163EE4@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D15ED94@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFCF6C8@US70UWXCHMBA02.zam.alcatel-lucent.com> <B304F67A-9EA2-44A0-86DD-9DD0E21CB86F@lurchi.franken.de> <52F4182F.60404@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17826DFD40C3@US70UWXCHMBA02.zam.alcatel-lucent.com>, <CABkgnnXMYt7teSpxm7chTvQPR8ThKzKQ_bq3Po_FNFv2tdBFGQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D163B26@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFD4581@US70UWXCHMBA02.zam.alcatel-lucent.com>, <23A98499-FA93-472A-B762-92BF72EAC2F1@lurchi.franken.de> <s2ej79huj4akcjl2tbtekvx8.1391851562242@email.android.com>, <E1FE4C082A89A246A11D7F32A95A17826DFD4E9D@US70UWXCHMBA02.zam.alcatel-lucent.com>
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Hi,

Without commenting on specifics: we shall NOT have a mechanism where one si=
de needs to know whether the remote peer is a browswer or not. The data cha=
nnel mechanism itself shall be generic.

Regards,

Christer



________________________________________
From: Makaraju, Maridi Raju (Raju) [Raju.Makaraju@alcatel-lucent.com]
Sent: Saturday, 08 February 2014 3:45 PM
To: Christer Holmberg; Michael Tuexen
Cc: Martin Thomson; rtcweb@ietf.org
Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: =
Usage of PPID for protocol multiplexing)

> We could have an informative note somewhere.in the draft, pointing out th=
e
> format used by JavaScript, if people think it would be useful.

[Raju] The format used by Javascript is already documented in http://dev.w3=
.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescription
via various API calls:
    void send (DOMString data);
    void send (Blob data);
    void send (ArrayBuffer data);
    void send (ArrayBufferView data);

As long as browser+javascript is used by both webrtc clients the receiving =
gets these formats in onmessage () callback (NOTE: please see exceptions be=
low).
So, what goes/comes on wire is controlled by browser to give the applicatio=
n suitable format.
If one side is a non-browser/javascript webrtc client then the wire format =
really matters, for which the Data Channel Core spec will have a note (per =
discussion and agreement).

The spec is already assuming, though not explictly, onwire UTF-8 format whi=
le doing "bufferedAmount" calculations. This is evident from the note:

"bufferedAmount of type unsigned long, readonly
The bufferedAmount attribute must return the number of bytes of application=
 data (UTF-8 text and binary data) that have been queued using send() but t=
hat, as of the last time the event loop started executing a task, had not y=
et been transmitted to the network."

May be an additional explicit note in the above text about "onwire format i=
s UTF-8" makes it very clear!?

Coming back to http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTC=
SessionDescription :
As shown above the API has send() API which takes Blob or ArrayBuffer or Ar=
rayBufferView as arguments. Now since there will only be one "binary" PPID =
defined, all these maps to it. Assuming browser on the other end, how does =
it knows which type of data type to give in the onmessage() callback? Is th=
ere any expectation on the other end it gets same data type? I guess not! O=
ther end may always default to using "Blob". In either case, we need a note=
 here about the data type to be expected in onmessage(), which may always b=
e "Blob"/binary or "DOMString"/text?!


-Raju

From Raju.Makaraju@alcatel-lucent.com  Sat Feb  8 08:38:43 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "christer.holmberg@ericsson.com" <christer.holmberg@ericsson.com>, "michael.tuexen@lurchi.franken.de" <michael.tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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Hi Christer, et. al.,
I completely agree and that's the intention of having just two PPIDs, one f=
or text and one for binary. However, it would be good to have the clarifica=
tions that I mentioned in my earlier reply.
I have a general comment on some of the non-browser/javascript webrtc specs=
 assuming, unnecessarily, browser+javascript as the client, which leads to =
some confusion like utf-8 vs DOMString confusion. It would be nice if these=
 specs make it generic to apply all webrtc clients: either browser, native =
or hybrid. In future I will point these places as I review the relevant spe=
cs. But it would be great if authors keep this mind.

Regards
-Raju

----- Reply message -----
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Sat, Feb 8, 2014 10:00 am
Subject: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usag=
e of PPID for protocol multiplexing)
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Mic=
hael Tuexen" <Michael.Tuexen@lurchi.franken.de>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>


Hi,

Without commenting on specifics: we shall NOT have a mechanism where one si=
de needs to know whether the remote peer is a browswer or not. The data cha=
nnel mechanism itself shall be generic.

Regards,

Christer



________________________________________
From: Makaraju, Maridi Raju (Raju) [Raju.Makaraju@alcatel-lucent.com]
Sent: Saturday, 08 February 2014 3:45 PM
To: Christer Holmberg; Michael Tuexen
Cc: Martin Thomson; rtcweb@ietf.org
Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: =
Usage of PPID for protocol multiplexing)

> We could have an informative note somewhere.in the draft, pointing out th=
e
> format used by JavaScript, if people think it would be useful.

[Raju] The format used by Javascript is already documented in http://dev.w3=
.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescription
via various API calls:
    void send (DOMString data);
    void send (Blob data);
    void send (ArrayBuffer data);
    void send (ArrayBufferView data);

As long as browser+javascript is used by both webrtc clients the receiving =
gets these formats in onmessage () callback (NOTE: please see exceptions be=
low).
So, what goes/comes on wire is controlled by browser to give the applicatio=
n suitable format.
If one side is a non-browser/javascript webrtc client then the wire format =
really matters, for which the Data Channel Core spec will have a note (per =
discussion and agreement).

The spec is already assuming, though not explictly, onwire UTF-8 format whi=
le doing "bufferedAmount" calculations. This is evident from the note:

"bufferedAmount of type unsigned long, readonly
The bufferedAmount attribute must return the number of bytes of application=
 data (UTF-8 text and binary data) that have been queued using send() but t=
hat, as of the last time the event loop started executing a task, had not y=
et been transmitted to the network."

May be an additional explicit note in the above text about "onwire format i=
s UTF-8" makes it very clear!?

Coming back to http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTC=
SessionDescription :
As shown above the API has send() API which takes Blob or ArrayBuffer or Ar=
rayBufferView as arguments. Now since there will only be one "binary" PPID =
defined, all these maps to it. Assuming browser on the other end, how does =
it knows which type of data type to give in the onmessage() callback? Is th=
ere any expectation on the other end it gets same data type? I guess not! O=
ther end may always default to using "Blob". In either case, we need a note=
 here about the data type to be expected in onmessage(), which may always b=
e "Blob"/binary or "DOMString"/text?!


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

--_000_0AD62566EF8C4E2B85D3CA0C33536906alcatellucentcom_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
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<div>Hi Christer, et. al.,<br>
I completely agree and that's the intention of having just two PPIDs, one f=
or text and one for binary. However, it would be good to have the clarifica=
tions that I mentioned in my earlier reply.<br>
I have a general comment on some of the non-browser/javascript webrtc specs=
 assuming, unnecessarily, browser&#43;javascript as the client, which leads=
 to some confusion like utf-8 vs DOMString confusion. It would be nice if t=
hese specs make it generic to apply
 all webrtc clients: either browser, native or hybrid. In future I will poi=
nt these places as I review the relevant specs. But it would be great if au=
thors keep this mind.<br>
<br>
Regards<br>
-Raju<br>
<br>
<div id=3D"x_htc_header" style=3D"">----- Reply message -----<br>
From: &quot;Christer Holmberg&quot; &lt;christer.holmberg@ericsson.com&gt;<=
br>
Date: Sat, Feb 8, 2014 10:00 am<br>
Subject: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usag=
e of PPID for protocol multiplexing)<br>
To: &quot;Makaraju, Maridi Raju (Raju)&quot; &lt;Raju.Makaraju@alcatel-luce=
nt.com&gt;, &quot;Michael Tuexen&quot; &lt;Michael.Tuexen@lurchi.franken.de=
&gt;<br>
Cc: &quot;rtcweb@ietf.org&quot; &lt;rtcweb@ietf.org&gt;<br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
Hi,<br>
<br>
Without commenting on specifics: we shall NOT have a mechanism where one si=
de needs to know whether the remote peer is a browswer or not. The data cha=
nnel mechanism itself shall be generic.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
<br>
________________________________________<br>
From: Makaraju, Maridi Raju (Raju) [Raju.Makaraju@alcatel-lucent.com]<br>
Sent: Saturday, 08 February 2014 3:45 PM<br>
To: Christer Holmberg; Michael Tuexen<br>
Cc: Martin Thomson; rtcweb@ietf.org<br>
Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: =
Usage of PPID for protocol multiplexing)<br>
<br>
&gt; We could have an informative note somewhere.in the draft, pointing out=
 the<br>
&gt; format used by JavaScript, if people think it would be useful.<br>
<br>
[Raju] The format used by Javascript is already documented in <a href=3D"ht=
tp://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescriptio=
n">
http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescript=
ion</a><br>
via various API calls:<br>
&nbsp;&nbsp;&nbsp; void send (DOMString data);<br>
&nbsp;&nbsp;&nbsp; void send (Blob data);<br>
&nbsp;&nbsp;&nbsp; void send (ArrayBuffer data);<br>
&nbsp;&nbsp;&nbsp; void send (ArrayBufferView data);<br>
<br>
As long as browser&#43;javascript is used by both webrtc clients the receiv=
ing gets these formats in onmessage () callback (NOTE: please see exception=
s below).<br>
So, what goes/comes on wire is controlled by browser to give the applicatio=
n suitable format.<br>
If one side is a non-browser/javascript webrtc client then the wire format =
really matters, for which the Data Channel Core spec will have a note (per =
discussion and agreement).<br>
<br>
The spec is already assuming, though not explictly, onwire UTF-8 format whi=
le doing &quot;bufferedAmount&quot; calculations. This is evident from the =
note:<br>
<br>
&quot;bufferedAmount of type unsigned long, readonly<br>
The bufferedAmount attribute must return the number of bytes of application=
 data (UTF-8 text and binary data) that have been queued using send() but t=
hat, as of the last time the event loop started executing a task, had not y=
et been transmitted to the network.&quot;<br>
<br>
May be an additional explicit note in the above text about &quot;onwire for=
mat is UTF-8&quot; makes it very clear!?<br>
<br>
Coming back to <a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html#=
idl-def-RTCSessionDescription">
http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSessionDescript=
ion</a> :<br>
As shown above the API has send() API which takes Blob or ArrayBuffer or Ar=
rayBufferView as arguments. Now since there will only be one &quot;binary&q=
uot; PPID defined, all these maps to it. Assuming browser on the other end,=
 how does it knows which type of data type
 to give in the onmessage() callback? Is there any expectation on the other=
 end it gets same data type? I guess not! Other end may always default to u=
sing &quot;Blob&quot;. In either case, we need a note here about the data t=
ype to be expected in onmessage(), which may
 always be &quot;Blob&quot;/binary or &quot;DOMString&quot;/text?!<br>
<br>
<br>
-Raju<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>

--_000_0AD62566EF8C4E2B85D3CA0C33536906alcatellucentcom_--

From randell-ietf@jesup.org  Sat Feb  8 12:24:10 2014
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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On 2/7/2014 6:45 PM, Martin Thomson wrote:
> I don't think that we need any complication here.  It's a string.
>
> Strings are UTF-8 on the wire.
>
> Strings are UTF-16 (mostly) in JavaScript.
>
> Anything else would generate pain.

(Back online while traveling; been off mostly due to extended power 
outage in Philadelphia suburbs)

100% agree.  And currently that's what we do in Firefox at the 
JS->protocol interface:

void
nsDOMDataChannel::Send(const nsAString& aData, ErrorResult& aRv)
{
   NS_ConvertUTF16toUTF8 msgString(aData);
   Send(nullptr, msgString, msgString.Length(), false, aRv);
}

The JS interface matches WebSockets Send().  The wireline interface is 
(and should be) UTF8; any registrations should be updated to make this 
clear.

-- 
Randell Jesup
randell-ietf@jesup.org


From Michael.Tuexen@lurchi.franken.de  Sat Feb  8 12:35:38 2014
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On Feb 8, 2014, at 9:24 PM, Randell Jesup <randell-ietf@jesup.org> =
wrote:

> On 2/7/2014 6:45 PM, Martin Thomson wrote:
>> I don't think that we need any complication here.  It's a string.
>>=20
>> Strings are UTF-8 on the wire.
>>=20
>> Strings are UTF-16 (mostly) in JavaScript.
>>=20
>> Anything else would generate pain.
>=20
> (Back online while traveling; been off mostly due to extended power =
outage in Philadelphia suburbs)
>=20
> 100% agree.  And currently that's what we do in Firefox at the =
JS->protocol interface:
>=20
> void
> nsDOMDataChannel::Send(const nsAString& aData, ErrorResult& aRv)
> {
>  NS_ConvertUTF16toUTF8 msgString(aData);
>  Send(nullptr, msgString, msgString.Length(), false, aRv);
> }
>=20
> The JS interface matches WebSockets Send().  The wireline interface is =
(and should be) UTF8; any registrations should be updated to make this =
clear.
Great. I'll update the spec. Let's change the name of the PPID...

Best regards
Michael
>=20
> --=20
> Randell Jesup
> randell-ietf@jesup.org
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From pkyzivat@alum.mit.edu  Sun Feb  9 10:38:00 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel: Usage of PPID for protocol multiplexing
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On 2/6/14 6:19 PM, Harald Alvestrand wrote:
> On 02/06/2014 09:47 PM, Michael Tuexen wrote:
>> On Feb 6, 2014, at 9:35 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>
>>> Hi,
>>>
>>>>>>> I hope that if values are retained simply for backward compatibility they are explicitly marked as deprecated.
>>>>>> My plan was just not to defined what they are used for...
>>>>> I would like to have some explanation text. Having registered values without any kind of explanation is not a good approach, in my opinion.
>>>> The reason why I registered them is that they are used by Firefox and there was
>>>> the possibility that the PPID based fragmentation and reassembly got
>>>> standardised. I wanted to avoid the case that Firefox uses some values
>>>> and other protocols register these values. That would have been pretty
>>>> bad. Since they are a very cheap resource, I just went ahead and "saved"
>>>> the values.
>>> That is a all good. But, when people wonder 5 years from now, I don't want them to have to look for this e-mail in oder to get an explanation :)
>> Understood. But do you think they walk through the IANA registry? I would expect them
>> to read RFCs and follow the procedures described there.
>
> Updating the registry to read "Obsolete; was used by <draft name>"
> should satisfy all comers.
>
> People do read registries.

Yes. In some cases the registry is how you discover what RFC to read.

Maybe five years from now, somebody debugging some app gets one of these 
values, and doesn't know what it means. It won't be mentioned in any of 
the current RFCs he has read, because none of them use it.

So he ends up looking in the registry to find where it is defined.

	Thanks,
	Paul


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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
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-protocol-02.txt
	Pages           : 12
	Date            : 2014-02-09

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 a simple protocol for establishing
   symmetric data channels between the peers.  It uses a two way
   handshake and allows sending of user data without waiting for the
   handshake to complete.


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-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-data-protocol-01
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On Nov 12, 2013, at 2:45 PM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> Hi,
>=20
> I did review the Data Channel Protocol on my way to IETF, and here are
> my comments.
Hi Magnus,

thank you very much for the comments. I've submitted a new version =
according
to the comments below.
>=20
> 1. Section 4:
>=20
>   o  the outgoing SCTP stream.
>=20
> What about incoming SCTP stream?
Changed to "the SCTP streams."
>=20
> 2. Section 4:
>   The data channel protocol uses a two way handshake to open a data
>   channel.  The side wanting to open a data channel selects an unused
>   Stream and sends a DATA_CHANNEL_OPEN message.  The peer responds =
with
>   a DATA_CHANNEL_ACK message.  Then the data channel is open.  Please
>   note that the opening side can send user messages before the
>   DATA_CHANNEL_ACK is received.  These data channel messages are sent
>   on the same Stream as the user messages belonging to the data
>   channel.  The demultiplexing is based on the SCTP payload protocol
>   identifier.
>=20
> I find this paragraph a bit confusing on the following things:
>=20
> A. Is the DATA_CHANNEL_OPEN message sent on that selected stream?
> B. Is the DATA_CHANNEL_ACK sent on the peers corresponding stream =
number
> as the DATA_CHANNEL_OPEN was received on?
> C. "These data channel messages" does this refer to
> DATA_CHANNEL_OPEN/ACK, maybe be more explicit.
> D. "The demultiplexing is based on the SCTP payload protocol
>   identifier." Do you mean PPIDs here. And could you be more explicit
> about of how this works. Am I wrong in the understanding that you have
> have one PPID for this control protocol. But, you are not explicit =
about
> if which PPID user messages could use.
OK, I have updated the above paragraph:

<t>The data channel protocol uses a two way handshake to open a data =
channel
by combining two SCTP streams, one in each direction, with the same
SCTP stream identifier.
The side wanting to open a data channel selects an SCTP stream =
identifier
for which the corresponding incoming and outgoing SCTP stream is unused
and sends a DATA_CHANNEL_OPEN message on this outgoing SCTP stream.
The peer responds with a DATA_CHANNEL_ACK message on its corresponding
outgoing SCTP stream. Then the data channel is open.
Please note that the opening side can send user messages before the
DATA_CHANNEL_ACK is received.
Data channel control messages are sent on the same Stream as the user
messages belonging to the data channel.
The demultiplexing is based on the SCTP payload protocol identifier =
(PPID),
since the data channel control protocol uses a specific PPID.</t>
>=20
> 3. Section 4:
>   The protocol field is to ease cross-application interoperation
>   ("federation") by identifying the data being passed with an IANA-
>   registered string, and may be useful for homogenous applications
>   which may create more than one type of Channel.
>=20
> One more case where I think it is fuzzy what is refered to. Is the
> "protocol field" an identifier for the PPID to use, or something else?
No, it is something different.
> What type of IANA registered string is being used here? Which registry
> and field value are referenced?
The one defined in Section 8.4.
>=20
> 4. Section 5.1:
>   Priority: 2 bytes (integer)
>      The priority of the channel.
>=20
> What is the definition of the priority field value?
It is the priority of the channel. I changed the text to:

The higher the number, the lower the priority. In particular,
0 is the highest priority.

However, I'm not sure if this should be unsigned. Also the W3C document
doesn't specify it yet.
>=20
> 5. Section 5.1:
>   Reliability Parameter: 4 bytes
>=20
> I would recommend that you make a table under this parameter where you
> make clear the interpretation of the field under differetn channel =
types.
OK, I have added:

The following table summarises this:</t>
</list></t>
<texttable>
<ttcol align=3D'left'>Channel Type</ttcol>
<ttcol align=3D'center'>Reliability Parameter</ttcol>
<c>DATA_CHANNEL_RELIABLE</c>                         <c>Ignored</c>
<c>DATA_CHANNEL_RELIABLE_UNORDERED</c>               <c>Ignored</c>
<c>DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT</c>          <c>Number of =
RTX</c>
<c>DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED</c><c>Number of =
RTX</c>
<c>DATA_CHANNEL_PARTIAL_RELIABLE_TIMED</c>           <c>Lifetime in =
ms</c>
<c>DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED</c> <c>Lifetime in =
ms</c>
</texttable>
>=20
> 6. Section 5.1
>=20
>   Protocol: Variable Length (sequence of characters)
>      The protocol for the channel.  This may be an empty string.  If
>      used, it is an IANA-registered protocol (see Section 8.4).
>=20
> Can you please be explicit about which IANA registry that is relevant
> here. What is the meaning of an empty string?
There is already a reference to the registry (Section 8.4). How
can it be more explicit?
If you use an empty string it is unspecified. I have changed it to:

The protocol for the channel. If this is an empty string the
protocol us unspecified. If it is an non-empty string, it specifies
an IANA-registered protocol (see <xref target=3D'iana_protocol'/>).

>=20
> 7. Section 6.
> What PPID shall be used when sending the user data?
The document describes a protocol for opening a data channel (see first =
sentence
of section 4), not for using or closing it. Sending user data is
done the same on all data channel, not matter how they are set up. That
is why sending user data is described in draft-ietf-rtcweb-data-channel.
>=20
> 8. Section 6.
> Therefore, receiving an
>   SCTP stream reset request for a stream on which no DATA_CHANNEL_ACK
>   message has been received indicates to the sender of the
>   corresponding DATA_CHANNEL_OPEN message the failure of the data
>   channel setup procedure.
>=20
> What is allowed to do after having received the reset on a
> DATA_CHANNEL_OPEN? Can one try to send a new OPEN?
Sure. I added:

After also successfully resetting the corresponding outgoing
SCTP stream, a new DATA_CHANNEL_OPEN message can be sent on the stream.

>=20
> 9. Section 8.2:
>               | Reserved          | 0x00      | [RFCXXXX] |
>               | Reserved          | 0x01      | [RFCXXXX] |
>=20
> Why is these values reserved, please include the motivation for this =
in
> the description of the registry.
OK. I have added:

<t>Please note that the values 0x00 and 0x01 are reserved to avoid
interoperability problems, since they have been used in earlier versions
of the document.</t>

>=20
> 10. Section 8.4:
>   IANA is requested to create a new registration table "Protocol
>   Registry" for the data channel protocol to manage the "Protocol"
>   field of type string in DATA_CHANNEL_OPEN messages (see Section =
5.1).
>=20
> "Protocol Registry" is a lousy name on a registry. Secondly, I think =
the
I agree that it is a lousy name, but I couldn't come up with a better =
one.
Maybe Randell can? Or you can?
> purpose of this registry needs to be better described.
Randell?

Best regards
Michael
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20


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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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On 02/08/2014 02:14 AM, Makaraju, Maridi Raju (Raju) wrote:
> That's good. Glad to see that no one liked the complex alternate option... yet! :-)
> I just put it there for completeness and some discussion.
> With "utf-8 on wire", which I too think is the right approach, there is some spec work todo:
> 1. Define IANA utf-8 ppid. Creating a new one is better instead of replacing existing DOMString as DOMString may be used by browsers for sometime to come and also to have backward compatibility/interworking.

Have you ever seen UTF-16 on the wire in the DataChannel protocol?

> 2. Data channel core spec (http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-06#section-6.6) need to specify explictly the wire format is utf-8.

Fully agreed.

>
> Then browsers need to do the conversion from utf-16 DOMstring to utf-8 before writing on wire also in the reverse direction. 
>
> -Raju
>
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Friday, February 07, 2014 6:01 PM
>> To: Martin Thomson; Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: RE: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel:
>> Usage of PPID for protocol multiplexing)
>>
>>
>> + 1
>>
>> Regards,
>>
>> Christer
>>
>> ________________________________________
>> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Martin Thomson
>> [martin.thomson@gmail.com]
>> Sent: Saturday, 08 February 2014 1:45 AM
>> To: Makaraju, Maridi Raju (Raju)
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel:
>> Usage of PPID for protocol multiplexing)
>>
>> I don't think that we need any complication here.  It's a string.
>>
>> Strings are UTF-8 on the wire.
>>
>> Strings are UTF-16 (mostly) in JavaScript.
>>
>> Anything else would generate pain.
>>
>> On 7 February 2014 14:01, Makaraju, Maridi Raju (Raju)
>> <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>>> Also, I think "DOMString" PPID is too specific to Javascript, instead
>> it
>>>> should probably have a generic name like "UTF-16 String". The send API
>> can
>>>> still use DOMString as this is Javascript API.
>>>>> Any comments from others?
>>>> Note: The WebSockets protocol defines the transferred strings as UTF-8.
>>>> http://tools.ietf.org/html/rfc6455#section-5.6
>>>>
>>>> As far as I can tell, we've always intended to follow that example.
>>>>
>>>> The fact that Javascript implementations currently choose to represent
>>>> text strings as UTF-16 at their API is sad, but not an argument for
>>>> sending that particular text representation over the wire, or reflecting
>>>> the name in the API.
>>> [Raju] I agree that using UTF-8 is desired and more appropriate! Then,
>> should the PPID be changed from "DOMString" to "UTF-8"? Javascript based
>> apps have to use some library to do the conversion of DOMString/UTF-16 to
>> UTF-8. Alternatively, browsers can do this conversion under the APIs (send
>> and onmessage) before sending and after receiving UTF-8 PPID data.
>>> Without such conversion webrtc interworking between browsers and native
>> clients will be problematic (basically, will not work).
>>> Another option, which is more flexible, is to define PPIDs for different
>> encodings like "UTF-8", "UTF-16" (or even "base64" for binary to text
>> instead of using "binary" directly); then pass this encoding information to
>> send() and onmessage() calls, which will use these PPIDs. Passing encoding
>> information might be implicit (like Javavscript DOMString) in most languages
>> by the type of the argument to send. This way it is upto the application to
>> deal with the encoding conversions as it see fits.
>>> The latter approach is best suitable for interworking between clients of
>> similar type (web or native); but it is bit painful (to do conversions) for
>> clients of different types.
>>> I am wondering what kind of API calls native WebRTC API stacks (e.g.
>> Google Chrome's http://www.webrtc.org/webrtc-native-code-package) provide
>> for data channel send/onmessage ()? UTF-16 strings? Or UTF-8?
>>> -Raju
>>>
>>> _______________________________________________
>>> 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


-- 
Surveillance is pervasive. Go Dark.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Subject: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Hi,

As is defined in the data channel protocol, an entity can send data once DA=
TA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is re=
ceived.

The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at the s=
ame time, using different stream id values, the result may be TWO data chan=
nels, with data sent on both.

However, as is also indicated, the draft does not provide a mechanism to pr=
event/handle such glare situation.

I personally think there shall be a way to prevent such situation, or other=
wise I'm sure we are going to end up with interoperability problems.

Couldn't we e.g. say that, by default, the DTLS client is responsible for s=
ending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it could=
 even use both an even or odd stream id value.

Regards,

Christer



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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As is defined in the data channel protocol, an entit=
y can send data once DATA_CHANNEL_OPEN has been sent, before the associated=
 DATA_CHANNEL_ACK is received.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft also says that, if both endpoints send DAT=
A_CHANNEL_OPEN at the same time, using different stream id values, the resu=
lt may be TWO data channels, with data sent on both.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">However, as is also indicated, the draft does not pr=
ovide a mechanism to prevent/handle such glare situation.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I personally think there shall be a way to prevent s=
uch situation, or otherwise I&#8217;m sure we are going to end up with inte=
roperability problems.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Couldn&#8217;t we e.g. say that, by default, the DTL=
S client is responsible for sending the DATA_CHANNEL_OPEN? Then there is no=
 risk for glare, and it could even use both an even or odd stream id value.=
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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From: Barry Dingle <btdingle@gmail.com>
Date: Mon, 10 Feb 2014 21:07:55 +1100
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-02.txt
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Protocol looks good.

One small point - the title of this protocol is "WebRTC Data Channel
Protocol" but the Abstract says "*This document specifies a simple protocol
for establishing symmetric data channels between the peers.*"  This could
cause confusion between what is a Data Channel Protocol and what is an
Establishing protocol.

It appears that the title of this protocol should be "WebRTC Data
Channel *Establishment
*Protocol" in order to avoid future confusion as to its purpose.

/Barry Dingle
Australia


On Mon, Feb 10, 2014 at 8:16 AM, <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
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-02.txt
>         Pages           : 12
>         Date            : 2014-02-09
>
> 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 a simple protocol for establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>
>
> 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-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-02
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> 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"><div>Protocol looks good. <br><br>One small point - the ti=
tle of this protocol is  &quot;WebRTC Data Channel Protocol&quot; but the A=
bstract says &quot;<i>This document specifies a simple protocol for <u>esta=
blishing</u>
   symmetric data channels between the peers.</i>&quot;=C2=A0 This could ca=
use confusion between what is a Data Channel Protocol and what is an Establ=
ishing protocol. <br><br>It appears that the title of this protocol should =
be &quot;WebRTC Data Channel <b>Establishment </b>Protocol&quot; in order t=
o avoid future confusion as to its purpose. <br>

<br></div><div class=3D"gmail_extra">/Barry Dingle<br><div>Australia</div>
<br><br><div class=3D"gmail_quote">On Mon, Feb 10, 2014 at 8:16 AM,  <span =
dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blan=
k">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft: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 Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Randell J=
esup<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-02.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 12<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2014-02-09<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 a simple protocol for =
establishing<br>
=C2=A0 =C2=A0symmetric data channels between the peers. =C2=A0It uses a two=
 way<br>
=C2=A0 =C2=A0handshake and allows sending of user data without waiting for =
the<br>
=C2=A0 =C2=A0handshake to complete.<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-dat=
a-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-02" t=
arget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol=
-02</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-protoc=
ol-02" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcw=
eb-data-protocol-02</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<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></div>

--bcaec53d5a8ba6e2b104f20a849e--

From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 02:11:56 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-02.txt
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On Feb 10, 2014, at 11:07 AM, Barry Dingle <btdingle@gmail.com> wrote:

> Protocol looks good.=20
>=20
> One small point - the title of this protocol is "WebRTC Data Channel =
Protocol" but the Abstract says "This document specifies a simple =
protocol for establishing symmetric data channels between the peers."  =
This could cause confusion between what is a Data Channel Protocol and =
what is an Establishing protocol.=20
>=20
> It appears that the title of this protocol should be "WebRTC Data =
Channel Establishment Protocol" in order to avoid future confusion as to =
its purpose.=20
You are right, the protocol is just for establishing data channels.
Other opinions regarding the name change?

Best regards
Michael
>=20
> /Barry Dingle
> Australia
>=20
>=20
> On Mon, Feb 10, 2014 at 8:16 AM, <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
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-02.txt
>         Pages           : 12
>         Date            : 2014-02-09
>=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 a simple protocol for =
establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>=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-02
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protocol-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=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


From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 02:17:05 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On Feb 10, 2014, at 11:07 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> As is defined in the data channel protocol, an entity can send data =
once DATA_CHANNEL_OPEN has been sent, before the associated =
DATA_CHANNEL_ACK is received.
> =20
> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at =
the same time, using different stream id values, the result may be TWO =
data channels, with data sent on both.
> =20
> However, as is also indicated, the draft does not provide a mechanism =
to prevent/handle such glare situation.
> =20
> I personally think there shall be a way to prevent such situation, or =
otherwise I=92m sure we are going to end up with interoperability =
problems.
> =20
> Couldn=92t we e.g. say that, by default, the DTLS client is =
responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for =
glare, and it could even use both an even or odd stream id value.
Section 4 reads:

   To avoid glare in opening Channels, each side MUST use either even or
   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers.

So there is no problem in setting up data channels.

This should not be confused by both sides setting up a data channel with
the same Label. This ends up in two data channels with the same label.
You can even create more data channels with the same label.

Best regards
Michael
> =20
> Regards,
> =20
> Christer
> =20
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Mon Feb 10 02:20:05 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Data Channel Protocol: Data before DATA_CHANNEL_ACK
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Date: Mon, 10 Feb 2014 10:20:02 +0000
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Subject: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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Hi,

As is defined in the data channel protocol, an entity can send data once DA=
TA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is re=
ceived.

What is the reason for allowing data to be sent before DATA_CHANNEL_ACK? Th=
e sender may not even know (depends on whatever external negotiation mechan=
isms are used) whether the remote peer supports the protocol to begin with.

It think it would be good to allow the remote peer to accept (DATA_CHANNEL_=
ACK) or reject (stream reset) the DATA_CHANNEL_OPEN before data is sent.

Regards,

Christer



From christer.holmberg@ericsson.com  Mon Feb 10 02:21:58 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Hi,
=20
>> As is defined in the data channel protocol, an entity can send data once=
 DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is=
 received.
>> =20
>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at th=
e same time, using different stream id values, the result may be TWO data c=
hannels, with data sent on both.
>> =20
>> However, as is also indicated, the draft does not provide a mechanism to=
 prevent/handle such glare situation.
>> =20
>> I personally think there shall be a way to prevent such situation, or ot=
herwise I'm sure we are going to end up with interoperability problems.
>> =20
>> Couldn't we e.g. say that, by default, the DTLS client is responsible fo=
r sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it co=
uld even use both an even or odd stream id value.
> Section 4 reads:
>
>   To avoid glare in opening Channels, each side MUST use either even or
>   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>   used to determine which side uses odd or even is based on the
>   underlying DTLS connection role when used in WebRTC, with the side
>   acting as the DTLS client using even stream identifiers.
>
> So there is no problem in setting up data channels.
>
> This should not be confused by both sides setting up a data channel with =
the same Label. This ends up in two data channels with the same label.
> You can even create more data channels with the same label.

My point is that I don't want to end up with more than one channel.

Regards,

Christer


From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 02:36:36 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On Feb 10, 2014, at 11:21 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> As is defined in the data channel protocol, an entity can send data =
once DATA_CHANNEL_OPEN has been sent, before the associated =
DATA_CHANNEL_ACK is received.
>>>=20
>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN =
at the same time, using different stream id values, the result may be =
TWO data channels, with data sent on both.
>>>=20
>>> However, as is also indicated, the draft does not provide a =
mechanism to prevent/handle such glare situation.
>>>=20
>>> I personally think there shall be a way to prevent such situation, =
or otherwise I'm sure we are going to end up with interoperability =
problems.
>>>=20
>>> Couldn't we e.g. say that, by default, the DTLS client is =
responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for =
glare, and it could even use both an even or odd stream id value.
>> Section 4 reads:
>>=20
>>  To avoid glare in opening Channels, each side MUST use either even =
or
>>  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>>  used to determine which side uses odd or even is based on the
>>  underlying DTLS connection role when used in WebRTC, with the side
>>  acting as the DTLS client using even stream identifiers.
>>=20
>> So there is no problem in setting up data channels.
>>=20
>> This should not be confused by both sides setting up a data channel =
with the same Label. This ends up in two data channels with the same =
label.
>> You can even create more data channels with the same label.
>=20
> My point is that I don't want to end up with more than one channel.
And you won't, see above. However, there is no mechanism to ensure
uniqueness for labels. The WebRTC API already allows one side
to create n data channels with the same label.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


From salvatore.loreto@ericsson.com  Mon Feb 10 02:53:02 2014
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-02.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-02.txt
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I think is a good suggestion

/Salvatore

On Feb 10, 2014, at 12:11 PM, Michael Tuexen <Michael.Tuexen@lurchi.franken=
.de>
 wrote:

> On Feb 10, 2014, at 11:07 AM, Barry Dingle <btdingle@gmail.com> wrote:
>=20
>> Protocol looks good.=20
>>=20
>> One small point - the title of this protocol is "WebRTC Data Channel Pro=
tocol" but the Abstract says "This document specifies a simple protocol for=
 establishing symmetric data channels between the peers."  This could cause=
 confusion between what is a Data Channel Protocol and what is an Establish=
ing protocol.=20
>>=20
>> It appears that the title of this protocol should be "WebRTC Data Channe=
l Establishment Protocol" in order to avoid future confusion as to its purp=
ose.=20
> You are right, the protocol is just for establishing data channels.
> Other opinions regarding the name change?
>=20
> Best regards
> Michael
>>=20
>> /Barry Dingle
>> Australia
>>=20
>>=20
>> On Mon, Feb 10, 2014 at 8:16 AM, <internet-drafts@ietf.org> wrote:
>>=20
>> A New Internet-Draft is available from the on-line Internet-Drafts direc=
tories.
>> 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
>>        Authors         : Randell Jesup
>>                          Salvatore Loreto
>>                          Michael Tuexen
>>        Filename        : draft-ietf-rtcweb-data-protocol-02.txt
>>        Pages           : 12
>>        Date            : 2014-02-09
>>=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 a simple protocol for establishing
>>   symmetric data channels between the peers.  It uses a two way
>>   handshake and allows sending of user data without waiting for the
>>   handshake to complete.
>>=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-02
>>=20
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protocol-02
>>=20
>>=20
>> Please note that it may take a couple of minutes from the time of submis=
sion
>> until the htmlized version and diff are available at tools.ietf.org.
>>=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
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From stefan.lk.hakansson@ericsson.com  Mon Feb 10 02:56:42 2014
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>, Parthasarathi R <partha@parthasarathi.co.in>, 'Cb B' <cb.list6@gmail.com>, 'Simon Perreault' <simon.perreault@viagenie.ca>
Thread-Topic: [rtcweb] Query/Comment	on draft-ietf-rtcweb-use-cases-and-requirements-12
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On 2014-02-03 16:41, Tirumaleswar Reddy (tireddy) wrote:=0A=
>> -----Original Message----- From: rtcweb=0A=
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund=0A=
>> Sent: Friday, January 31, 2014 2:32 PM To: Parthasarathi R; 'Cb B';=0A=
>> 'Simon Perreault' Cc: rtcweb@ietf.org Subject: Re: [rtcweb]=0A=
>> Query/Comment on draft-ietf-rtcweb-use-cases-and- requirements-12=0A=
>>=0A=
>> Hi Partha,=0A=
>>=0A=
>> Personal opinion:=0A=
>>=0A=
>> I think the below places the text in the wrong context. The note is=0A=
>> in my mind relevant in the context of the general NAT/FW traversal=0A=
>> requirements, not the one discussing need to support multiple=0A=
>> NAT/FW traversal servers. Thus, I think Section 3.3.2 and thus=0A=
>> requirement F29. Or potentially regarding Requirement F2. Is more=0A=
>> appropriate places to include this.=0A=
>=0A=
> F29, F2 does not mention any NAT/FW traversal techniques. F29 just=0A=
> discusses the problems with NAT but IPv6 Firewalls could also be=0A=
> configured to block UDP traffic. F19 Requirement seems to be missing=0A=
> in the latest version.=0A=
=0A=
F19 was removed to the -12 version when the following text was removed =0A=
from the use-case "Video conferencing system with central server":=0A=
=0A=
=0A=
"The organization has an internal network set up with an aggressive =0A=
firewall handling access to the Internet.  If users cannot physically =0A=
access the internal network, they can establish a Virtual Private =0A=
Network (VPN)."=0A=
=0A=
Stefan=0A=
=0A=
>=0A=
> -Tiru.=0A=
>=0A=
>>=0A=
>> Cheers=0A=
>>=0A=
>> Magnus=0A=
>>=0A=
>> On 2014-01-31 02:26, Parthasarathi R wrote:=0A=
>>> Hi Magnus,=0A=
>>>=0A=
>>> I can live with Simon text in case it is documented in Sec 4.2=0A=
>>> as=0A=
>>>=0A=
>>> F31     The browser must be able to use several STUN and TURN=0A=
>>> servers. Note that TURN support being mandatory does not preclude=0A=
>>> the browser from supporting additional traversal mechanisms.=0A=
>>> ----------------------------------------------------------------=0A=
>>> F32     There browser must support that STUN and TURN servers to=0A=
>>> use are supplied by other entities than via the web application=0A=
>>> (i.e. the network provider). Note that TURN support being=0A=
>>> mandatory does not preclude the browser from supporting=0A=
>>> additional traversal mechanisms.=0A=
>>>=0A=
>>> and also Appendix A:=0A=
>>>=0A=
>>> A22     The Web API must provide means for the application to=0A=
>>> specify several STUN and/or TURN servers to use. Note that TURN=0A=
>>> support being mandatory does not preclude a Web API from=0A=
>>> supporting additional traversal mechanisms.=0A=
>>>=0A=
>>> Please let me know in case you have any issue in the above text.=0A=
>>>=0A=
>>> BTW, just for the record,=0A=
>>> draft-ietf-rtcweb-use-cases-and-requirements-12 does not specify=0A=
>>> the list of traversal mechanism requirements for WebRTC=0A=
>>> Gateway/Server.=0A=
>>>=0A=
>>> Thanks Partha=0A=
>>>=0A=
>>>> -----Original Message----- From: Magnus Westerlund=0A=
>>>> [mailto:magnus.westerlund@ericsson.com] Sent: Wednesday,=0A=
>>>> January 29, 2014 2:59 PM To: Parthasarathi R; 'Cb B'; 'Simon=0A=
>>>> Perreault' Cc: rtcweb@ietf.org Subject: Re: [rtcweb]=0A=
>>>> Query/Comment on draft-ietf-rtcweb-use-cases-and-=0A=
>>>> requirements-12=0A=
>>>>=0A=
>>>> Hi Partha and WG,=0A=
>>>>=0A=
>>>> I don't see any support for the changes you proposes in this=0A=
>>>> discussion. What I see some support for is to add a statement=0A=
>>>> making clear that there might be additional NAT/Firewall=0A=
>>>> traversal mechanisms than STUN/TURN. Simon proposed:=0A=
>>>>=0A=
>>>> "Note that TURN support being mandatory does not preclude a=0A=
>>>> WebRTC endpoint from supporting additional traversal=0A=
>>>> mechanisms."=0A=
>>>>=0A=
>>>> However, looking at the document as it is currently written, I=0A=
>>>> am uncertain where this would be added. The first mention of=0A=
>>>> TURN is in Section 3.3.4.1, and that section is focused on the=0A=
>>>> global service provider perspective and the need for location=0A=
>>>> based provisioning of NAT/Firewall traversal server resources.=0A=
>>>>=0A=
>>>> I think it can be added to Section 3.3.5.1 without being=0A=
>>>> misplaced, but it would be given a slightly narrower scope.=0A=
>>>>=0A=
>>>> I any of you want to be more explicit where this should be=0A=
>>>> included, please be. If you are not forthcoming I will request=0A=
>>>> the authors to add this in what they consider sensible=0A=
>>>> position.=0A=
>>>>=0A=
>>>> Cheers=0A=
>>>>=0A=
>>>> Magnus=0A=
>>>>=0A=
>>>>=0A=
>>>> On 2014-01-25 17:46, Parthasarathi R wrote:=0A=
>>>>> Hi Simon,=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> Thanks for your understanding about my firewall/NAT related=0A=
>>>>> problem statement in this draft.=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> I have proposed the firewall/NAT related text by which the=0A=
>>>>> specific mechanism is not highlighted in the requirement=0A=
>>>>> document as there is=0A=
>>>> no=0A=
>>>>> WG consensus for any of the mechanism including TURN. It is=0A=
>>>>> possible=0A=
>>>> to=0A=
>>>>> argue hypothetically in PNTAW alias that PCP is the only=0A=
>>>>> mechanism required in WebRTC endpoint.   Also, I'm more=0A=
>>>>> interested in WebRTC gateway/server (Sec 4.3. of=0A=
>>>>> draft-ietf-rtcweb-use-cases-and-requirements-12)=0A=
>>>>> requirements wherein=0A=
>>>> it=0A=
>>>>> is not required to support TURN and the related mail thread=0A=
>>>>> is=0A=
>>>>> http://www.ietf.org/mail-archive/web/pntaw/current/msg00181.html.=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
IMO, my proposed text without mentioning any firewall/NAT mechanism=0A=
>>>> in=0A=
>>>>> the requirement document helps to move forward without depend=0A=
>>>>> on=0A=
>> the=0A=
>>>>> solution discussion in PNTAW alias.=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> Thanks=0A=
>>>>>=0A=
>>>>> Partha=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> *From:*Cb B [mailto:cb.list6@gmail.com] *Sent:* Saturday,=0A=
>>>>> January 25, 2014 6:22 AM *To:* Simon Perreault *Cc:*=0A=
>>>>> rtcweb@ietf.org; Parthasarathi R *Subject:* Re: [rtcweb]=0A=
>>>>> Query/Comment on=0A=
>>>>> draft-ietf-rtcweb-use-cases-and-requirements-12=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> On Jan 24, 2014 10:17 AM, "Simon Perreault"=0A=
>>>> <simon.perreault@viagenie.ca=0A=
>>>>> <mailto:simon.perreault@viagenie.ca>> wrote:=0A=
>>>>>>=0A=
>>>>>> Le 2014-01-24 12:14, Parthasarathi R a =E9crit :=0A=
>>>>>>=0A=
>>>>>>> Please note that when non-IETFers read this requirement=0A=
>>>>>>> document,=0A=
>>>>> they come=0A=
>>>>>>> to the conclusion that IETF RTCWeb WG recommends TURN and=0A=
>>>>>>> not other mechanisms. I'm saying that requirement=0A=
>>>>>>> document should not be used=0A=
>>>>> as the=0A=
>>>>>>> mechanism to eliminate the other alternatives when there=0A=
>>>>>>> is a=0A=
>>>> discussion=0A=
>>>>>>> going-on in PNTAW alias. So, I'm asking for the change.=0A=
>>>>>>=0A=
>>>>>>=0A=
>>>>>> I would totally agree with that sentiment, although I don't=0A=
>>>>>> see your=0A=
>>>>> proposed text change reflecting it accurately. How about=0A=
>>>>> simply:=0A=
>>>>>>=0A=
>>>>>> "Note that TURN support being mandatory does not preclude=0A=
>>>>>> a=0A=
>> WebRTC=0A=
>>>>> endpoint from supporting additional traversal mechanisms."=0A=
>>>>>>=0A=
>>>>>>=0A=
>>>>>=0A=
>>>>> +1 for the above text.=0A=
>>>>>=0A=
>>>>> CB=0A=
>>>>>=0A=
>>>>>> Simon -- DTN made easy, lean, and smart -->=0A=
>>>>>> http://postellation.viagenie.ca NAT64/DNS64 open-source=0A=
>>>>>> --> http://ecdysis.viagenie.ca STUN/TURN server=0A=
>>>>>> --> http://numb.viagenie.ca=0A=
>>>>>> _______________________________________________ rtcweb=0A=
>>>>>> mailing list rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>>>>>=0A=
>>>>>=0A=
>>>>>=0A=
>>>>> _______________________________________________ rtcweb=0A=
>>>>> mailing list rtcweb@ietf.org=0A=
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>>>>>=0A=
>>>>=0A=
>>>>=0A=
>>>> --=0A=
>>>>=0A=
>>>> Magnus Westerlund=0A=
>>>>=0A=
>>>> ---------------------------------------------------------------------=
=0A=
>>>>=0A=
>>>>=0A=
- Services, Media and Network features, Ericsson Research EAB/TXM=0A=
>>>> ----------------------------------------------------------------------=
=0A=
>>>>=0A=
>>>>=0A=
Ericsson AB                 | Phone  +46 10 7148287=0A=
>>>> F=E4r=F6gatan 6                 | Mobile +46 73 0949079 SE-164 80=0A=
>>>> Stockholm, Sweden | mailto:=0A=
>> magnus.westerlund@ericsson.com=0A=
>>>> ---------------------------------------------------------------------=
=0A=
>>>>=0A=
>>>>=0A=
-=0A=
>>>=0A=
>>>=0A=
>>>=0A=
>>=0A=
>>=0A=
>> --=0A=
>>=0A=
>> Magnus Westerlund=0A=
>>=0A=
>> ----------------------------------------------------------------------=
=0A=
>>=0A=
>>=0A=
Services, Media and Network features, Ericsson Research EAB/TXM=0A=
>> ----------------------------------------------------------------------=
=0A=
>>=0A=
>>=0A=
Ericsson AB                 | Phone  +46 10 7148287=0A=
>> F=E4r=F6gatan 6                 | Mobile +46 73 0949079 SE-164 80=0A=
>> Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com=0A=
>> ----------------------------------------------------------------------=
=0A=
>>=0A=
>>=0A=
>>=0A=
_______________________________________________=0A=
>> rtcweb mailing list rtcweb@ietf.org=0A=
>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
> _______________________________________________ rtcweb mailing list=0A=
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

From magnus.westerlund@ericsson.com  Mon Feb 10 03:02:06 2014
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Partha,

See inline

On 2014-02-08 10:12, Parthasarathi R wrote:
> Hi Stefan,
> 
> <snip>
>> Yes, I did that change (TURN -> ICE). My understanding was that ICE is
>> what is used/mandated (and it in turn makes use of STUN and TURN).
>>
> </snip>
> 
> ICE is mandated in RTCWeb but I disagree to your assumption that ICE in turn
> makes use of TURN. My concern is that The requirement document is misleading
> about TURN requirements. ICE-TCP avoids TURN server as the middle-box in the
> network. ICE-TCP is host candidate whereas TURN is relay candidate and as
> per ICE candidate selection, host candidate is preferred over relay
> candidates. 

I will agree that the note as currently worded is not completely
accurate and clear. Based on the text in the preceding section, it makes
sense to note that ICE using STUN and TURN is not the only solution
including additional ICE modes or relay services. I request the authors
propose a new formulation of the note.

> 
> In case of WebRTC servers (like conference)/gateway implementation, WebRTC
> server is never going to be behind firewall, ICE-TCP serve the purpose and
> there is no need of TURN server as a middle-box in the network. So, Could
> you please correct the F31, F32 requirement text in the next revision of the
> draft.
>   

This is a request to change two requirements. I have not seen a clear
consensus on this change. Thus, the change is rejected for now.

>From my perspective your proposal for ICE-TCP based solution is a choice
to fulfill F2 and F29, not F31 and F32. Thus, I don't see an issue for
F31 and F32 being STUN and TURN specific.

Is really the current formulations in the requirement documents a hinder
against the WG considering an ICE-TCP based traversal addition? I don't
see it, and if you do, can you please be clearer in your explanation
what the issue is? If not, I think your time is better spent working on
building a clear motivation why ICE-TCP is needed to help establish a WG
consensus on this.

Cheers

Magnus Westerlund
(As co-chair)

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] PPID, UTF-16 and DOMString (Re: RTCWEB Data Channel: Usage of PPID for protocol multiplexing)
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=20
> > That's good. Glad to see that no one liked the complex alternate option=
...
> yet! :-)
> > I just put it there for completeness and some discussion.
> > With "utf-8 on wire", which I too think is the right approach, there is
> some spec work todo:
> > 1. Define IANA utf-8 ppid. Creating a new one is better instead of
> replacing existing DOMString as DOMString may be used by browsers for
> sometime to come and also to have backward compatibility/interworking.
>=20
> Have you ever seen UTF-16 on the wire in the DataChannel protocol?

[Raju] I have used websockets and no, I did not see UTF-16 on wire. Further=
, after going thru websocket API spec http://www.w3.org/TR/2011/WD-websocke=
ts-20110419/ , which does have same send(DOMString) API call, I realized th=
at browsers are very well doing this conversion already in other areas. Lat=
er, Randell Jesup also comfirmed this for Firefox. So, no concerns on this.=
 Thanks!

-Raju

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Hi,

>>>> As is defined in the data channel protocol, an entity can send data on=
ce DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK =
is received.
>>>>=20
>>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at =
the same time, using different stream id values, the result may be TWO data=
 channels, with data sent on both.
>>>>=20
>>>> However, as is also indicated, the draft does not provide a mechanism =
to prevent/handle such glare situation.
>>>>=20
>>>> I personally think there shall be a way to prevent such situation, or =
otherwise I'm sure we are going to end up with interoperability problems.
>>>>=20
>>>> Couldn't we e.g. say that, by default, the DTLS client is responsible =
for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it =
could even use both an even or odd stream id value.
>>> Section 4 reads:
>>>=20
>>>  To avoid glare in opening Channels, each side MUST use either even=20
>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method =
=20
>>> used to determine which side uses odd or even is based on the =20
>>> underlying DTLS connection role when used in WebRTC, with the side =20
>>> acting as the DTLS client using even stream identifiers.
>>>=20
>>> So there is no problem in setting up data channels.
>>>=20
>>> This should not be confused by both sides setting up a data channel wit=
h the same Label. This ends up in two data channels with the same label.
>>> You can even create more data channels with the same label.
>>=20
>> My point is that I don't want to end up with more than one channel.
> And you won't, see above.=20

I am not sure I understand. What text do you refer to?

> However, there is no mechanism to ensure uniqueness for labels. The WebRT=
C API already allows one side to create n data channels with the same label=
.

I am not saying it should be forbidden to do so. But, for protocol X, I wan=
t to end up with ONE data channel between my peers.

If I end up with two data channels for protocol X, can I send data on which=
ever I want to?=20

Regards,

Christer


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Subject: Re: [rtcweb] Query/Comment on draft-ietf-rtcweb-use-cases-and-requirements-12
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On 2014-02-03 16:41, Tirumaleswar Reddy (tireddy) wrote:
>> -----Original Message----- From: rtcweb
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund 
>> Sent: Friday, January 31, 2014 2:32 PM To: Parthasarathi R; 'Cb B';
>> 'Simon Perreault' Cc: rtcweb@ietf.org Subject: Re: [rtcweb]
>> Query/Comment on draft-ietf-rtcweb-use-cases-and- requirements-12
>> 
>> Hi Partha,
>> 
>> Personal opinion:
>> 
>> I think the below places the text in the wrong context. The note is
>> in my mind relevant in the context of the general NAT/FW traversal
>> requirements, not the one discussing need to support multiple
>> NAT/FW traversal servers. Thus, I think Section 3.3.2 and thus
>> requirement F29. Or potentially regarding Requirement F2. Is more
>> appropriate places to include this.
> 
> F29, F2 does not mention any NAT/FW traversal techniques. F29 just
> discusses the problems with NAT but IPv6 Firewalls could also be
> configured to block UDP traffic. F19 Requirement seems to be missing
> in the latest version.
> 

Regarding F29 and F2, that is the benefit of having solution neutral
requirements. Thus, these requirements are not blocking or hindering you
from proposing additional solutions or implementation requirements when
it comes to NAT/FW traversal. Thus, these requirements applies to UDP
blocking IPv6 firewalls as well as other types.

cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: How to determine TLS roles?
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Subject: [rtcweb] How to determine TLS roles?
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Hi,

If I understand correctly, in RTCWEB we are going to use a single DTLS asso=
ciation for two things:


1)      Key exchange for SRTP

2)      The data channel

Now, this means that there needs to be a generic way to determine the TLS r=
oles.

In WEBRTC, how are the TLS roles determined?

Regards,

Christer



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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">If I understand correctly, in RTCWEB we are going to=
 use a single DTLS association for two things:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">1)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Key exchange for SRTP<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">2)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>The data channel<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Now, this means that there needs to be a generic way=
 to determine the TLS roles.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In WEBRTC, how are the TLS roles determined?<o:p></o=
:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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--_000_7594FB04B1934943A5C02806D1A2204B1D1672FCESESSMB209erics_--

From tim@phonefromhere.com  Mon Feb 10 05:45:46 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 10 Feb 2014, at 13:29, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> If I understand correctly, in RTCWEB we are going to use a single DTLS =
association for two things:
>=20
> 1)      Key exchange for SRTP
> 2)      The data channel
>=20
> Now, this means that there needs to be a generic way to determine the =
TLS roles.
>=20
> In WEBRTC, how are the TLS roles determined?


That was the subject of a rant of mine.

I think the correct answer is that the party that ends up being =
IceControlling then becomes the
DTLS client. Typically the initiator of a session is IceControlling.=20

However the recent addition of=20
a=3Dsetup:passive=20
to chrome's SDP clouds the story somewhat, allowing an IceController to =
say that it is DTLS passive
and allow the non-initiator to be the DTLS client.

Ostensibly this is to support early media - where the receiver of a call =
wishes to send media before the
initiator does.

As I said (rather intemperately) this added complexity is not worth the =
benefit.

Tim.

http://webrtchacks.com/tim-rant/

>=20
> Regards,
>=20
> Christer
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




From christer.holmberg@ericsson.com  Mon Feb 10 05:51:49 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Hi,

>> If I understand correctly, in RTCWEB we are going to use a single DTLS a=
ssociation for two things:
>>=20
>> 1)      Key exchange for SRTP
>> 2)      The data channel
>>=20
>> Now, this means that there needs to be a generic way to determine the TL=
S roles.
>>=20
>> In WEBRTC, how are the TLS roles determined?
>
> That was the subject of a rant of mine.
>
> I think the correct answer is that the party that ends up being IceContro=
lling then becomes the DTLS client. Typically the initiator of a session is=
 IceControlling.=20
>
> However the recent addition of
> a=3Dsetup:passive
> to chrome's SDP clouds the story somewhat, allowing an IceController to s=
ay that it is DTLS passive and allow the non-initiator to be the DTLS clien=
t.
>
> Ostensibly this is to support early media - where the receiver of a call =
wishes to send media before the initiator does.
>
> As I said (rather intemperately) this added complexity is not worth the b=
enefit.

I think that it should be possible for a JS App to explicitly set the roles=
.

Because, when SDP O/A is used on the wire, there are specified rules on how=
 the roles are determined. But, some other on-the-wire protocol may have di=
fferent rules.

Regards,

Christer


From christer.holmberg@ericsson.com  Mon Feb 10 06:23:57 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Hi,

>>>> If I understand correctly, in RTCWEB we are going to use a single DTLS=
 association for two things:
>>>>=20
>>>> 1)      Key exchange for SRTP
>>>> 2)      The data channel
>>>>=20
>>>> Now, this means that there needs to be a generic way to determine the =
TLS roles.
>>>>=20
>>>> In WEBRTC, how are the TLS roles determined?
>>>=20
>>> That was the subject of a rant of mine.
>>>=20
>>> I think the correct answer is that the party that ends up being IceCont=
rolling then becomes the DTLS client. Typically the initiator of a session =
is IceControlling.=20
>>>=20
>>> However the recent addition of
>>>> a=3Dsetup:passive
>>> to chrome's SDP clouds the story somewhat, allowing an IceController to=
 say that it is DTLS passive and allow the non-initiator to be the DTLS cli=
ent.
>>>=20
>>> Ostensibly this is to support early media - where the receiver of a cal=
l wishes to send media before the initiator does.
>>>=20
>>> As I said (rather intemperately) this added complexity is not worth the=
 benefit.
>>=20
>> I think that it should be possible for a JS App to explicitly set the ro=
les.
>>=20
>> Because, when SDP O/A is used on the wire, there are specified rules on =
how the roles are determined. But, some other on-the-wire protocol may have=
 different rules.
>>=20
> There are many things that the JS app will need to do if we aren't bound =
by SDP O/A - but I agreed not to talk about that - until after 1.0 gets out=
 :-)

Fair enough, but for 1.0 we still need to know how the roles are determined=
 :)

The security-arch document does say that a DTLS handshake is performed on e=
ach channel that has been established by ICE, but I don't find anything reg=
arding the roles.

Regards,

Christer

From tim@phonefromhere.com  Mon Feb 10 06:24:43 2014
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 10 Feb 2014, at 13:51, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> If I understand correctly, in RTCWEB we are going to use a single =
DTLS association for two things:
>>>=20
>>> 1)      Key exchange for SRTP
>>> 2)      The data channel
>>>=20
>>> Now, this means that there needs to be a generic way to determine =
the TLS roles.
>>>=20
>>> In WEBRTC, how are the TLS roles determined?
>>=20
>> That was the subject of a rant of mine.
>>=20
>> I think the correct answer is that the party that ends up being =
IceControlling then becomes the DTLS client. Typically the initiator of =
a session is IceControlling.=20
>>=20
>> However the recent addition of
>> a=3Dsetup:passive
>> to chrome's SDP clouds the story somewhat, allowing an IceController =
to say that it is DTLS passive and allow the non-initiator to be the =
DTLS client.
>>=20
>> Ostensibly this is to support early media - where the receiver of a =
call wishes to send media before the initiator does.
>>=20
>> As I said (rather intemperately) this added complexity is not worth =
the benefit.
>=20
> I think that it should be possible for a JS App to explicitly set the =
roles.
>=20
> Because, when SDP O/A is used on the wire, there are specified rules =
on how the roles are determined. But, some other on-the-wire protocol =
may have different rules.
>=20
> Regards,
>=20
> Christer
>=20

There are many things that the JS app will need to do if we aren't bound =
by SDP O/A - but I agreed not to talk about that -
until after 1.0 gets out :-)

T.



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--001a1136b37443522804f20e7035
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This is defined in RFC 5763 S 5:
http://tools.ietf.org/html/rfc5763#section-5

Which points to:
http://tools.ietf.org/html/rfc4145

-Ekr





On Mon, Feb 10, 2014 at 6:23 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>>> If I understand correctly, in RTCWEB we are going to use a single
> DTLS association for two things:
> >>>>
> >>>> 1)      Key exchange for SRTP
> >>>> 2)      The data channel
> >>>>
> >>>> Now, this means that there needs to be a generic way to determine the
> TLS roles.
> >>>>
> >>>> In WEBRTC, how are the TLS roles determined?
> >>>
> >>> That was the subject of a rant of mine.
> >>>
> >>> I think the correct answer is that the party that ends up being
> IceControlling then becomes the DTLS client. Typically the initiator of a
> session is IceControlling.
> >>>
> >>> However the recent addition of
> >>>> a=setup:passive
> >>> to chrome's SDP clouds the story somewhat, allowing an IceController
> to say that it is DTLS passive and allow the non-initiator to be the DTLS
> client.
> >>>
> >>> Ostensibly this is to support early media - where the receiver of a
> call wishes to send media before the initiator does.
> >>>
> >>> As I said (rather intemperately) this added complexity is not worth
> the benefit.
> >>
> >> I think that it should be possible for a JS App to explicitly set the
> roles.
> >>
> >> Because, when SDP O/A is used on the wire, there are specified rules on
> how the roles are determined. But, some other on-the-wire protocol may have
> different rules.
> >>
> > There are many things that the JS app will need to do if we aren't bound
> by SDP O/A - but I agreed not to talk about that - until after 1.0 gets out
> :-)
>
> Fair enough, but for 1.0 we still need to know how the roles are
> determined :)
>
> The security-arch document does say that a DTLS handshake is performed on
> each channel that has been established by ICE, but I don't find anything
> regarding the roles.
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a1136b37443522804f20e7035
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">This is defined in RFC 5763 S 5:<div><a href=3D"http://too=
ls.ietf.org/html/rfc5763#section-5">http://tools.ietf.org/html/rfc5763#sect=
ion-5</a></div><div><br></div><div>Which points to:</div><div><a href=3D"ht=
tp://tools.ietf.org/html/rfc4145">http://tools.ietf.org/html/rfc4145</a></d=
iv>

<div><br></div><div>-Ekr</div><div><br><div><br></div><div><br></div></div>=
</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Feb 10, 2014 at 6:23 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@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"">Hi,<br>
<br>
&gt;&gt;&gt;&gt; If I understand correctly, in RTCWEB we are going to use a=
 single DTLS association for two things:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; 1) =A0 =A0 =A0Key exchange for SRTP<br>
&gt;&gt;&gt;&gt; 2) =A0 =A0 =A0The data channel<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Now, this means that there needs to be a generic way to de=
termine the TLS roles.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; In WEBRTC, how are the TLS roles determined?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; That was the subject of a rant of mine.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I think the correct answer is that the party that ends up bein=
g IceControlling then becomes the DTLS client. Typically the initiator of a=
 session is IceControlling.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; However the recent addition of<br>
&gt;&gt;&gt;&gt; a=3Dsetup:passive<br>
&gt;&gt;&gt; to chrome&#39;s SDP clouds the story somewhat, allowing an Ice=
Controller to say that it is DTLS passive and allow the non-initiator to be=
 the DTLS client.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Ostensibly this is to support early media - where the receiver=
 of a call wishes to send media before the initiator does.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; As I said (rather intemperately) this added complexity is not =
worth the benefit.<br>
&gt;&gt;<br>
&gt;&gt; I think that it should be possible for a JS App to explicitly set =
the roles.<br>
&gt;&gt;<br>
&gt;&gt; Because, when SDP O/A is used on the wire, there are specified rul=
es on how the roles are determined. But, some other on-the-wire protocol ma=
y have different rules.<br>
&gt;&gt;<br>
</div><div class=3D"">&gt; There are many things that the JS app will need =
to do if we aren&#39;t bound by SDP O/A - but I agreed not to talk about th=
at - until after 1.0 gets out :-)<br>
<br>
</div>Fair enough, but for 1.0 we still need to know how the roles are dete=
rmined :)<br>
<br>
The security-arch document does say that a DTLS handshake is performed on e=
ach channel that has been established by ICE, but I don&#39;t find anything=
 regarding the roles.<br>
<br>
Regards,<br>
<br>
Christer<br>
<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></div>

--001a1136b37443522804f20e7035--

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To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Hi,

>This is defined in RFC 5763 S 5:
>http://tools.ietf.org/html/rfc5763#section-5
>
>Which points to:
>http://tools.ietf.org/html/rfc4145

Ok. So, a few questions to for clarification:

Q1: This means that the JS App must set the setup attrbute value before pas=
sing an SDP to the browser?

Q2: If SDP O/A is not used on the wire, there needs to be another mechanism=
 for the peers to negotiate/indicate who is "active" and who is "passive"?

Q3: If you have mulitple m- lines, all using the same DTLS association, the=
 setup attribute value must be identical in all m- lines?

Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using the se=
tup attribute value) as DTLS/SRTP for determining the TLS role?

Regards,

Christer







On Mon, Feb 10, 2014 at 6:23 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

>>>> If I understand correctly, in RTCWEB we are going to use a single DTLS=
 association for two things:
>>>>
>>>> 1)      Key exchange for SRTP
>>>> 2)      The data channel
>>>>
>>>> Now, this means that there needs to be a generic way to determine the =
TLS roles.
>>>>
>>>> In WEBRTC, how are the TLS roles determined?
>>>
>>> That was the subject of a rant of mine.
>>>
>>> I think the correct answer is that the party that ends up being IceCont=
rolling then becomes the DTLS client. Typically the initiator of a session =
is IceControlling.
>>>
>>> However the recent addition of
>>>> a=3Dsetup:passive
>>> to chrome's SDP clouds the story somewhat, allowing an IceController to=
 say that it is DTLS passive and allow the non-initiator to be the DTLS cli=
ent.
>>>
>>> Ostensibly this is to support early media - where the receiver of a cal=
l wishes to send media before the initiator does.
>>>
>>> As I said (rather intemperately) this added complexity is not worth the=
 benefit.
>>
>> I think that it should be possible for a JS App to explicitly set the ro=
les.
>>
>> Because, when SDP O/A is used on the wire, there are specified rules on =
how the roles are determined. But, some other on-the-wire protocol may have=
 different rules.
>>
> There are many things that the JS app will need to do if we aren't bound =
by SDP O/A - but I agreed not to talk about that - until after 1.0 gets out=
 :-)

Fair enough, but for 1.0 we still need to know how the roles are determined=
 :)

The security-arch document does say that a DTLS handshake is performed on e=
ach channel that has been established by ICE, but I don't find anything reg=
arding the roles.

Regards,

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


From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 08:56:54 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On Feb 10, 2014, at 12:31 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>>> As is defined in the data channel protocol, an entity can send =
data once DATA_CHANNEL_OPEN has been sent, before the associated =
DATA_CHANNEL_ACK is received.
>>>>>=20
>>>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN =
at the same time, using different stream id values, the result may be =
TWO data channels, with data sent on both.
>>>>>=20
>>>>> However, as is also indicated, the draft does not provide a =
mechanism to prevent/handle such glare situation.
>>>>>=20
>>>>> I personally think there shall be a way to prevent such situation, =
or otherwise I'm sure we are going to end up with interoperability =
problems.
>>>>>=20
>>>>> Couldn't we e.g. say that, by default, the DTLS client is =
responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for =
glare, and it could even use both an even or odd stream id value.
>>>> Section 4 reads:
>>>>=20
>>>> To avoid glare in opening Channels, each side MUST use either even=20=

>>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The =
method =20
>>>> used to determine which side uses odd or even is based on the =20
>>>> underlying DTLS connection role when used in WebRTC, with the side =20=

>>>> acting as the DTLS client using even stream identifiers.
>>>>=20
>>>> So there is no problem in setting up data channels.
>>>>=20
>>>> This should not be confused by both sides setting up a data channel =
with the same Label. This ends up in two data channels with the same =
label.
>>>> You can even create more data channels with the same label.
>>>=20
>>> My point is that I don't want to end up with more than one channel.
>> And you won't, see above.=20
>=20
> I am not sure I understand. What text do you refer to?
To avoid glare in opening Channels, each side MUST use either even=20
or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method =20=

used to determine which side uses odd or even is based on the =20
underlying DTLS connection role when used in WebRTC, with the side =20
acting as the DTLS client using even stream identifiers.
>=20
>> However, there is no mechanism to ensure uniqueness for labels. The =
WebRTC API already allows one side to create n data channels with the =
same label.
>=20
> I am not saying it should be forbidden to do so. But, for protocol X, =
I want to end up with ONE data channel between my peers.
Then only one side should create a data channel.
>=20
> If I end up with two data channels for protocol X, can I send data on =
whichever I want to?=20
Sure. These would be two independent bidirectional data channels.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


From christer.holmberg@ericsson.com  Mon Feb 10 09:30:40 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Hi,

>>>>>> As is defined in the data channel protocol, an entity can send data =
once DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_AC=
K is received.
>>>>>>
>>>>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN a=
t the same time, using different stream id values, the result may be TWO da=
ta channels, with data sent on both.
>>>>>>
>>>>>> However, as is also indicated, the draft does not provide a mechanis=
m to prevent/handle such glare situation.
>>>>>>
>>>>>> I personally think there shall be a way to prevent such situation, o=
r otherwise I'm sure we are going to end up with interoperability problems.
>>>>>>
>>>>>> Couldn't we e.g. say that, by default, the DTLS client is responsibl=
e for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and i=
t could even use both an even or odd stream id value.
>>>>> Section 4 reads:
>>>>>
>>>>> To avoid glare in opening Channels, each side MUST use either even
>>>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>>>>> used to determine which side uses odd or even is based on the
>>>>> underlying DTLS connection role when used in WebRTC, with the side
>>>>> acting as the DTLS client using even stream identifiers.
>>>>>
>>>>> So there is no problem in setting up data channels.
>>>>>
>>>>> This should not be confused by both sides setting up a data channel w=
ith the same Label. This ends up in two data channels with the same label.
>>>>> You can even create more data channels with the same label.
>>>>
>>>> My point is that I don't want to end up with more than one channel.
>>> And you won't, see above.
>>
>> I am not sure I understand. What text do you refer to?
> To avoid glare in opening Channels, each side MUST use either even
> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
> used to determine which side uses odd or even is based on the
> underlying DTLS connection role when used in WebRTC, with the side
> acting as the DTLS client using even stream identifiers.

That part I understand.

But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the sam=
e protocol, and THAT is what I would like to avoid. I don't want to have tw=
o bi-directional data channels for the same thing.

For that reason I suggested that we could also specify which endpoint is re=
sponsisble for sending DATA_CHANNEL_OPEN.


>>> However, there is no mechanism to ensure uniqueness for labels. The Web=
RTC API already allows one side to create n data channels with the same lab=
el.
>>
>> I am not saying it should be forbidden to do so. But, for protocol X, I =
want to end up with ONE data channel between my peers.
>>Then only one side should create a data channel.
>>
>> If I end up with two data channels for protocol X, can I send data on wh=
ichever I want to?
> Sure. These would be two independent bidirectional data channels.

Possible for the same protocol - and that is what I would want to avoid. I =
want to send my protocol X messages on one data channel, and I would like t=
o receive the protocol X messages from my peer on the same data channel.

Now, one can of course solve this by reseting one of the data channels, but=
 then the question is which endpoint should reset :)

Regards,

Christer


From tim@phonefromhere.com  Mon Feb 10 09:40:08 2014
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 10 Feb 2014, at 16:37, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
> Hi,
>=20
>> This is defined in RFC 5763 S 5:
>> http://tools.ietf.org/html/rfc5763#section-5
>>=20
>> Which points to:
>> http://tools.ietf.org/html/rfc4145
>=20
> Ok. So, a few questions to for clarification:
>=20
> Q1: This means that the JS App must set the setup attrbute value =
before passing an SDP to the browser?
No, peer-to-peer browser calls should 'do the right thing' - based on =
the initator/ice-controlling rule=20
Both browsers set=20
a=3Dsetup:act-pass
meaning that we fall back to the old rules.

>=20
> Q2: If SDP O/A is not used on the wire, there needs to be another =
mechanism for the peers to negotiate/indicate who is "active" and who is =
"passive"?
>=20
If you don't care, and are locally generating the SDP for a remote offer =
or answer, set=20
a=3Dsetup:act-pass=20
and the roles sort themselves out.

If however you have a gateway initating a call to a browser, and it =
wants to do early media (it's an 800 number gateway for example)
then it sets =20
a=3Dsetup:active

in the SDP it sends (or you create on it's behalf).



> Q3: If you have mulitple m- lines, all using the same DTLS =
association, the setup attribute value must be identical in all m- =
lines?
>=20

Yes, I think so.

> Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using =
the setup attribute value) as DTLS/SRTP for determining the TLS role?

That's the way chrome seems to implement it - and presumably the way the =
spec is intended.

T.


From pthatcher@google.com  Mon Feb 10 10:21:38 2014
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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 10 Feb 2014 10:20:52 -0800
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Subject: Re: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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--047d7b2e138dc6c64504f21168d2
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If you have to wait for the ACK, it's one more RTT you have to wait before
you can send data.  We've already got many RTTs form ICE and DTLS and
what's built into SCTP.  We originally didn't have an ACK message at all,
in large part because we didn't want more RTTs.  The ACK was put in to
handle the edge case of unordered data arriving before the OPEN message.
 But we added it in a way that it doesn't add more RTTs.


On Mon, Feb 10, 2014 at 2:20 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> As is defined in the data channel protocol, an entity can send data once
> DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is
> received.
>
> What is the reason for allowing data to be sent before DATA_CHANNEL_ACK?
> The sender may not even know (depends on whatever external negotiation
> mechanisms are used) whether the remote peer supports the protocol to begin
> with.
>
> It think it would be good to allow the remote peer to accept
> (DATA_CHANNEL_ACK) or reject (stream reset) the DATA_CHANNEL_OPEN before
> data is sent.
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b2e138dc6c64504f21168d2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:verdana,=
sans-serif">If you have to wait for the ACK, it&#39;s one more RTT you have=
 to wait before you can send data. =C2=A0We&#39;ve already got many RTTs fo=
rm ICE and DTLS and what&#39;s built into SCTP. =C2=A0We originally didn&#3=
9;t have an ACK message at all, in large part because we didn&#39;t want mo=
re RTTs. =C2=A0The ACK was put in to handle the edge case of unordered data=
 arriving before the OPEN message. =C2=A0But we added it in a way that it d=
oesn&#39;t add more RTTs.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Feb 10, 2014 at 2:20 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@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">Hi,<br>
<br>
As is defined in the data channel protocol, an entity can send data once DA=
TA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is re=
ceived.<br>
<br>
What is the reason for allowing data to be sent before DATA_CHANNEL_ACK? Th=
e sender may not even know (depends on whatever external negotiation mechan=
isms are used) whether the remote peer supports the protocol to begin with.=
<br>


<br>
It think it would be good to allow the remote peer to accept (DATA_CHANNEL_=
ACK) or reject (stream reset) the DATA_CHANNEL_OPEN before data is sent.<br=
>
<br>
Regards,<br>
<br>
Christer<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><br></div>

--047d7b2e138dc6c64504f21168d2--

From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 11:08:36 2014
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On Feb 10, 2014, at 6:29 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
> Hi,
>=20
>>>>>>> As is defined in the data channel protocol, an entity can send =
data once DATA_CHANNEL_OPEN has been sent, before the associated =
DATA_CHANNEL_ACK is received.
>>>>>>>=20
>>>>>>> The draft also says that, if both endpoints send =
DATA_CHANNEL_OPEN at the same time, using different stream id values, =
the result may be TWO data channels, with data sent on both.
>>>>>>>=20
>>>>>>> However, as is also indicated, the draft does not provide a =
mechanism to prevent/handle such glare situation.
>>>>>>>=20
>>>>>>> I personally think there shall be a way to prevent such =
situation, or otherwise I'm sure we are going to end up with =
interoperability problems.
>>>>>>>=20
>>>>>>> Couldn't we e.g. say that, by default, the DTLS client is =
responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for =
glare, and it could even use both an even or odd stream id value.
>>>>>> Section 4 reads:
>>>>>>=20
>>>>>> To avoid glare in opening Channels, each side MUST use either =
even
>>>>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The =
method
>>>>>> used to determine which side uses odd or even is based on the
>>>>>> underlying DTLS connection role when used in WebRTC, with the =
side
>>>>>> acting as the DTLS client using even stream identifiers.
>>>>>>=20
>>>>>> So there is no problem in setting up data channels.
>>>>>>=20
>>>>>> This should not be confused by both sides setting up a data =
channel with the same Label. This ends up in two data channels with the =
same label.
>>>>>> You can even create more data channels with the same label.
>>>>>=20
>>>>> My point is that I don't want to end up with more than one =
channel.
>>>> And you won't, see above.
>>>=20
>>> I am not sure I understand. What text do you refer to?
>> To avoid glare in opening Channels, each side MUST use either even
>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>> used to determine which side uses odd or even is based on the
>> underlying DTLS connection role when used in WebRTC, with the side
>> acting as the DTLS client using even stream identifiers.
>=20
> That part I understand.
>=20
> But, this means that both endpoints may send DATA_CHANNEL_OPEN, for =
the same protocol, and THAT is what I would like to avoid. I don't want =
to have two bi-directional data channels for the same thing.
>=20
> For that reason I suggested that we could also specify which endpoint =
is responsisble for sending DATA_CHANNEL_OPEN.
But what if the side not allowed to send DATA_CHANNEL_OPEN messages =
creates a data channel?
The JS API allows both sides to create data channels.
>=20
>=20
>>>> However, there is no mechanism to ensure uniqueness for labels. The =
WebRTC API already allows one side to create n data channels with the =
same label.
>>>=20
>>> I am not saying it should be forbidden to do so. But, for protocol =
X, I want to end up with ONE data channel between my peers.
>>> Then only one side should create a data channel.
>>>=20
>>> If I end up with two data channels for protocol X, can I send data =
on whichever I want to?
>> Sure. These would be two independent bidirectional data channels.
>=20
> Possible for the same protocol - and that is what I would want to =
avoid. I want to send my protocol X messages on one data channel, and I =
would like to receive the protocol X messages from my peer on the same =
data channel.
That is fine. Then you have two choices:
* Use the data channel (establishment) protocol and manage yourself that =
only
  on side creates a data channel.
* Don't use the data channel protocol and create the channel manually. =
You need
  to negotiate the parameters out of band.
>=20
> Now, one can of course solve this by reseting one of the data =
channels, but then the question is which endpoint should reset :)
You could always close the data channel with the lower stream id. Both =
sides can close it...

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20


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To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, =?iso-8859-1?Q?'Stefan_H=E5kansson_LK'?= <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Magnus,

Thanks for proposing a new formulation of the note by the authors. I =
agree
with you that F2 and F29 are generic WebRTC NAT/Firewall requirements =
which
does not restrict ICE-TCP as one of the solution.

The confusion with F31 and F32 requirements is because of the missing of
WebRTC server/gateway requirements. F31 and F32 leads to the assumption =
of
mandating TURN servers in the WebRTC server/gateway (by some of the =
WebRTC
gateway Providers) as there is no text in this usecase & requirement
document to clarify. I have already wrote in the separate mail thread
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg11338.html) to
provide WebRTC server/gateway guidelines document for more clarity.

Thanks
Partha

Note:
   F31     The browser must be able to use several STUN
           and TURN servers
   ----------------------------------------------------------------
   F32     The browser must support the use of STUN and TURN
           servers that are supplied by entities other than
           the web application (i.e. the network provider).
   ----------------------------------------------------------------

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, February 10, 2014 4:32 PM
> To: Parthasarathi R; 'Stefan H=E5kansson LK'; rtcweb@ietf.org
> Subject: Re: [rtcweb] New version of use-case draft uploaded
>=20
> Hi Partha,
>=20
> See inline
>=20
> On 2014-02-08 10:12, Parthasarathi R wrote:
> > Hi Stefan,
> >
> > <snip>
> >> Yes, I did that change (TURN -> ICE). My understanding was that ICE
> is
> >> what is used/mandated (and it in turn makes use of STUN and TURN).
> >>
> > </snip>
> >
> > ICE is mandated in RTCWeb but I disagree to your assumption that ICE
> in turn
> > makes use of TURN. My concern is that The requirement document is
> misleading
> > about TURN requirements. ICE-TCP avoids TURN server as the =
middle-box
> in the
> > network. ICE-TCP is host candidate whereas TURN is relay candidate
> and as
> > per ICE candidate selection, host candidate is preferred over relay
> > candidates.
>=20
> I will agree that the note as currently worded is not completely
> accurate and clear. Based on the text in the preceding section, it
> makes
> sense to note that ICE using STUN and TURN is not the only solution
> including additional ICE modes or relay services. I request the =
authors
> propose a new formulation of the note.
>=20
> >
> > In case of WebRTC servers (like conference)/gateway implementation,
> WebRTC
> > server is never going to be behind firewall, ICE-TCP serve the
> purpose and
> > there is no need of TURN server as a middle-box in the network. So,
> Could
> > you please correct the F31, F32 requirement text in the next =
revision
> of the
> > draft.
> >
>=20
> This is a request to change two requirements. I have not seen a clear
> consensus on this change. Thus, the change is rejected for now.
>=20
> From my perspective your proposal for ICE-TCP based solution is a
> choice
> to fulfill F2 and F29, not F31 and F32. Thus, I don't see an issue for
> F31 and F32 being STUN and TURN specific.
>=20
> Is really the current formulations in the requirement documents a
> hinder
> against the WG considering an ICE-TCP based traversal addition? I =
don't
> see it, and if you do, can you please be clearer in your explanation
> what the issue is? If not, I think your time is better spent working =
on
> building a clear motivation why ICE-TCP is needed to help establish a
> WG
> consensus on this.
>=20
> Cheers
>=20
> Magnus Westerlund
> (As co-chair)
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>> This is defined in RFC 5763 S 5:
>>> http://tools.ietf.org/html/rfc5763#section-5
>>>=20
>>> Which points to:
>>> http://tools.ietf.org/html/rfc4145
>>=20
>> Ok. So, a few questions to for clarification:
>>=20
>> Q1: This means that the JS App must set the setup attrbute value before =
passing an SDP to the browser?
> No, peer-to-peer browser calls should 'do the right thing' - based on the=
 initator/ice-controlling rule Both browsers set a=3Dsetup:act-pass meaning=
 that we fall back to the old rules.

What "old rules"? :)

>> Q2: If SDP O/A is not used on the wire, there needs to be another mechan=
ism for the peers to negotiate/indicate who is "active" and who is "passive=
"?
>>=20
> If you don't care, and are locally generating the SDP for a remote offer =
or answer, set a=3Dsetup:act-pass and the roles sort themselves out.

According to the "old rules", right? :)

>If however you have a gateway initating a call to a browser, and it wants =
to do early media (it's an 800 number gateway for example) then it sets a=
=3Dsetup:active
>in the SDP it sends (or you create on it's behalf).

There doesn't necessarily have to be a gateway. It may be a browser-to-brow=
ser call, using SIP and SDP O/A on the wire - meaning that the SDP setup at=
tribute is used to determine the roles.

>> Q3: If you have mulitple m- lines, all using the same DTLS association, =
the setup attribute value must be identical in all m- lines?
>>=20
>
> Yes, I think so.
>
>> Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using the=
 setup attribute value) as DTLS/SRTP for determining the TLS role?
>
> That's the way chrome seems to implement it - and presumably the way the =
spec is intended.

Ok.

Regards,

Christer


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Subject: Re: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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On Mon, Feb 10, 2014 at 11:55 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Peter,
>
> > If you have to wait for the ACK, it's one more RTT you have to wait
> before you can send data.  We've already
> > got many RTTs form ICE and DTLS and what's built into SCTP.  We
> originally didn't have an ACK message at all, in
> > large part because we didn't want more RTTs.  The ACK was put in to
> handle the edge case of unordered data arriving
> > before the OPEN message.  But we added it in a way that it doesn't add
> more RTTs.
>
> But, you may not even know whether I support the protocol you are sending=
.
> What should I do if I don't like/support what you send?
>
>
=E2=80=8BThe signalling is up to the application to define.  If your applic=
ation
wants to send a message from the receiver to the sender saying "I support
X" or "please send me X", then by all means=E2=80=8B add that to your appli=
cation,
and don't call channel.send() until the sender knows it should send.



> I think it would be useful to have a REJ message, so that I can explicitl=
y
> tell you that I don't want what you send - rather than me reseting the
> stream, or me not sending an ACK and hope that you will eventually stop
> sending data.
>

=E2=80=8B=E2=80=8BThe signalling is up to the application to define.  If yo=
ur application
wants to send a message from the receiver to the sender saying "I don't
support X" or "please don't send me X", then by all means=E2=80=8B add that=
 to your
application, and don't call channel.send() if the receiver says it doesn't
want the message.


I think what is already in the standard is sufficient for your use case.
 You just need to add a little application-specific signalling.  I don't
think the in-band signalling is the right place to put things like this.



>
> Regards,
>
> Christer
>
>

--047d7b6dc31ed9aa4404f212db59
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:verdana,=
sans-serif"><br></div><div class=3D"gmail_extra"><br><br><div class=3D"gmai=
l_quote">On Mon, Feb 10, 2014 at 11:55 AM, Christer Holmberg <span dir=3D"l=
tr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank"=
>christer.holmberg@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">Hi Peter,<br>
<div class=3D""><br>
&gt; If you have to wait for the ACK, it&#39;s one more RTT you have to wai=
t before you can send data. =C2=A0We&#39;ve already<br>
&gt; got many RTTs form ICE and DTLS and what&#39;s built into SCTP. =C2=A0=
We originally didn&#39;t have an ACK message at all, in<br>
&gt; large part because we didn&#39;t want more RTTs. =C2=A0The ACK was put=
 in to handle the edge case of unordered data arriving<br>
&gt; before the OPEN message. =C2=A0But we added it in a way that it doesn&=
#39;t add more RTTs.<br>
<br>
</div>But, you may not even know whether I support the protocol you are sen=
ding. What should I do if I don&#39;t like/support what you send?<br>
<br></blockquote><div><br></div><div><div class=3D"gmail_default" style=3D"=
font-family:verdana,sans-serif">=E2=80=8BThe signalling is up to the applic=
ation to define. =C2=A0If your application wants to send a message from the=
 receiver to the sender saying &quot;I support X&quot; or &quot;please send=
 me X&quot;, then by all means=E2=80=8B add that to your application, and d=
on&#39;t call channel.send() until the sender knows it should send.</div>

<br></div><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 think it would be useful to have a REJ message, so that I can explicitly =
tell you that I don&#39;t want what you send - rather than me reseting the =
stream, or me not sending an ACK and hope that you will eventually stop sen=
ding data.<br>

</blockquote><div><br></div><div><div class=3D"gmail_default" style=3D"font=
-family:verdana,sans-serif">=E2=80=8B=E2=80=8BThe signalling is up to the a=
pplication to define. =C2=A0If your application wants to send a message fro=
m the receiver to the sender saying &quot;I don&#39;t support X&quot; or &q=
uot;please don&#39;t send me X&quot;, then by all means=E2=80=8B add that t=
o your application, and don&#39;t call channel.send() if the receiver says =
it doesn&#39;t want the message.</div>

<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif"><br><=
/div><div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif">=
<br></div><div class=3D"gmail_default" style=3D"font-family:verdana,sans-se=
rif">I think what is already in the standard is sufficient for your use cas=
e. =C2=A0You just need to add a little application-specific signalling. =C2=
=A0I don&#39;t think the in-band signalling is the right place to put thing=
s like this.</div>

<br></div><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">
<br>
Regards,<br>
<br>
Christer<br>
<br>
</blockquote></div><br></div></div>

--047d7b6dc31ed9aa4404f212db59--


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Thread-Topic: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 10 Feb 2014, at 20:03, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>> This is defined in RFC 5763 S 5:
>>>> http://tools.ietf.org/html/rfc5763#section-5
>>>>=20
>>>> Which points to:
>>>> http://tools.ietf.org/html/rfc4145
>>>=20
>>> Ok. So, a few questions to for clarification:
>>>=20
>>> Q1: This means that the JS App must set the setup attrbute value =
before passing an SDP to the browser?
>> No, peer-to-peer browser calls should 'do the right thing' - based on =
the initator/ice-controlling rule Both browsers set a=3Dsetup:act-pass =
meaning that we fall back to the old rules.
>=20
> What "old rules"? :)
>=20

Sorry that was unclear - As I read it, putting=20
a=3Dsetup:actpass=20
in the SDP indicates that the ice-controlling entity should act as DTLS =
client, and typically the ice-controlling entity is the initiator of the =
session. So it isn't a matter of O/A negotiation.

>>> Q2: If SDP O/A is not used on the wire, there needs to be another =
mechanism for the peers to negotiate/indicate who is "active" and who is =
"passive"?
>>>=20
>> If you don't care, and are locally generating the SDP for a remote =
offer or answer, set a=3Dsetup:act-pass and the roles sort themselves =
out.
>=20
> According to the "old rules", right? :)

Yeah, Ice controlling is the DTLS client.=20

>=20
>> If however you have a gateway initating a call to a browser, and it =
wants to do early media (it's an 800 number gateway for example) then it =
sets a=3Dsetup:active
>> in the SDP it sends (or you create on it's behalf).
>=20
> There doesn't necessarily have to be a gateway. It may be a =
browser-to-browser call, using SIP and SDP O/A on the wire - meaning =
that the SDP setup attribute is used to determine the roles.

But there is no concept of early media on a peer-to-peer webRTC call, so =
there isn't any need to manage which side is the DTLS client or server.

>=20
>>> Q3: If you have mulitple m- lines, all using the same DTLS =
association, the setup attribute value must be identical in all m- =
lines?
>>>=20
>>=20
>> Yes, I think so.
>>=20
>>> Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using =
the setup attribute value) as DTLS/SRTP for determining the TLS role?
>>=20
>> That's the way chrome seems to implement it - and presumably the way =
the spec is intended.
>=20
> Ok.
>=20
> Regards,
>=20
> Christer


From Michael.Tuexen@lurchi.franken.de  Mon Feb 10 12:23:56 2014
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Subject: Re: [rtcweb] Data Channel Protocol: Data before DATA_CHANNEL_ACK
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On Feb 10, 2014, at 9:18 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
>=20
>> I think what is already in the standard is sufficient for your use =
case.  You just need to add a little application-specific signalling.  I =
don't think the in-band signalling is the right place to put things like =
this.
>=20
> If it is sufficient to use in-band signalling to say =94I would like =
to send you this data=94, I think it is equally sufficient to use =
in-band signaling to reply =93No, I don=92t want that data=94...
>=20
> Anyway, if people don't think it's needed, then I guess one can simply =
reset the stream.
Exactly. If the peer resets the stream before sending a =
DATA_CHANNEL_ACK, it is an error.
Section 6 of draft-ietf-rtcweb-data-protocol-02.txt contains

Therefore, receiving an SCTP stream reset request for a stream on which =
no DATA_CHANNEL_ACK message has been received indicates to the sender of =
the corresponding DATA_CHANNEL_OPEN message the failure of the data =
channel setup procedure.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>>>> This is defined in RFC 5763 S 5:
>>>>> http://tools.ietf.org/html/rfc5763#section-5
>>>>>=20
>>>>> Which points to:
>>>>> http://tools.ietf.org/html/rfc4145
>>>>=20
>>>> Ok. So, a few questions to for clarification:
>>>>=20
>>>> Q1: This means that the JS App must set the setup attrbute value befor=
e passing an SDP to the browser?
>>> No, peer-to-peer browser calls should 'do the right thing' - based on t=
he initator/ice-controlling rule Both browsers set a=3Dsetup:act-pass meani=
ng that we fall back to the old rules.
>>=20
>> What "old rules"? :)
>>=20
>
> Sorry that was unclear - As I read it, putting a=3Dsetup:actpass in the S=
DP indicates that the ice-controlling entity should act as DTLS client, and=
 typically the ice-controlling entity is the initiator of the session. So i=
t isn't a matter of O/A negotiation.

Do you have a reference to where you read that?

>>> If however you have a gateway initating a call to a browser, and it=20
>>> wants to do early media (it's an 800 number gateway for example) then i=
t sets a=3Dsetup:active in the SDP it sends (or you create on it's behalf).
>>=20
>> There doesn't necessarily have to be a gateway. It may be a browser-to-b=
rowser call, using SIP and SDP O/A on the wire - meaning that the SDP setup=
 attribute is used to determine the roles.
>
> But there is no concept of early media on a peer-to-peer webRTC call, so =
there isn't any need to manage which side is the DTLS client or server.

Not sure what this has to do with early media. You can have two browser bas=
ed applications, using SDP O/A to negotiate the DTLS roles.

Regards,

Christer


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Subject: Re: [rtcweb] How to determine TLS roles?
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--bcaec547c741fa1d3704f2136056
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On Mon, Feb 10, 2014 at 8:37 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Hi,
>
> >This is defined in RFC 5763 S 5:
> >http://tools.ietf.org/html/rfc5763#section-5
> >
> >Which points to:
> >http://tools.ietf.org/html/rfc4145
>
> Ok. So, a few questions to for clarification:
>
> Q1: This means that the JS App must set the setup attrbute value before
> passing an SDP to the browser?
>

No. The browser does it.

Q2: If SDP O/A is not used on the wire, there needs to be another mechanism
> for the peers to negotiate/indicate who is "active" and who is "passive"?
>

I don't see how this is our problem.



> Q3: If you have mulitple m- lines, all using the same DTLS association,
> the setup attribute value must be identical in all m- lines?
>

You mean because they are bundled? This should follow the same rules we
generally
use for things that are bundled.


> Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using the
> setup attribute value) as DTLS/SRTP for determining the TLS role?
>

I would think so.

-Ekr

Regards,
>
> Christer
>
>
>
>
>
>
>
> On Mon, Feb 10, 2014 at 6:23 AM, Christer Holmberg <
> christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>>
> wrote:
> Hi,
>
> >>>> If I understand correctly, in RTCWEB we are going to use a single
> DTLS association for two things:
> >>>>
> >>>> 1)      Key exchange for SRTP
> >>>> 2)      The data channel
> >>>>
> >>>> Now, this means that there needs to be a generic way to determine the
> TLS roles.
> >>>>
> >>>> In WEBRTC, how are the TLS roles determined?
> >>>
> >>> That was the subject of a rant of mine.
> >>>
> >>> I think the correct answer is that the party that ends up being
> IceControlling then becomes the DTLS client. Typically the initiator of a
> session is IceControlling.
> >>>
> >>> However the recent addition of
> >>>> a=setup:passive
> >>> to chrome's SDP clouds the story somewhat, allowing an IceController
> to say that it is DTLS passive and allow the non-initiator to be the DTLS
> client.
> >>>
> >>> Ostensibly this is to support early media - where the receiver of a
> call wishes to send media before the initiator does.
> >>>
> >>> As I said (rather intemperately) this added complexity is not worth
> the benefit.
> >>
> >> I think that it should be possible for a JS App to explicitly set the
> roles.
> >>
> >> Because, when SDP O/A is used on the wire, there are specified rules on
> how the roles are determined. But, some other on-the-wire protocol may have
> different rules.
> >>
> > There are many things that the JS app will need to do if we aren't bound
> by SDP O/A - but I agreed not to talk about that - until after 1.0 gets out
> :-)
>
> Fair enough, but for 1.0 we still need to know how the roles are
> determined :)
>
> The security-arch document does say that a DTLS handshake is performed on
> each channel that has been established by ICE, but I don't find anything
> regarding the roles.
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto: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, Feb 10, 2014 at 8:37 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@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"><br>
Hi,<br>
<div class=3D""><br>
&gt;This is defined in RFC 5763 S 5:<br>
&gt;<a href=3D"http://tools.ietf.org/html/rfc5763#section-5" target=3D"_bla=
nk">http://tools.ietf.org/html/rfc5763#section-5</a><br>
&gt;<br>
&gt;Which points to:<br>
&gt;<a href=3D"http://tools.ietf.org/html/rfc4145" target=3D"_blank">http:/=
/tools.ietf.org/html/rfc4145</a><br>
<br>
</div>Ok. So, a few questions to for clarification:<br>
<br>
Q1: This means that the JS App must set the setup attrbute value before pas=
sing an SDP to the browser?<br></blockquote><div><br></div><div>No. The bro=
wser does it.=A0</div><div><br></div><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


Q2: If SDP O/A is not used on the wire, there needs to be another mechanism=
 for the peers to negotiate/indicate who is &quot;active&quot; and who is &=
quot;passive&quot;?<br></blockquote><div><br></div><div>I don&#39;t see how=
 this is our problem.</div>

<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Q3: If you have mulitple m- lines, all using the same DTLS association, the=
 setup attribute value must be identical in all m- lines?<br></blockquote><=
div><br></div><div>You mean because they are bundled? This should follow th=
e same rules we generally</div>

<div>use for things that are bundled.</div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
Q4: The data channel (DTLS/SCTP) will have the same rule (e.g. using the se=
tup attribute value) as DTLS/SRTP for determining the TLS role?<br></blockq=
uote><div><br></div><div>I would think so.</div><div><br></div><div>-Ekr</d=
iv>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
Regards,<br>
<br>
Christer<br>
<div class=3D""><br>
<br>
<br>
<br>
<br>
<br>
<br>
On Mon, Feb 10, 2014 at 6:23 AM, Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&lt;mailto:=
<a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsso=
n.com</a>&gt;&gt; wrote:<br>


Hi,<br>
<br>
&gt;&gt;&gt;&gt; If I understand correctly, in RTCWEB we are going to use a=
 single DTLS association for two things:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; 1) =A0 =A0 =A0Key exchange for SRTP<br>
&gt;&gt;&gt;&gt; 2) =A0 =A0 =A0The data channel<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Now, this means that there needs to be a generic way to de=
termine the TLS roles.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; In WEBRTC, how are the TLS roles determined?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; That was the subject of a rant of mine.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I think the correct answer is that the party that ends up bein=
g IceControlling then becomes the DTLS client. Typically the initiator of a=
 session is IceControlling.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; However the recent addition of<br>
&gt;&gt;&gt;&gt; a=3Dsetup:passive<br>
&gt;&gt;&gt; to chrome&#39;s SDP clouds the story somewhat, allowing an Ice=
Controller to say that it is DTLS passive and allow the non-initiator to be=
 the DTLS client.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Ostensibly this is to support early media - where the receiver=
 of a call wishes to send media before the initiator does.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; As I said (rather intemperately) this added complexity is not =
worth the benefit.<br>
&gt;&gt;<br>
&gt;&gt; I think that it should be possible for a JS App to explicitly set =
the roles.<br>
&gt;&gt;<br>
&gt;&gt; Because, when SDP O/A is used on the wire, there are specified rul=
es on how the roles are determined. But, some other on-the-wire protocol ma=
y have different rules.<br>
&gt;&gt;<br>
&gt; There are many things that the JS app will need to do if we aren&#39;t=
 bound by SDP O/A - but I agreed not to talk about that - until after 1.0 g=
ets out :-)<br>
<br>
Fair enough, but for 1.0 we still need to know how the roles are determined=
 :)<br>
<br>
The security-arch document does say that a DTLS handshake is performed on e=
ach channel that has been established by ICE, but I don&#39;t find anything=
 regarding the roles.<br>
<br>
Regards,<br>
<br>
Christer<br>
_______________________________________________<br>
rtcweb mailing list<br>
</div><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&lt;mailto:<a h=
ref=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>This is defined in RFC 5763 S 5:
>>http://tools.ietf.org/html/rfc5763#section-5
>>
>>Which points to:
>>http://tools.ietf.org/html/rfc4145
> Ok. So, a few questions to for clarification:
>
> Q1: This means that the JS App must set the setup attrbute value before p=
assing an SDP to the browser?
>
> No. The browser does it.=A0
>
>> Q2: If SDP O/A is not used on the wire, there needs to be another mechan=
ism for the peers to negotiate/indicate who is "active" and who is "passive=
"?
>
> I don't see how this is our problem.

Ok, let me rephrase: we use SDP O/A in the API between the JS App and the b=
rowser, and the RFCs you pointed to above say that the SDP setup attribute =
is used to negotiate the roles.=20

So, can the JS App, using the setup attribute, control the DTLS role in the=
 browser?

>> Q3: If you have mulitple m- lines, all using the same DTLS association, =
the setup attribute value must be identical in all m- lines?
>
> You mean because they are bundled? This should follow the same rules we g=
enerally use for things that are bundled.

We haven't specified the rules for the setup attribute yet, but I assume it=
 will have to be identical for each m- line within a BUNDLE group.

And, in case the m- lines are NOT bundled, there will be separate DTLS asso=
ciations, with separate roles, right?

Regards,

Christer


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Subject: Re: [rtcweb] How to determine TLS roles?
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--089e01183daa6d8b5c04f2149a4b
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On Mon, Feb 10, 2014 at 12:54 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>This is defined in RFC 5763 S 5:
> >>http://tools.ietf.org/html/rfc5763#section-5
> >>
> >>Which points to:
> >>http://tools.ietf.org/html/rfc4145
> > Ok. So, a few questions to for clarification:
> >
> > Q1: This means that the JS App must set the setup attrbute value before
> passing an SDP to the browser?
> >
> > No. The browser does it.
> >
> >> Q2: If SDP O/A is not used on the wire, there needs to be another
> mechanism for the peers to negotiate/indicate who is "active" and who is
> "passive"?
> >
> > I don't see how this is our problem.
>
> Ok, let me rephrase: we use SDP O/A in the API between the JS App and the
> browser, and the RFCs you pointed to above say that the SDP setup attribute
> is used to negotiate the roles.
>
> So, can the JS App, using the setup attribute, control the DTLS role in
> the browser


That ties into the general question of which a-lines can be modified in
the JS app. It's no more decided than those questions.



> >> Q3: If you have mulitple m- lines, all using the same DTLS association,
> the setup attribute value must be identical in all m- lines?
> >
> > You mean because they are bundled? This should follow the same rules we
> generally use for things that are bundled.
>
> We haven't specified the rules for the setup attribute yet, but I assume
> it will have to be identical for each m- line within a BUNDLE group.
>

OK.


> And, in case the m- lines are NOT bundled, there will be separate DTLS
> associations, with separate roles, right?
>

Yes.

-Ekr



> Regards,
>
> Christer
>

--089e01183daa6d8b5c04f2149a4b
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 10, 2014 at 12:54 PM, Christer Holmberg <span dir=3D"lt=
r">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">=
christer.holmberg@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"">Hi,<br>
<br>
&gt;&gt;This is defined in RFC 5763 S 5:<br>
&gt;&gt;<a href=3D"http://tools.ietf.org/html/rfc5763#section-5" target=3D"=
_blank">http://tools.ietf.org/html/rfc5763#section-5</a><br>
&gt;&gt;<br>
&gt;&gt;Which points to:<br>
&gt;&gt;<a href=3D"http://tools.ietf.org/html/rfc4145" target=3D"_blank">ht=
tp://tools.ietf.org/html/rfc4145</a><br>
&gt; Ok. So, a few questions to for clarification:<br>
&gt;<br>
&gt; Q1: This means that the JS App must set the setup attrbute value befor=
e passing an SDP to the browser?<br>
&gt;<br>
&gt; No. The browser does it.=A0<br>
&gt;<br>
&gt;&gt; Q2: If SDP O/A is not used on the wire, there needs to be another =
mechanism for the peers to negotiate/indicate who is &quot;active&quot; and=
 who is &quot;passive&quot;?<br>
&gt;<br>
&gt; I don&#39;t see how this is our problem.<br>
<br>
</div>Ok, let me rephrase: we use SDP O/A in the API between the JS App and=
 the browser, and the RFCs you pointed to above say that the SDP setup attr=
ibute is used to negotiate the roles.<br>
<br>
So, can the JS App, using the setup attribute, control the DTLS role in the=
 browser</blockquote><div><br></div><div>That ties into the general questio=
n of which a-lines can be modified in</div><div>the JS app. It&#39;s no mor=
e decided than those questions.</div>

<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D""=
>
&gt;&gt; Q3: If you have mulitple m- lines, all using the same DTLS associa=
tion, the setup attribute value must be identical in all m- lines?<br>
&gt;<br>
&gt; You mean because they are bundled? This should follow the same rules w=
e generally use for things that are bundled.<br>
<br>
</div>We haven&#39;t specified the rules for the setup attribute yet, but I=
 assume it will have to be identical for each m- line within a BUNDLE group=
.<br></blockquote><div><br></div><div>OK.</div><div>=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex">


And, in case the m- lines are NOT bundled, there will be separate DTLS asso=
ciations, with separate roles, right?<br></blockquote><div><br></div><div>Y=
es.</div><div><br></div><div>-Ekr</div><div><br></div><div>=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">


Regards,<br>
<br>
Christer<br>
</blockquote></div><br></div></div>

--089e01183daa6d8b5c04f2149a4b--


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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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You two guys are talking past one another.

Christer's point is that the *application" distributed between the two 
endpoints, needs a *single* instance of a channel for a particular 
application specific purpose. But the two ends are generally symmetric. 
There is no obvious choice of which side should create the channel. So 
how do we use the mechanism to achieve this result?

I think I can predict what Michael is going to say: it isn't a protocol 
issue - the two ends should negotiate this some other way. (In the 
normal rtcweb scenario, a *single* app in the middle can arrange for one 
side to do the open.)

This is a problem with a sip app - there is no middle, just two ends.

Previously Christer said:

> Couldn’t we e.g. say that, by default, the DTLS client is responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it could even use both an even or odd stream id value.

That is a bad thing for data channel in general. But I suppose the app 
can choose to follow this rule if it wants for particular channels.

	Thanks,
	Paul





On 2/10/14 12:29 PM, Christer Holmberg wrote:
>
> Hi,
>
>>>>>>> As is defined in the data channel protocol, an entity can send data once DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is received.
>>>>>>>
>>>>>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at the same time, using different stream id values, the result may be TWO data channels, with data sent on both.
>>>>>>>
>>>>>>> However, as is also indicated, the draft does not provide a mechanism to prevent/handle such glare situation.
>>>>>>>
>>>>>>> I personally think there shall be a way to prevent such situation, or otherwise I'm sure we are going to end up with interoperability problems.
>>>>>>>
>>>>>>> Couldn't we e.g. say that, by default, the DTLS client is responsible for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it could even use both an even or odd stream id value.
>>>>>> Section 4 reads:
>>>>>>
>>>>>> To avoid glare in opening Channels, each side MUST use either even
>>>>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>>>>>> used to determine which side uses odd or even is based on the
>>>>>> underlying DTLS connection role when used in WebRTC, with the side
>>>>>> acting as the DTLS client using even stream identifiers.
>>>>>>
>>>>>> So there is no problem in setting up data channels.
>>>>>>
>>>>>> This should not be confused by both sides setting up a data channel with the same Label. This ends up in two data channels with the same label.
>>>>>> You can even create more data channels with the same label.
>>>>>
>>>>> My point is that I don't want to end up with more than one channel.
>>>> And you won't, see above.
>>>
>>> I am not sure I understand. What text do you refer to?
>> To avoid glare in opening Channels, each side MUST use either even
>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>> used to determine which side uses odd or even is based on the
>> underlying DTLS connection role when used in WebRTC, with the side
>> acting as the DTLS client using even stream identifiers.
>
> That part I understand.
>
> But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the same protocol, and THAT is what I would like to avoid. I don't want to have two bi-directional data channels for the same thing.
>
> For that reason I suggested that we could also specify which endpoint is responsisble for sending DATA_CHANNEL_OPEN.
>
>
>>>> However, there is no mechanism to ensure uniqueness for labels. The WebRTC API already allows one side to create n data channels with the same label.
>>>
>>> I am not saying it should be forbidden to do so. But, for protocol X, I want to end up with ONE data channel between my peers.
>>> Then only one side should create a data channel.
>>>
>>> If I end up with two data channels for protocol X, can I send data on whichever I want to?
>> Sure. These would be two independent bidirectional data channels.
>
> Possible for the same protocol - and that is what I would want to avoid. I want to send my protocol X messages on one data channel, and I would like to receive the protocol X messages from my peer on the same data channel.
>
> Now, one can of course solve this by reseting one of the data channels, but then the question is which endpoint should reset :)
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>>> Q2: If SDP O/A is not used on the wire, there needs to be another mech=
anism for the peers to negotiate/indicate who is "active" and who is "passi=
ve"?
>>>
>>> I don't see how this is our problem.
>>Ok, let me rephrase: we use SDP O/A in the API between the JS App and the=
 browser, and the RFCs you pointed to above say that the SDP setup attribut=
e is used to negotiate the roles.
>>
>>So, can the JS App, using the setup attribute, control the DTLS role in t=
he browser
>
> That ties into the general question of which a-lines can be modified in
> the JS app. It's no more decided than those questions.

So, if I understand correctly, currently the JS App CANNOT control the brow=
ser DTLS role?

Regards,

Christer


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On Mon, Feb 10, 2014 at 11:56 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>>> Q2: If SDP O/A is not used on the wire, there needs to be another
> mechanism for the peers to negotiate/indicate who is "active" and who is
> "passive"?
> >>>
> >>> I don't see how this is our problem.
> >>Ok, let me rephrase: we use SDP O/A in the API between the JS App and
> the browser, and the RFCs you pointed to above say that the SDP setup
> attribute is used to negotiate the roles.
> >>
> >>So, can the JS App, using the setup attribute, control the DTLS role in
> the browser
> >
> > That ties into the general question of which a-lines can be modified in
> > the JS app. It's no more decided than those questions.
>
> So, if I understand correctly, currently the JS App CANNOT control the
> browser DTLS role?
>

Well, that may turn out to be the case, but it's not what I said.

JSEP Section 6 contains a list of which SDP parameters can
be manipulated by the JS, however, it's clearly unfinished.

http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05#section-6

Thus, it seems to me to be an open question whether the JS
can manipulate this field.

-Ekr

--047d7b33d050516a7304f21ce1a1
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 10, 2014 at 11:56 PM, Christer Holmberg <span dir=3D"lt=
r">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">=
christer.holmberg@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">Hi,<br>
<div class=3D""><br>
&gt;&gt;&gt;&gt; Q2: If SDP O/A is not used on the wire, there needs to be =
another mechanism for the peers to negotiate/indicate who is &quot;active&q=
uot; and who is &quot;passive&quot;?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I don&#39;t see how this is our problem.<br>
&gt;&gt;Ok, let me rephrase: we use SDP O/A in the API between the JS App a=
nd the browser, and the RFCs you pointed to above say that the SDP setup at=
tribute is used to negotiate the roles.<br>
&gt;&gt;<br>
&gt;&gt;So, can the JS App, using the setup attribute, control the DTLS rol=
e in the browser<br>
&gt;<br>
&gt; That ties into the general question of which a-lines can be modified i=
n<br>
&gt; the JS app. It&#39;s no more decided than those questions.<br>
<br>
</div>So, if I understand correctly, currently the JS App CANNOT control th=
e browser DTLS role?<br></blockquote><div><br></div><div>Well, that may tur=
n out to be the case, but it&#39;s not what I said.</div><div><br></div>

<div>JSEP Section 6 contains a list of which SDP parameters can</div><div>b=
e manipulated by the JS, however, it&#39;s clearly unfinished.</div><div><b=
r></div><div><a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-0=
5#section-6">http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05#section-6=
</a><br>

</div><div><br></div><div>Thus, it seems to me to be an open question wheth=
er the JS</div><div>can manipulate this field.</div><div><br></div><div>-Ek=
r</div><div><br></div></div></div></div>

--047d7b33d050516a7304f21ce1a1--


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On 11 Feb 2014, at 08:01, Eric Rescorla <ekr@rtfm.com> wrote:

>=20
>=20
>=20
> On Mon, Feb 10, 2014 at 11:56 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> >>>> Q2: If SDP O/A is not used on the wire, there needs to be another =
mechanism for the peers to negotiate/indicate who is "active" and who is =
"passive"?
> >>>
> >>> I don't see how this is our problem.
> >>Ok, let me rephrase: we use SDP O/A in the API between the JS App =
and the browser, and the RFCs you pointed to above say that the SDP =
setup attribute is used to negotiate the roles.
> >>
> >>So, can the JS App, using the setup attribute, control the DTLS role =
in the browser
> >
> > That ties into the general question of which a-lines can be modified =
in
> > the JS app. It's no more decided than those questions.
>=20
> So, if I understand correctly, currently the JS App CANNOT control the =
browser DTLS role?
>=20
> Well, that may turn out to be the case, but it's not what I said.
>=20
> JSEP Section 6 contains a list of which SDP parameters can
> be manipulated by the JS, however, it's clearly unfinished.
>=20
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05#section-6
>=20
> Thus, it seems to me to be an open question whether the JS
> can manipulate this field.

But be aware those rules apply to manipulation that happens between=20
createOffer and setLocalDescription

In concrete terms this means that a browser might create an offer (with =
setup:actpass)
set it unaltered as the local description, then send it to a peer.
The peer might reply with an answer with setup:active

This signals that the remote peer wants to be the DTLS client - and the =
offer said that the local browser was ok with that.
So this example allows the normal roles to be reversed, the =
non-initiator would be the DTLS client.

However, in a browser-to-browser call this may not be possible, because =
the JS at the far end may not be
allowed (see above) to change the setup: field between createAnswer() =
and setting it locally.

A gateway might not feel bound by such rules (code permitting).

Tim.

>=20
> -Ekr
>=20


--Apple-Mail=_F9F913AA-78E7-4BA2-9C1E-543A1977AB7D
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=iso-8859-1

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On 11 Feb 2014, at 08:01, Eric =
Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div=
 class=3D"gmail_quote">On Mon, Feb 10, 2014 at 11:56 PM, Christer =
Holmberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@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;padding-left:1ex">Hi,<br>
<div class=3D""><br>
&gt;&gt;&gt;&gt; Q2: If SDP O/A is not used on the wire, there needs to =
be another mechanism for the peers to negotiate/indicate who is "active" =
and who is "passive"?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I don't see how this is our problem.<br>
&gt;&gt;Ok, let me rephrase: we use SDP O/A in the API between the JS =
App and the browser, and the RFCs you pointed to above say that the SDP =
setup attribute is used to negotiate the roles.<br>
&gt;&gt;<br>
&gt;&gt;So, can the JS App, using the setup attribute, control the DTLS =
role in the browser<br>
&gt;<br>
&gt; That ties into the general question of which a-lines can be =
modified in<br>
&gt; the JS app. It's no more decided than those questions.<br>
<br>
</div>So, if I understand correctly, currently the JS App CANNOT control =
the browser DTLS role?<br></blockquote><div><br></div><div>Well, that =
may turn out to be the case, but it's not what I =
said.</div><div><br></div>

<div>JSEP Section 6 contains a list of which SDP parameters =
can</div><div>be manipulated by the JS, however, it's clearly =
unfinished.</div><div><br></div><div><a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05#section-6">ht=
tp://tools.ietf.org/html/draft-ietf-rtcweb-jsep-05#section-6</a><br>

</div><div><br></div><div>Thus, it seems to me to be an open question =
whether the JS</div><div>can manipulate this =
field.</div></div></div></div></blockquote><div><br></div><div>But be =
aware those rules apply to manipulation that happens =
between&nbsp;</div><div>createOffer and =
setLocalDescription</div><div><br></div><div>In concrete terms this =
means that a browser might create an offer (with =
setup:actpass)</div><div>set it unaltered as the local description, then =
send it to a peer.</div><div>The peer might reply with an answer with =
setup:active</div><div><br></div><div>This signals that the remote peer =
wants to be the DTLS client - and the offer said that the local browser =
was ok with that.</div><div>So this example allows the normal roles to =
be reversed, the non-initiator would be the DTLS =
client.</div><div><br></div><div>However, in a browser-to-browser call =
this may not be possible, because the JS at the far end may not =
be</div><div>allowed (see above) to change the setup: field between =
createAnswer() and setting it locally.</div><div><br></div><div>A =
gateway might not feel bound by such rules (code =
permitting).</div><div><br></div>Tim.</div><div><br><blockquote =
type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><div><br></div><div>-Ekr</div><div><br></div></div><=
/div></div>
</blockquote></div><br></body></html>=

--Apple-Mail=_F9F913AA-78E7-4BA2-9C1E-543A1977AB7D--


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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Parthasarathi R <partha@parthasarathi.co.in>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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On 2014-02-10 12:02, Magnus Westerlund wrote:=0A=
> Hi Partha,=0A=
>=0A=
> See inline=0A=
>=0A=
> On 2014-02-08 10:12, Parthasarathi R wrote:=0A=
>> Hi Stefan,=0A=
>>=0A=
>> <snip>=0A=
>>> Yes, I did that change (TURN -> ICE). My understanding was that ICE is=
=0A=
>>> what is used/mandated (and it in turn makes use of STUN and TURN).=0A=
>>>=0A=
>> </snip>=0A=
>>=0A=
>> ICE is mandated in RTCWeb but I disagree to your assumption that ICE in =
turn=0A=
>> makes use of TURN. My concern is that The requirement document is mislea=
ding=0A=
>> about TURN requirements. ICE-TCP avoids TURN server as the middle-box in=
 the=0A=
>> network. ICE-TCP is host candidate whereas TURN is relay candidate and a=
s=0A=
>> per ICE candidate selection, host candidate is preferred over relay=0A=
>> candidates.=0A=
>=0A=
> I will agree that the note as currently worded is not completely=0A=
> accurate and clear. Based on the text in the preceding section, it makes=
=0A=
> sense to note that ICE using STUN and TURN is not the only solution=0A=
> including additional ICE modes or relay services. I request the authors=
=0A=
> propose a new formulation of the note.=0A=
=0A=
Proposal:=0A=
=0A=
"Note that ICE support being mandatory does not preclude a WebRTC =0A=
endpoint from supporting more traversal mechanisms than ICE using STUN =0A=
and TURN."=0A=
=0A=
OK?=0A=
=0A=


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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 10 Feb 2014, at 20:30, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>>>> This is defined in RFC 5763 S 5:
>>>>>> http://tools.ietf.org/html/rfc5763#section-5
>>>>>>=20
>>>>>> Which points to:
>>>>>> http://tools.ietf.org/html/rfc4145
>>>>>=20
>>>>> Ok. So, a few questions to for clarification:
>>>>>=20
>>>>> Q1: This means that the JS App must set the setup attrbute value =
before passing an SDP to the browser?
>>>> No, peer-to-peer browser calls should 'do the right thing' - based =
on the initator/ice-controlling rule Both browsers set a=3Dsetup:act-pass =
meaning that we fall back to the old rules.
>>>=20
>>> What "old rules"? :)
>>>=20
>>=20
>> Sorry that was unclear - As I read it, putting a=3Dsetup:actpass in =
the SDP indicates that the ice-controlling entity should act as DTLS =
client, and typically the ice-controlling entity is the initiator of the =
session. So it isn't a matter of O/A negotiation.
>=20
> Do you have a reference to where you read that?

No, I don't think I have seen it explicitly stated.


>=20
>>>> If however you have a gateway initating a call to a browser, and it=20=

>>>> wants to do early media (it's an 800 number gateway for example) =
then it sets a=3Dsetup:active in the SDP it sends (or you create on it's =
behalf).
>>>=20
>>> There doesn't necessarily have to be a gateway. It may be a =
browser-to-browser call, using SIP and SDP O/A on the wire - meaning =
that the SDP setup attribute is used to determine the roles.
>>=20
>> But there is no concept of early media on a peer-to-peer webRTC call, =
so there isn't any need to manage which side is the DTLS client or =
server.
>=20
> Not sure what this has to do with early media. You can have two =
browser based applications, using SDP O/A to negotiate the DTLS roles.

You only care about manipulating the DTLS roles if you want the side =
that _didn't_ initiate the call to initiate the media flow. The only =
instance
where that happens is early media.=20
Without early media there is no need to manipulate which end is =
client/server since the default or initiator is client makes sense - or =
am I missing something?

>=20
> Regards,
>=20
> Christer


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To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>>>> If however you have a gateway initating a call to a browser, and it=20
>>>>> wants to do early media (it's an 800 number gateway for example) then=
 it sets a=3Dsetup:active in the SDP it sends (or you create on it's behalf=
).
>>>>=20
>>>> There doesn't necessarily have to be a gateway. It may be a browser-to=
-browser call, using SIP and SDP O/A on the wire - meaning that the SDP set=
up attribute is used to determine the roles.
>>>=20
>>> But there is no concept of early media on a peer-to-peer webRTC call, s=
o there isn't any need to manage which side is the DTLS client or server.
>>=20
>> Not sure what this has to do with early media. You can have two browser =
based applications, using SDP O/A to negotiate the DTLS roles.
>
> You only care about manipulating the DTLS roles if you want the side that=
 _didn't_ initiate the call to initiate the media flow. The only instance w=
here that happens is early media.=20
> Without early media there is no need to manipulate which end is client/se=
rver since the default or initiator is client makes sense - or am I missing=
 something?

I don't really have a need to "manipulate" the roles. I just want my JS App=
 to know what role its browser will take, so it can inform its peer (using =
whatever signaling protocol) about the role.

Regards,

Christer



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Subject: Re: [rtcweb] New version of use-case draft uploaded
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On 2014-02-10 20:09, Parthasarathi R wrote:
> Hi Magnus,
> 
> Thanks for proposing a new formulation of the note by the authors. I agree
> with you that F2 and F29 are generic WebRTC NAT/Firewall requirements which
> does not restrict ICE-TCP as one of the solution.
> 
> The confusion with F31 and F32 requirements is because of the missing of
> WebRTC server/gateway requirements. F31 and F32 leads to the assumption of
> mandating TURN servers in the WebRTC server/gateway (by some of the WebRTC
> gateway Providers) as there is no text in this usecase & requirement
> document to clarify.

Sorry, I fail to understand which of multiple possible angles you are
arguing here.

A) That the use case document is missing a whole requirement related to
WebRTC Servers. WebRTC server, is something I am uncertain to what you
refer to, a Web server or a media plan related server dealing with peer
connections?

B) That F31 and F32 would lead to an assumption of mandating something,
which currently is the only solution described?

C) That there needs to be text describing some aspect of the global
presence?


 I have already wrote in the separate mail thread
> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg11338.html) to
> provide WebRTC server/gateway guidelines document for more clarity.

Good, can you leave with the text and Stefan's text proposal for the note?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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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 Channels
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-07.txt
	Pages           : 15
	Date            : 2014-02-11

Abstract:
   The Real-Time Communication in WEB-browsers 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-browsers 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-07

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http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-channel-07


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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 Establishment Protocol
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-protocol-03.txt
	Pages           : 12
	Date            : 2014-02-11

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 a simple protocol for establishing
   symmetric data channels between the peers.  It uses a two way
   handshake and allows sending of user data without waiting for the
   handshake to complete.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

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On Nov 11, 2013, at 2:54 PM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> Hi,
>=20
> I did review the data channel draft prior to the meeting and thought I
> would provide my comments.
>=20
> 1. Usage of RTCWeb vs WebRTC when referring to the whole solution. My
> recollection was that we agreed to use WebRTC for this. I think we
> should try to align this language. Audio, Security and RTP usage uses
> WebRTC, while Data channel, overview and Transport uses RTCWeb.
OK. Changed to WebRTC. Although the WG uses rtcweb as its acronym...
>=20
> 2. Section 3.2:
>   U-C 6:  Renegotiation of the set of media streams in the
>      PeerConnection.
>=20
> What "media streams" are being referred to here. Or is i better to
> express this as "Renegotiation of the configuration of the =
PeerConnection."?
I think the media streams of the PeerConnection are considered...
However, you might want to reconfigure the PeerConnection. So, yes, your
text makes sense. I changed the use case.
>=20
> 3. Section 3.2:
>   U-C 7:  Proxy browsing, where a browser uses data channels of a
>      PeerConnection to send and receive HTTP/HTTPS requests and data,
>      for example to avoid local internet filtering or monitoring.
>=20
> Yes, this may be something that is possible, but to express it as a =
use
> cases intended to be supported might not be the best. We are after all
> likely talking about circumventing local security policies. I would =
also
> note that "Internet" is with capital I.
Change to Internet. The use case seems important, since there are =
already
implementations doing this (possibly not using data channels, don't =
know).
Randell: What do you think?
>=20
> 4. Section 4:
>   Req. 1:   Multiple simultaneous data channels MUST be supported.
>      Note that there may 0 or more media streams in parallel with the
>      data channels, and the number and state (active/inactive) of the
>      media streams may change at any time.
>=20
> Which "media stream" are you referring to in the above?
The PeerConnection can have 0 or more media streams (SRTP on the wire)
in addition to data channels. I changed it to:

<t>Multiple simultaneous data channels MUST be supported.
Note that there may 0 or more media streams in parallel with the data =
channels
in the same PeerConnection, and the number and state (active/inactive) =
of
these media streams may change at any time.</t>

>=20
> 5. Section 4:
>   Req. 3:   Data channels MUST be congestion controlled; either
>      individually, as a class, or in conjunction with the media
>      streams, to ensure that data channels don't cause congestion
>      problems for the media streams, and that the RTCWeb =
PeerConnection
>      as a whole is fair with competing traffic such as TCP.
>=20
> A: What "media stream" are you referring to here?
The media streams of the PeerConnection...
>=20
> B: "and that the RTCWeb PeerConnection
>      as a whole is fair with competing traffic such as TCP."
>=20
> I don't think it is a MUST requirement. Neither am I convinced that a
> PeerConnection needs to be fair with an unspecified number of TCP
> traffic. It might be simplest to strike the last part.
But if you remove the last part the CC of the data channel only needs
to ensure that it doesn't case congestion problems for the media =
channels.
So if there are no media streams, the CC of the data channels could be
much more aggressive than TCP. I don't think this is what we want. So
I keep that for now. But I'm open for discussion.

Changed to:

<t>Data channels of a PeerConnection MUST be congestion controlled;
either individually, as a class, or in conjunction with the media =
streams
of the PeerConnection, to ensure that data channels don't cause =
congestion
problems for these media streams, and that the WebRTC PeerConnection as
a whole is fair with competing traffic such as TCP.</t>

>=20
> 6. Section 4:
> [ TBD: how this is encoded and what the impact of this is. ]
>=20
> Didn't we have some direction decisions on how priority was to be
> handled at least locally by the end-point. Please add this to the =
draft.
I think we decided to have priorities for data channels, which are not =
strict
priorities. That is all. The W3C document doesn't mention it.
> I also hope that the way of representing priorities can get clearer =
from
> an API perspective.
I agree. However, I'm not a member of W3C...
>=20
> 7. Section 5:
> "   o  The congestion control is modifiable for integration with media
>      stream congestion control."
>=20
> I think this is an exaggeration, and not supported by current sets of
> specifications, if it is supported please provide a reference.
I think you can specify an alternate CC for SCTP. This is not done
right now, but it is doable. Therefore, I don't think it is a =
exaggeration,
it is just not done yet... If you have a suggestion for an better =
wording,
I'm open for suggestions.
>=20
> 8. Section 5:
> "Using DTLS over UDP in combination with ICE enables NAT traversal in
> IPv4 based networks."
>=20
> I would like to point out that ICE do provide firewall traversal for
> firewalls with certain type of filtering rules like the basic stateful
> filtering. These apply to IPv6 equally, not only IPv4.
Correct. I changed the wording to

Using DTLS over UDP in combination with ICE enables middlebox traversal
in IPv4 and IPv6 based networks.

>=20
> 9.  Section 5:
> "   Each SCTP user message contains a so called Payload Protocol
>   Identifier (PPID) that is passed to SCTP by its upper layer and sent
>   to its peer.  This value can be used to multiplex multiple protocols
>   over a single SCTP association.  The sender provides for each
>   protocol a specific PPID and the receiver can demultiplex the
>   messages based on the received PPID."
>=20
> I have some difficulties in determining the relevance of this in the
> context of WebRTC. Can you please provide a clear explanation why PPID
> are relevant here. Will not all traffic from an WebRTC endpoint use =
the
> PPIDs associated with the data protocol, rather than any "native" =
PPIDs?
We use one PPID for the Data Channel Establishment Protocol, one for
UTF-8 encoded user data and one for binary encoded user data. So we use
it...

I added at the end of the paragraph:

The PPID is used to distinguish UTF-8 encoded user data and binary =
encoded
userdata. The Data Channel Establishment Protocol defined in
<xref target=3D'I-D.ietf-rtcweb-data-protocol'/> uses also a specific =
PPID to
be distinguished from user data.
>=20
> 10. Section 5:
>=20
> Can you please clarifiy the demultiplexing for WebRTC by discussing
> STUN, vs SRTP vs DTLS and secondly that DTLS-SRTP and DTLS frames with
> SCTP can co-exist in the same DTLS stack. At least on an level where =
you
> can point to all the relevant sections in other specifications to
> explain that these can coexist given certain constraints. This is =
likely
> its own sub-section.
Not sure...=20
The demultiplexing STUN vs. SRTP vs. DTLS is described in
http://tools.ietf.org/html/rfc5764#section-5.1.2
SCTP is the only payload of DTLS. So there is no demultiplexing
done. I have added:

Please note that the demultiplexing STUN vs. SRTP vs. DTLS is done
as described in Section 5.1.2 of <xref target=3D'RFC5764'/> and SCTP
is the only payload of DTLS.
>=20
> 11. Section 5:
>=20
>   o  shares the DTLS connection with the media channels.
>=20
> what is "media channels"?
... the media channels of the PeerConnection.
>=20
> 12. Section 5:
> "  Since DTLS is typically implemented in user-land, the SCTP stack =
also
>   needs to be a user-land stack."
>=20
> I wonder what the message of this sentence is? The reason is that I
That you can't use a kernel SCTP stack, even if available.
> wonder what it should be saying if one implements DTLS in a kernel, or
I haven't seen this yet. I think you can do the encrypt/decrypt part,
but doing the key management stuff in kernel land doesn't seem =
appropriate.
> similar if one despite a DTLS user-land stack uses a SCTP stack in the
> kernel through access to raw sockets.
How can the lower layer run in userland when the upper layer runs in
kernel land?
>=20
> 13. Section 5:
>   Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
>   layer, since there is no way to identify the corresponding
>   association.
>=20
> Also here I get a bit confused, I assume what you are after is the =
issue
> that incoming ICMP messages will be arriving related to the UDP flow,
> not in relation to an kernel SCTP association, thus there is a routing
> issue for ICMP to reach the above DTLS existing SCTP association. Not
> impossible to solve, just not available with existing APIs.
No, it is not an API issue. The ICMP message contains the encrypted
DTLS packet which contains the SCTP packet. Normally, only part of the =
message
is returned. I think only 8 bytes of payload is returned. So this is
not enough for the DTLS header at all. So you simply might not have
enough information.
>=20
> 14. Section 5:
>   "This protocol stack MUST support the usage of multiple SCTP =
streams."
>=20
> I wonder which layer is affected by this. Isn't this only affecting =
SCTP
> implementation and the higher layers? Can't you be more specific thus?
Yes. Changed to
<t>This SCTP stack and its upper layer MUST support the usage of =
multiple
SCTP streams.
>=20
> 15. Section 5:
>   Using a congestion control
>   different from the standard one might improve the impact on the
>   parallel SRTP media streams.  Since SCTP does not support the
>   negotiation of a congestion control algorithm, alternate congestion
>   controls SHOULD only require a different sender side behavior using
>   existing information carried in the association.
>=20
> A: I wonder if adding a "yet" into the second sentence to make clear
> that in the future there might be support for negotiating congestion
> control.
>=20
> B: Also, what happens if one violate the SHOULD and require receiver
> side information and the peer don't support it. That clearly can't be
> acceptable? Can this be formulated in some other way that is tighter
> without preventing sender side changes?
OK. Changed to
Since SCTP does not support the negotiation of a congestion control =
algorithm
yet, alternate congestion controls SHOULD either only require a =
different
sender side behavior using existing information carried in the =
association or
need also specify a negotiation of of a congestion control =
algorithm.</t>
>=20
> 16. Section 6.1:
>   o  The dynamic address reconfiguration extension defined in =
[RFC5061]
>      MUST be used to signal the support of the stream reset extension
>      defined in [RFC6525], other features of [RFC5061] MUST NOT be
>      used.
>=20
> A question, are theses other features, destructive to the SCTP
> association, or simply unnecessary to implement? If it is the first,
> then I think MUST NOT be used is fine, but if else, why not simple say
> that they are NOT REQUIRED to be implemented?

Changed to not REQUIRED, since NOT REQUIRED is not an RFC 2119 term.
>=20
>=20
> 17. Section 6.1:
>   o  The partial reliability extension defined in [RFC3758] MUST be
>      supported.  In addition to the timed reliability PR-SCTP policy
>      defined in [RFC3758], the limited retransmission policy defined =
in
>      [I-D.tuexen-tsvwg-sctp-prpolicies] MUST be supported.
>=20
> So what is the status of this individual document, did it get adopted
> into TSVWG? I think this is an important question, as it is one thing
> where we can consider going forward without the extension and add it
> later when ready?
It has been adopted.
>=20
> 18. Section 6.2:
> "  Additionally,
>   the negotiation SHOULD include some type of congestion control
>   selection. "
>=20
> Okay, this appears very fuzzy. And what name space or specifications =
are
> you going to use to negotiate what the different options are?
Randell? Salvatore? I don't think there is anything specified yet.
>=20
> 19. Section 6.3:
> It will use the DTLS connection selected via SDP;
>   typically this will be shared via BUNDLE or equivalent with DTLS
>   connections used to key the DTLS-SRTP media streams.
>=20
> In which way is the DTSL connection "selected" via SDP?
I think it should read ICE, not SDP...
>=20
> 20. Section 6.2:
>=20
>   When one side wants to open a channel using external negotiation, it
>   picks a Stream.  This can be based on the DTLS role (the client =
picks
>   even stream identifiers, the server odd stream identifiers) or done
>   in a different way.  However, the application is responsible for
>   avoiding collisions with existing Streams.  If it attempts to re-use
>   a Stream which is part of an existing Channel, the addition SHOULD
>   fail.  In addition to choosing a Stream, the application SHOULD also
>   inform the protocol of the options to use for sending messages.  The
>   application MUST ensure in an application-specific manner that the
>   other side will also inform the protocol that the selected Stream is
>   to be used, and the parameters for sending data from that side.
>=20
> So why is this specified here and not in the Data Protocol? Especially
> using RFC 2119 terms, it appears to be colliding or at least =
overlapping
> specifications?
Since the Data Channel Establishment Protocol only covers setting up
the data channel via an internal mechanism. The above described the
external mechanism.
>=20
> 21. Section 6.6
>   It is recommended that message size be kept within certain size
>   bounds (TBD) as applications will not be able to support =
arbitrarily-
>   large single messages.
>=20
> I guess this will be resolved by the WG session agreement, or?
It will be signalled via SDP...
>=20
> 22.
>=20
> Section 8:
>              +---------------------+-----------+-----------+
>              | Value               | SCTP PPID | Reference |
>              +---------------------+-----------+-----------+
>              | DOMString Last      | 51        | [RFCXXXX] |
>              | Binary Data Partial | 52        | [RFCXXXX] |
>              | Binary Data Last    | 53        | [RFCXXXX] |
>              | DOMString Partial   | 54        | [RFCXXXX] |
>              +---------------------+-----------+-----------+
>=20
> Where are the definitions of these definitions?
In Section 6.6.
>=20
> 23. Section 10.2:
>=20
>   [I-D.tuexen-tsvwg-sctp-prpolicies]
>              Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
>              "Additional Policies for the Partial Delivery Extension =
of
>              the Stream Control Transmission Protocol", draft-tuexen-
>              tsvwg-sctp-prpolicies-03 (work in progress), October =
2013.
>=20
> This is a normative reference.
OK. Please note that that ID is targeting for an Informational RFC...

Base on the above, I submitted a new revision.

Best regards
Michael
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20


From tim@phonefromhere.com  Tue Feb 11 06:54:11 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] How to determine TLS roles?
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On 11 Feb 2014, at 10:49, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>>>> If however you have a gateway initating a call to a browser, and =
it=20
>>>>>> wants to do early media (it's an 800 number gateway for example) =
then it sets a=3Dsetup:active in the SDP it sends (or you create on it's =
behalf).
>>>>>=20
>>>>> There doesn't necessarily have to be a gateway. It may be a =
browser-to-browser call, using SIP and SDP O/A on the wire - meaning =
that the SDP setup attribute is used to determine the roles.
>>>>=20
>>>> But there is no concept of early media on a peer-to-peer webRTC =
call, so there isn't any need to manage which side is the DTLS client or =
server.
>>>=20
>>> Not sure what this has to do with early media. You can have two =
browser based applications, using SDP O/A to negotiate the DTLS roles.
>>=20
>> You only care about manipulating the DTLS roles if you want the side =
that _didn't_ initiate the call to initiate the media flow. The only =
instance where that happens is early media.=20
>> Without early media there is no need to manipulate which end is =
client/server since the default or initiator is client makes sense - or =
am I missing something?
>=20
> I don't really have a need to "manipulate" the roles. I just want my =
JS App to know what role its browser will take, so it can inform its =
peer (using whatever signaling protocol) about the role.

well - assuming you leave  a=3Dsetup as actpass (the default in the =
browser)  .
If you called createOffer, then you'll be DTLS client.=20
If you called createAnswer you'll be DTLS server.
That's assuming no weirdness in the ICE implementation. So your JS can =
_deduce_ the role but neither influence it nor _know_ it directly.

However, signalling it over your high level signalling protocol is the =
wrong way to do it (IMHO). Your peer should deduce the appropriate DTLS =
role from the 'signalling'=20
in the ICE packets, by looking at the ICE CONTROLLING flag.=20

>=20
> Regards,
>=20
> Christer
>=20
>=20


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To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Hi,

>>>>>>> If however you have a gateway initating a call to a browser, and it
>>>>>>> wants to do early media (it's an 800 number gateway for example) th=
en it sets a=3Dsetup:active in the SDP it sends (or you create on it's beha=
lf).
>>>>>>
>>>>>> There doesn't necessarily have to be a gateway. It may be a browser-=
to-browser call, using SIP and SDP O/A on the wire - meaning that the SDP s=
etup attribute is used to determine the roles.
>>>>>
>>>>> But there is no concept of early media on a peer-to-peer webRTC call,=
 so there isn't any need to manage which side is the DTLS client or server.
>>>>
>>>> Not sure what this has to do with early media. You can have two browse=
r based applications, using SDP O/A to negotiate the DTLS roles.
>>>
>>> You only care about manipulating the DTLS roles if you want the side th=
at _didn't_ initiate the call to initiate the media flow. The only instance=
 where that happens is early media.
>>> Without early media there is no need to manipulate which end is client/=
server since the default or initiator is client makes sense - or am I missi=
ng something?
>>
>> I don't really have a need to "manipulate" the roles. I just want my JS =
App to know what role its browser will take, so it can inform its peer (usi=
ng whatever signaling protocol) about the role.
>
> well - assuming you leave  a=3Dsetup as actpass (the default in the brows=
er)  .
> If you called createOffer, then you'll be DTLS client.
> If you called createAnswer you'll be DTLS server.
> That's assuming no weirdness in the ICE implementation. So your JS can _d=
educe_ the role but neither influence it nor _know_ it directly.
>
> However, signalling it over your high level signalling protocol is the wr=
ong way to do it (IMHO). Your peer should deduce the appropriate DTLS role =
from the 'signalling'
> in the ICE packets, by looking at the ICE CONTROLLING flag.

In SIP/SDP O/A it is done using the setup attribute (red: high level signal=
ling protocol), and JSEP is based on SDP O/A, so...

If we do NOT want to use the setup attribute to determine the DTLS roles, t=
hen I think we need to have a separate discussion about that.=20

Regards,

Christer


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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] How to determine TLS roles?
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On 11 Feb 2014, at 15:04, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>=20
> Hi,
>=20
>>>>>>>> If however you have a gateway initating a call to a browser, =
and it
>>>>>>>> wants to do early media (it's an 800 number gateway for =
example) then it sets a=3Dsetup:active in the SDP it sends (or you =
create on it's behalf).
>>>>>>>=20
>>>>>>> There doesn't necessarily have to be a gateway. It may be a =
browser-to-browser call, using SIP and SDP O/A on the wire - meaning =
that the SDP setup attribute is used to determine the roles.
>>>>>>=20
>>>>>> But there is no concept of early media on a peer-to-peer webRTC =
call, so there isn't any need to manage which side is the DTLS client or =
server.
>>>>>=20
>>>>> Not sure what this has to do with early media. You can have two =
browser based applications, using SDP O/A to negotiate the DTLS roles.
>>>>=20
>>>> You only care about manipulating the DTLS roles if you want the =
side that _didn't_ initiate the call to initiate the media flow. The =
only instance where that happens is early media.
>>>> Without early media there is no need to manipulate which end is =
client/server since the default or initiator is client makes sense - or =
am I missing something?
>>>=20
>>> I don't really have a need to "manipulate" the roles. I just want my =
JS App to know what role its browser will take, so it can inform its =
peer (using whatever signaling protocol) about the role.
>>=20
>> well - assuming you leave  a=3Dsetup as actpass (the default in the =
browser)  .
>> If you called createOffer, then you'll be DTLS client.
>> If you called createAnswer you'll be DTLS server.
>> That's assuming no weirdness in the ICE implementation. So your JS =
can _deduce_ the role but neither influence it nor _know_ it directly.
>>=20
>> However, signalling it over your high level signalling protocol is =
the wrong way to do it (IMHO). Your peer should deduce the appropriate =
DTLS role from the 'signalling'
>> in the ICE packets, by looking at the ICE CONTROLLING flag.
>=20
> In SIP/SDP O/A it is done using the setup attribute (red: high level =
signalling protocol), and JSEP is based on SDP O/A, so...
>=20
> If we do NOT want to use the setup attribute to determine the DTLS =
roles, then I think we need to have a separate discussion about that.=20

we are using the  setup attribute in a way. - actpass means we are =
saying "we don't care". That leaves it to the lower levels to sort it =
out - via the iceControlling flag.

So now you see what caused the rant :-)

T.=


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] How to determine TLS roles?
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] How to determine TLS roles?
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Hi,

>>> However, signalling it over your high level signalling protocol is the =
wrong way to do it (IMHO). Your peer should=20
>>> deduce the appropriate DTLS role from the 'signalling' in the ICE packe=
ts, by looking at the ICE CONTROLLING flag.
>>
>> In SIP/SDP O/A it is done using the setup attribute (red: high level sig=
nalling protocol), and JSEP is based on SDP O/A, so...
>>
>> If we do NOT want to use the setup attribute to determine the DTLS roles=
, then I think we need to have a separate discussion about that.
>
> we are using the  setup attribute in a way. - actpass means we are saying=
 "we don't care".=20

That is NOT what it means in SDP O/A - it means "the other side decides" :)

> That leaves it to the lower levels to sort it out - via the iceControllin=
g flag.
>
> So now you see what caused the rant :-)

One way would be to add some flag/parameter, to explicitly indicate the usa=
ge of the iceControlling flag to determine the roles - if people want to do=
 that. The advantage is that it would be independent of the signalling prot=
ocol used between the peers.

Regards,

Chrsiter


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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Stefan,

<snip> > "Note that ICE support being mandatory does not preclude a =
WebRTC
> endpoint from supporting more traversal mechanisms than ICE using STUN
> and TURN."
>=20
> OK?
</snip>

OK for me.

Thanks
Partha

> -----Original Message-----
> From: Stefan H=E5kansson LK [mailto:stefan.lk.hakansson@ericsson.com]
> Sent: Tuesday, February 11, 2014 3:46 PM
> To: Magnus Westerlund; Parthasarathi R; rtcweb@ietf.org
> Subject: Re: [rtcweb] New version of use-case draft uploaded
>=20
> On 2014-02-10 12:02, Magnus Westerlund wrote:
> > Hi Partha,
> >
> > See inline
> >
> > On 2014-02-08 10:12, Parthasarathi R wrote:
> >> Hi Stefan,
> >>
> >> <snip>
> >>> Yes, I did that change (TURN -> ICE). My understanding was that =
ICE
> is
> >>> what is used/mandated (and it in turn makes use of STUN and TURN).
> >>>
> >> </snip>
> >>
> >> ICE is mandated in RTCWeb but I disagree to your assumption that =
ICE
> in turn
> >> makes use of TURN. My concern is that The requirement document is
> misleading
> >> about TURN requirements. ICE-TCP avoids TURN server as the middle-
> box in the
> >> network. ICE-TCP is host candidate whereas TURN is relay candidate
> and as
> >> per ICE candidate selection, host candidate is preferred over relay
> >> candidates.
> >
> > I will agree that the note as currently worded is not completely
> > accurate and clear. Based on the text in the preceding section, it
> makes
> > sense to note that ICE using STUN and TURN is not the only solution
> > including additional ICE modes or relay services. I request the
> authors
> > propose a new formulation of the note.
>=20
> Proposal:
>=20
> "Note that ICE support being mandatory does not preclude a WebRTC
> endpoint from supporting more traversal mechanisms than ICE using STUN
> and TURN."
>=20
> OK?
>=20
> =3D


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, =?iso-8859-1?Q?'Stefan_H=E5kansson_LK'?= <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] New version of use-case draft uploaded
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Hi Magnus,

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, February 11, 2014 6:17 PM
> To: Parthasarathi R; 'Stefan H=E5kansson LK'; rtcweb@ietf.org
> Subject: Re: [rtcweb] New version of use-case draft uploaded
>=20
> On 2014-02-10 20:09, Parthasarathi R wrote:
> > Hi Magnus,
> >
> > Thanks for proposing a new formulation of the note by the authors. I
> agree
> > with you that F2 and F29 are generic WebRTC NAT/Firewall =
requirements
> which
> > does not restrict ICE-TCP as one of the solution.
> >
> > The confusion with F31 and F32 requirements is because of the =
missing
> of
> > WebRTC server/gateway requirements. F31 and F32 leads to the
> assumption of
> > mandating TURN servers in the WebRTC server/gateway (by some of the
> WebRTC
> > gateway Providers) as there is no text in this usecase & requirement
> > document to clarify.
>=20
> Sorry, I fail to understand which of multiple possible angles you are
> arguing here.
>=20
> A) That the use case document is missing a whole requirement related =
to
> WebRTC Servers. WebRTC server, is something I am uncertain to what you
> refer to, a Web server or a media plan related server dealing with =
peer
> connections?
>=20
<Partha> WebRTC server/gateway is a server which is a combination of Web
Server which handles WebRTC signaling and Media Plane server which deals
with peer-connection. Sec 3.4 of
draft-ietf-rtcweb-use-cases-and-requirements-13 describes 3 usecases for =
the
same. Unfortunately, draft-ietf-rtcweb-use-cases-and-requirements-13 is
missing NAT/Firewall traversal requirement related to WebRTC =
server/gateway.
</Partha>

> B) That F31 and F32 would lead to an assumption of mandating =
something,
> which currently is the only solution described?
>=20
<Partha>=20
IMO, WebRTC server/gateway does not require TURN server and ICE-TCP =
serve
the purpose (Maximum of single WebRTC endpoint behind the firewall =
scenario
in a two party WebRTC session). As F31 and F32 mandates for TURN and =
there
is no NAT/Firewall explicit requirement for WebRTC server/gateway, it =
leads
to the conclusion of TURN server as an only solution by some of WebRTC
gateway providers. Hope this clarifies.
</Partha>

> C) That there needs to be text describing some aspect of the global
> presence?
>=20
<Partha> It is good to have the text which describe NAT/firewall =
traversal
for WebRTC server/gatway perspective as well. I could not understand why =
you
are not allowing the changing F31 and F32 or adding the text about =
WebRTC
server NAT/firewall traversal in sec 3.4. Could you please clarify.
</Partha>

>=20
>  I have already wrote in the separate mail thread
> > (http://www.ietf.org/mail-archive/web/rtcweb/current/msg11338.html)
> to
> > provide WebRTC server/gateway guidelines document for more clarity.
>=20
> Good, can you leave with the text and Stefan's text proposal for the
> note?
>=20
<Partha> I agree for Stefan's proposal in the separate mail thread as it =
is
related to WebRTC endpoints requirement. </Partha>

> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Tim Panton'" <tim@phonefromhere.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] How to determine TLS roles?
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Hi Christer,

I agree with you that SDP setup attribute has to be used to determine the
TLS role as it is not required that the remote side is also WebRTC endpoint.

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: Tuesday, February 11, 2014 9:26 PM
> To: Tim Panton
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] How to determine TLS roles?
> 
> 
> Hi,
> 
> >>> However, signalling it over your high level signalling protocol is
> the wrong way to do it (IMHO). Your peer should
> >>> deduce the appropriate DTLS role from the 'signalling' in the ICE
> packets, by looking at the ICE CONTROLLING flag.
> >>
> >> In SIP/SDP O/A it is done using the setup attribute (red: high level
> signalling protocol), and JSEP is based on SDP O/A, so...
> >>
> >> If we do NOT want to use the setup attribute to determine the DTLS
> roles, then I think we need to have a separate discussion about that.
> >
> > we are using the  setup attribute in a way. - actpass means we are
> saying "we don't care".
> 
> That is NOT what it means in SDP O/A - it means "the other side
> decides" :)
> 
> > That leaves it to the lower levels to sort it out - via the
> iceControlling flag.
> >
> > So now you see what caused the rant :-)
> 
> One way would be to add some flag/parameter, to explicitly indicate the
> usage of the iceControlling flag to determine the roles - if people
> want to do that. The advantage is that it would be independent of the
> signalling protocol used between the peers.
> 
> Regards,
> 
> Chrsiter
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: RTCWEB Data Channel Protocol: Label vs Protocol glare
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Subject: [rtcweb] RTCWEB Data Channel Protocol: Label vs Protocol glare
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Hi,

The draft says:

"Note: There is no attempt to resolve label glare; if both sides open
                a Channel labeled "x" at the same time, there will be two C=
hannels
                labeled "x" - one on an even Stream pair, one on an odd pai=
r."

I think the text shall be extended to also cover the Protocol, i.e. both en=
ds can try to open a channel for protocol X at the same time.

Also, even if there is a label glare, it doesn't have to be for the same pr=
otocol. As far as I know, endpoints can choose whatever label value they wa=
nt, and they don't even need to be for the same protcol (unless specific pr=
otocols define what label value(s) must be used for channels associated wit=
h that protocol).

Regards,

Christer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:.5in">&#8220;Note: There is no =
attempt to resolve label glare; if both sides open<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a Channel labeled &quot;x&quot; at the sa=
me time, there will be two Channels<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; labeled &quot;x&quot; - one on an even St=
ream pair, one on an odd pair.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I think the text shall be extended to also cover the=
 Protocol, i.e. both ends can try to open a channel for protocol X at the s=
ame time.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Also, even if there is a label glare, it doesn&#8217=
;t have to be for the same protocol. As far as I know, endpoints can choose=
 whatever label value they want, and they don&#8217;t even need to be for t=
he same protcol (unless specific protocols define
 what label value(s) must be used for channels associated with that protoco=
l).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

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From: Peter Thatcher <pthatcher@google.com>
Date: Tue, 11 Feb 2014 10:06:05 -0800
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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--e89a8ffba0e3bd8c1704f2255158
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I believe the API already gives you all the tools you need for this use
case.  Here's one way to create a data channel where both ends call
createDataChannel() at the same time, with no "middle":

1.  Both sides call createDataChannel() with the same label.
2.  If an endpoint ends up with two data channels with the same label,
close the one with the higher .id, and keep the one with the lower .id.

Now both sides will use the same channel, and you can pretend like the
other never existed.   If you're worried about data sent before this
process completes, either accept data on both channels for a while, or
don't send data until the process completes.

Is there any reason this would not work?


On Mon, Feb 10, 2014 at 5:10 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote=
:

> You two guys are talking past one another.
>
> Christer's point is that the *application" distributed between the two
> endpoints, needs a *single* instance of a channel for a particular
> application specific purpose. But the two ends are generally symmetric.
> There is no obvious choice of which side should create the channel. So ho=
w
> do we use the mechanism to achieve this result?
>
> I think I can predict what Michael is going to say: it isn't a protocol
> issue - the two ends should negotiate this some other way. (In the normal
> rtcweb scenario, a *single* app in the middle can arrange for one side to
> do the open.)
>
> This is a problem with a sip app - there is no middle, just two ends.
>
>
> Previously Christer said:
>
>  Couldn=E2=80=99t we e.g. say that, by default, the DTLS client is respon=
sible for
>> sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it
>> could even use both an even or odd stream id value.
>>
>
> That is a bad thing for data channel in general. But I suppose the app ca=
n
> choose to follow this rule if it wants for particular channels.
>
>         Thanks,
>         Paul
>
>
>
>
>
>
> On 2/10/14 12:29 PM, Christer Holmberg wrote:
>
>>
>> Hi,
>>
>>  As is defined in the data channel protocol, an entity can send data
>>>>>>>> once DATA_CHANNEL_OPEN has been sent, before the associated
>>>>>>>> DATA_CHANNEL_ACK is received.
>>>>>>>>
>>>>>>>> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN
>>>>>>>> at the same time, using different stream id values, the result may=
 be TWO
>>>>>>>> data channels, with data sent on both.
>>>>>>>>
>>>>>>>> However, as is also indicated, the draft does not provide a
>>>>>>>> mechanism to prevent/handle such glare situation.
>>>>>>>>
>>>>>>>> I personally think there shall be a way to prevent such situation,
>>>>>>>> or otherwise I'm sure we are going to end up with interoperability=
 problems.
>>>>>>>>
>>>>>>>> Couldn't we e.g. say that, by default, the DTLS client is
>>>>>>>> responsible for sending the DATA_CHANNEL_OPEN? Then there is no ri=
sk for
>>>>>>>> glare, and it could even use both an even or odd stream id value.
>>>>>>>>
>>>>>>> Section 4 reads:
>>>>>>>
>>>>>>> To avoid glare in opening Channels, each side MUST use either even
>>>>>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The meth=
od
>>>>>>> used to determine which side uses odd or even is based on the
>>>>>>> underlying DTLS connection role when used in WebRTC, with the side
>>>>>>> acting as the DTLS client using even stream identifiers.
>>>>>>>
>>>>>>> So there is no problem in setting up data channels.
>>>>>>>
>>>>>>> This should not be confused by both sides setting up a data channel
>>>>>>> with the same Label. This ends up in two data channels with the sam=
e label.
>>>>>>> You can even create more data channels with the same label.
>>>>>>>
>>>>>>
>>>>>> My point is that I don't want to end up with more than one channel.
>>>>>>
>>>>> And you won't, see above.
>>>>>
>>>>
>>>> I am not sure I understand. What text do you refer to?
>>>>
>>> To avoid glare in opening Channels, each side MUST use either even
>>> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
>>> used to determine which side uses odd or even is based on the
>>> underlying DTLS connection role when used in WebRTC, with the side
>>> acting as the DTLS client using even stream identifiers.
>>>
>>
>> That part I understand.
>>
>> But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the
>> same protocol, and THAT is what I would like to avoid. I don't want to h=
ave
>> two bi-directional data channels for the same thing.
>>
>> For that reason I suggested that we could also specify which endpoint is
>> responsisble for sending DATA_CHANNEL_OPEN.
>>
>>
>>  However, there is no mechanism to ensure uniqueness for labels. The
>>>>> WebRTC API already allows one side to create n data channels with the=
 same
>>>>> label.
>>>>>
>>>>
>>>> I am not saying it should be forbidden to do so. But, for protocol X, =
I
>>>> want to end up with ONE data channel between my peers.
>>>> Then only one side should create a data channel.
>>>>
>>>> If I end up with two data channels for protocol X, can I send data on
>>>> whichever I want to?
>>>>
>>> Sure. These would be two independent bidirectional data channels.
>>>
>>
>> Possible for the same protocol - and that is what I would want to avoid.
>> I want to send my protocol X messages on one data channel, and I would l=
ike
>> to receive the protocol X messages from my peer on the same data channel=
.
>>
>> Now, one can of course solve this by reseting one of the data channels,
>> but then the question is which endpoint should reset :)
>>
>> Regards,
>>
>> Christer
>>
>> _______________________________________________
>> 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
>

--e89a8ffba0e3bd8c1704f2255158
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:verdana,=
sans-serif">I believe the API already gives you all the tools you need for =
this use case. =C2=A0Here&#39;s one way to create a data channel where both=
 ends call createDataChannel() at the same time, with no &quot;middle&quot;=
:</div>

<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif"><br><=
/div><div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif">=
1. =C2=A0Both sides call createDataChannel() with the same label.<br></div>=
<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif">

2. =C2=A0If an endpoint ends up with two data channels with the same label,=
 close the one with the higher .id, and keep the one with the lower .id.</d=
iv><div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif"><b=
r></div>

<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif">Now b=
oth sides will use the same channel, and you can pretend like the other nev=
er existed. =C2=A0 If you&#39;re worried about data sent before this proces=
s completes, either accept data on both channels for a while, or don&#39;t =
send data until the process completes.</div>

<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif"><br><=
/div><div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif">=
Is there any reason this would not work?</div><div class=3D"gmail_extra"><b=
r><br>

<div class=3D"gmail_quote">On Mon, Feb 10, 2014 at 5:10 PM, Paul Kyzivat <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_bla=
nk">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">

You two guys are talking past one another.<br>
<br>
Christer&#39;s point is that the *application&quot; distributed between the=
 two endpoints, needs a *single* instance of a channel for a particular app=
lication specific purpose. But the two ends are generally symmetric. There =
is no obvious choice of which side should create the channel. So how do we =
use the mechanism to achieve this result?<br>


<br>
I think I can predict what Michael is going to say: it isn&#39;t a protocol=
 issue - the two ends should negotiate this some other way. (In the normal =
rtcweb scenario, a *single* app in the middle can arrange for one side to d=
o the open.)<br>


<br>
This is a problem with a sip app - there is no middle, just two ends.<div c=
lass=3D""><br>
<br>
Previously Christer said:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Couldn=E2=80=99t we e.g. say that, by default, the DTLS client is responsib=
le for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and =
it could even use both an even or odd stream id value.<br>
</blockquote>
<br></div>
That is a bad thing for data channel in general. But I suppose the app can =
choose to follow this rule if it wants for particular channels.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<div class=3D"HOEnZb"><div class=3D"h5"><br=
>
<br>
<br>
<br>
<br>
<br>
On 2/10/14 12:29 PM, Christer Holmberg wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Hi,<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

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


As is defined in the data channel protocol, an entity can send data once DA=
TA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is re=
ceived.<br>
<br>
The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at the s=
ame time, using different stream id values, the result may be TWO data chan=
nels, with data sent on both.<br>
<br>
However, as is also indicated, the draft does not provide a mechanism to pr=
event/handle such glare situation.<br>
<br>
I personally think there shall be a way to prevent such situation, or other=
wise I&#39;m sure we are going to end up with interoperability problems.<br=
>
<br>
Couldn&#39;t we e.g. say that, by default, the DTLS client is responsible f=
or sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it c=
ould even use both an even or odd stream id value.<br>
</blockquote>
Section 4 reads:<br>
<br>
To avoid glare in opening Channels, each side MUST use either even<br>
or =C2=A0odd Streams when sending a DATA_CHANNEL_OPEN message. =C2=A0The me=
thod<br>
used to determine which side uses odd or even is based on the<br>
underlying DTLS connection role when used in WebRTC, with the side<br>
acting as the DTLS client using even stream identifiers.<br>
<br>
So there is no problem in setting up data channels.<br>
<br>
This should not be confused by both sides setting up a data channel with th=
e same Label. This ends up in two data channels with the same label.<br>
You can even create more data channels with the same label.<br>
</blockquote>
<br>
My point is that I don&#39;t want to end up with more than one channel.<br>
</blockquote>
And you won&#39;t, see above.<br>
</blockquote>
<br>
I am not sure I understand. What text do you refer to?<br>
</blockquote>
To avoid glare in opening Channels, each side MUST use either even<br>
or =C2=A0odd Streams when sending a DATA_CHANNEL_OPEN message. =C2=A0The me=
thod<br>
used to determine which side uses odd or even is based on the<br>
underlying DTLS connection role when used in WebRTC, with the side<br>
acting as the DTLS client using even stream identifiers.<br>
</blockquote>
<br>
That part I understand.<br>
<br>
But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the sam=
e protocol, and THAT is what I would like to avoid. I don&#39;t want to hav=
e two bi-directional data channels for the same thing.<br>
<br>
For that reason I suggested that we could also specify which endpoint is re=
sponsisble for sending DATA_CHANNEL_OPEN.<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">


However, there is no mechanism to ensure uniqueness for labels. The WebRTC =
API already allows one side to create n data channels with the same label.<=
br>
</blockquote>
<br>
I am not saying it should be forbidden to do so. But, for protocol X, I wan=
t to end up with ONE data channel between my peers.<br>
Then only one side should create a data channel.<br>
<br>
If I end up with two data channels for protocol X, can I send data on which=
ever I want to?<br>
</blockquote>
Sure. These would be two independent bidirectional data channels.<br>
</blockquote>
<br>
Possible for the same protocol - and that is what I would want to avoid. I =
want to send my protocol X messages on one data channel, and I would like t=
o receive the protocol X messages from my peer on the same data channel.<br=
>


<br>
Now, one can of course solve this by reseting one of the data channels, but=
 then the question is which endpoint should reset :)<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
______________________________<u></u>_________________<br>
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--e89a8ffba0e3bd8c1704f2255158--


From christer.holmberg@ericsson.com  Tue Feb 11 10:12:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Some editorial comments on section 5
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Date: Tue, 11 Feb 2014 18:12:39 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Some editorial comments on section 5
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Hi,

Some editorial comments:

Q1:

Section 5 says:

	"SCTP as specified in [RFC4960] MUST be used in
   	combination with the extension defined in [RFC3758] and provides the
   	following interesting features for transporting non-media data
   	between browsers:"

I suggest to say:

	"The SCTP extension defined in [RFC3758] MUST be supported, and
     	provides the following features:


Q2:

Section 5 says:

	"Each SCTP user message contains a so called Payload Protocol
   	Identifier (PPID) that is passed..."

I suggest to remove "so called".


Q3:

Section 5 says:

	"This value can be used to multiplex multiple protocols
   	over a single SCTP association.  The sender provides for each
   	protocol a specific PPID and the receiver can demultiplex the
   	messages based on the received PPID."

I think we discussed this earlier. The PPID value CAN NOT (at least not as =
used by WebRTC) be used to multiplex multiple protocols over a single SCTP =
association. If you create multiple channels, for multiple protocols, the s=
ame PPID values may be used for each protocol.

 The PPID CAN, however, be used within a data channel, e.g. to demultiplex =
the DCP from channel data.


Q4:

Section 5 says:

	"When using DTLS as the lower layer, only single homed SCTP
   	associations MUST be used, since DTLS does not expose any address
   	management to its upper layer."

I suggest to change "MUST be used" to "are supported".


Q5:

Sometimes you say "SCTP" and sometimes "SCTP as defined in [RFC4960]".

I suggest to e.g. use the reference on the first occurrence, and then only =
use "SCTP".


Regards,

Christer



From christer.holmberg@ericsson.com  Tue Feb 11 10:15:36 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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Hi,

Section 6 describes the characteristics of the data channel.

However, the section name is "The Usage of SCTP in the WebRTC Context".

I think we should rename that to something like "The Usage of SCTP for the =
WebRTC Data Channel".

Because, the definition of the channel is the same no matter in which conte=
xt it is used :)

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Peter Thatcher' <pthatcher@google.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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From: Peter Thatcher <pthatcher@google.com>
Date: Tue, 11 Feb 2014 10:32:41 -0800
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On Tue, Feb 11, 2014 at 10:22 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Peter,
>
> > I believe the API already gives you all the tools you need for this use
> case.  Here's one way to create a data channel where both ends call
> createDataChannel() at the same time, with no "middle":
> >
> > 1.  Both sides call createDataChannel() with the same label.
>
> Or, with the same Protocol.
>
> It is a little unclear to me what "label" really is. The data channel
> draft says it's a "name of the channel"...
>

=E2=80=8BLabel is whatever the JS  or native app chooses it to be.=E2=80=8B=
  You can give
it whatever semantics you want.


>
> > 2.  If an endpoint ends up with two data channels with the same label,
> close the one with the higher .id, and keep the one with the lower .id.
>
> What is ".id"? Is it something API specific? Keep in mind that non-WebRTC
> clients may also use the data channel.
>
>
In=E2=80=8B
=E2=80=8B
=E2=80=8B JS, it's this:
http://dev.w3.org/2011/webrtc/editor/webrtc.html#widl-RTCDataChannel-id

=E2=80=8BOn the wire, it's the SCTP SID.=E2=80=8B
=E2=80=8B

 > Now both sides will use the same channel, and you can pretend like the
other never existed.   If you're worried about data sent before this
process completes, either accept data on both channels for a while, or
don't send data until the process completes.

> >
> > Is there any reason this would not work?
>
> Assuming it's generic, and not API specific, I guess it could work.
>
>
=E2=80=8B=E2=80=8B
=E2=80=8BYes, it's generic.  The logic is basically, "if you end up with tw=
o SIDs
with the same label, close the channel with the higher SID".  Applies to
any endpoint.=E2=80=8B



> Also, as has been indicated, specific protocols can specify rules
> regarding who opens the channel.
>
> I think it would be good to have some informative guidance text about thi=
s
> somewhere, e.g. in the Data Channel draft.
>
> Regards,
>
> Christer
>
>
>
>
>
>
> On Mon, Feb 10, 2014 at 5:10 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
> wrote:
> You two guys are talking past one another.
>
> Christer's point is that the *application" distributed between the two
> endpoints, needs a *single* instance of a channel for a particular
> application specific purpose. But the two ends are generally symmetric.
> There is no obvious choice of which side should create the channel. So ho=
w
> do we use the mechanism to achieve this result?
>
> I think I can predict what Michael is going to say: it isn't a protocol
> issue - the two ends should negotiate this some other way. (In the normal
> rtcweb scenario, a *single* app in the middle can arrange for one side to
> do the open.)
>
> This is a problem with a sip app - there is no middle, just two ends.
>
>
> Previously Christer said:
> Couldn=E2=80=99t we e.g. say that, by default, the DTLS client is respons=
ible for
> sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it
> could even use both an even or odd stream id value.
>
> That is a bad thing for data channel in general. But I suppose the app ca=
n
> choose to follow this rule if it wants for particular channels.
>
>         Thanks,
>         Paul
>
>
>
>
>
>
> On 2/10/14 12:29 PM, Christer Holmberg wrote:
>
> Hi,
> As is defined in the data channel protocol, an entity can send data once
> DATA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK i=
s
> received.
>
> The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at the
> same time, using different stream id values, the result may be TWO data
> channels, with data sent on both.
>
> However, as is also indicated, the draft does not provide a mechanism to
> prevent/handle such glare situation.
>
> I personally think there shall be a way to prevent such situation, or
> otherwise I'm sure we are going to end up with interoperability problems.
>
> Couldn't we e.g. say that, by default, the DTLS client is responsible for
> sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it
> could even use both an even or odd stream id value.
> Section 4 reads:
>
> To avoid glare in opening Channels, each side MUST use either even
> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
> used to determine which side uses odd or even is based on the
> underlying DTLS connection role when used in WebRTC, with the side
> acting as the DTLS client using even stream identifiers.
>
> So there is no problem in setting up data channels.
>
> This should not be confused by both sides setting up a data channel with
> the same Label. This ends up in two data channels with the same label.
> You can even create more data channels with the same label.
>
> My point is that I don't want to end up with more than one channel.
> And you won't, see above.
>
> I am not sure I understand. What text do you refer to?
> To avoid glare in opening Channels, each side MUST use either even
> or  odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
> used to determine which side uses odd or even is based on the
> underlying DTLS connection role when used in WebRTC, with the side
> acting as the DTLS client using even stream identifiers.
>
> That part I understand.
>
> But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the
> same protocol, and THAT is what I would like to avoid. I don't want to ha=
ve
> two bi-directional data channels for the same thing.
>
> For that reason I suggested that we could also specify which endpoint is
> responsisble for sending DATA_CHANNEL_OPEN.
>
> However, there is no mechanism to ensure uniqueness for labels. The WebRT=
C
> API already allows one side to create n data channels with the same label=
.
>
> I am not saying it should be forbidden to do so. But, for protocol X, I
> want to end up with ONE data channel between my peers.
> Then only one side should create a data channel.
>
> If I end up with two data channels for protocol X, can I send data on
> whichever I want to?
> Sure. These would be two independent bidirectional data channels.
>
> Possible for the same protocol - and that is what I would want to avoid. =
I
> want to send my protocol X messages on one data channel, and I would like
> to receive the protocol X messages from my peer on the same data channel.
>
> Now, one can of course solve this by reseting one of the data channels,
> but then the question is which endpoint should reset :)
>
> Regards,
>
> Christer
>
> _______________________________________________
> 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
>
>

--047d7b15a811d840f204f225b0a7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:verdana,=
sans-serif"><br></div><div class=3D"gmail_extra"><br><br><div class=3D"gmai=
l_quote">On Tue, Feb 11, 2014 at 10:22 AM, Christer Holmberg <span dir=3D"l=
tr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank"=
>christer.holmberg@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">Hi Peter,<br>
<div class=3D""><br>
&gt; I believe the API already gives you all the tools you need for this us=
e case. =C2=A0Here&#39;s one way to create a data channel where both ends c=
all createDataChannel() at the same time, with no &quot;middle&quot;:<br>
&gt;<br>
&gt; 1. =C2=A0Both sides call createDataChannel() with the same label.<br>
<br>
</div>Or, with the same Protocol.<br>
<br>
It is a little unclear to me what &quot;label&quot; really is. The data cha=
nnel draft says it&#39;s a &quot;name of the channel&quot;...<br></blockquo=
te><div><br></div><div><div class=3D"gmail_default" style=3D"font-family:ve=
rdana,sans-serif">

=E2=80=8BLabel is whatever the JS =C2=A0or native app chooses it to be.=E2=
=80=8B =C2=A0You can give it whatever semantics you want.</div></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""><br>
&gt; 2. =C2=A0If an endpoint ends up with two data channels with the same l=
abel, close the one with the higher .id, and keep the one with the lower .i=
d.<br>
<br>
</div>What is &quot;.id&quot;? Is it something API specific? Keep in mind t=
hat non-WebRTC clients may also use the data channel.<br>
<div class=3D""><br></div></blockquote><div><div class=3D"gmail_default" st=
yle=3D"font-family:verdana,sans-serif;display:inline"><br></div></div><div>=
<div class=3D"gmail_default" style=3D"font-family:verdana,sans-serif;displa=
y:inline">

In=E2=80=8B</div><span style=3D"font-family:verdana,sans-serif">=E2=80=8B<d=
iv class=3D"gmail_default" style=3D"font-family:verdana,sans-serif;display:=
inline">=E2=80=8B JS, it&#39;s this:=C2=A0</div></span><font face=3D"verdan=
a, sans-serif"><a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html#=
widl-RTCDataChannel-id">http://dev.w3.org/2011/webrtc/editor/webrtc.html#wi=
dl-RTCDataChannel-id</a></font></div>

<div><span style=3D"font-family:verdana,sans-serif"><br></span></div><div><=
span style=3D"font-family:verdana,sans-serif"><div class=3D"gmail_default" =
style=3D"font-family:verdana,sans-serif;display:inline">=E2=80=8BOn the wir=
e, it&#39;s the SCTP SID.=E2=80=8B</div>

=E2=80=8B</span></div><div><br></div><div>=C2=A0&gt; Now both sides will us=
e the same channel, and you can pretend like the other never existed. =C2=
=A0 If you&#39;re worried about data sent before this process completes, ei=
ther accept data on both channels for a while, or don&#39;t send data until=
 the process completes.</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"">
&gt;<br>
&gt; Is there any reason this would not work?<br>
<br>
</div>Assuming it&#39;s generic, and not API specific, I guess it could wor=
k.<br>
<br>
</blockquote><div><br></div><div><div class=3D"gmail_default" style=3D"font=
-family:verdana,sans-serif">=E2=80=8B=E2=80=8B</div><div class=3D"gmail_def=
ault" style=3D"font-family:verdana,sans-serif">=E2=80=8BYes, it&#39;s gener=
ic. =C2=A0The logic is basically, &quot;if you end up with two SIDs with th=
e same label, close the channel with the higher SID&quot;. =C2=A0Applies to=
 any endpoint.=E2=80=8B</div>

<br></div><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">Also, as has been indicated, sp=
ecific protocols can specify rules regarding who opens the channel.<br>


<br>
I think it would be good to have some informative guidance text about this =
somewhere, e.g. in the Data Channel draft.<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">Regards,<br>
<br>
Christer<br>
<div class=3D""><div class=3D"h5"><br>
<br>
<br>
<br>
<br>
<br>
On Mon, Feb 10, 2014 at 5:10 PM, Paul Kyzivat &lt;<a href=3D"mailto:pkyziva=
t@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
You two guys are talking past one another.<br>
<br>
Christer&#39;s point is that the *application&quot; distributed between the=
 two endpoints, needs a *single* instance of a channel for a particular app=
lication specific purpose. But the two ends are generally symmetric. There =
is no obvious choice of which side should create the channel. So how do we =
use the mechanism to achieve this result?<br>


<br>
I think I can predict what Michael is going to say: it isn&#39;t a protocol=
 issue - the two ends should negotiate this some other way. (In the normal =
rtcweb scenario, a *single* app in the middle can arrange for one side to d=
o the open.)<br>


<br>
This is a problem with a sip app - there is no middle, just two ends.<br>
<br>
<br>
Previously Christer said:<br>
Couldn=E2=80=99t we e.g. say that, by default, the DTLS client is responsib=
le for sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and =
it could even use both an even or odd stream id value.<br>
<br>
That is a bad thing for data channel in general. But I suppose the app can =
choose to follow this rule if it wants for particular channels.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<br>
<br>
<br>
<br>
<br>
<br>
<br>
On 2/10/14 12:29 PM, Christer Holmberg wrote:<br>
<br>
Hi,<br>
As is defined in the data channel protocol, an entity can send data once DA=
TA_CHANNEL_OPEN has been sent, before the associated DATA_CHANNEL_ACK is re=
ceived.<br>
<br>
The draft also says that, if both endpoints send DATA_CHANNEL_OPEN at the s=
ame time, using different stream id values, the result may be TWO data chan=
nels, with data sent on both.<br>
<br>
However, as is also indicated, the draft does not provide a mechanism to pr=
event/handle such glare situation.<br>
<br>
I personally think there shall be a way to prevent such situation, or other=
wise I&#39;m sure we are going to end up with interoperability problems.<br=
>
<br>
Couldn&#39;t we e.g. say that, by default, the DTLS client is responsible f=
or sending the DATA_CHANNEL_OPEN? Then there is no risk for glare, and it c=
ould even use both an even or odd stream id value.<br>
Section 4 reads:<br>
<br>
To avoid glare in opening Channels, each side MUST use either even<br>
or =C2=A0odd Streams when sending a DATA_CHANNEL_OPEN message. =C2=A0The me=
thod<br>
used to determine which side uses odd or even is based on the<br>
underlying DTLS connection role when used in WebRTC, with the side<br>
acting as the DTLS client using even stream identifiers.<br>
<br>
So there is no problem in setting up data channels.<br>
<br>
This should not be confused by both sides setting up a data channel with th=
e same Label. This ends up in two data channels with the same label.<br>
You can even create more data channels with the same label.<br>
<br>
My point is that I don&#39;t want to end up with more than one channel.<br>
And you won&#39;t, see above.<br>
<br>
I am not sure I understand. What text do you refer to?<br>
To avoid glare in opening Channels, each side MUST use either even<br>
or =C2=A0odd Streams when sending a DATA_CHANNEL_OPEN message. =C2=A0The me=
thod<br>
used to determine which side uses odd or even is based on the<br>
underlying DTLS connection role when used in WebRTC, with the side<br>
acting as the DTLS client using even stream identifiers.<br>
<br>
That part I understand.<br>
<br>
But, this means that both endpoints may send DATA_CHANNEL_OPEN, for the sam=
e protocol, and THAT is what I would like to avoid. I don&#39;t want to hav=
e two bi-directional data channels for the same thing.<br>
<br>
For that reason I suggested that we could also specify which endpoint is re=
sponsisble for sending DATA_CHANNEL_OPEN.<br>
<br>
However, there is no mechanism to ensure uniqueness for labels. The WebRTC =
API already allows one side to create n data channels with the same label.<=
br>
<br>
I am not saying it should be forbidden to do so. But, for protocol X, I wan=
t to end up with ONE data channel between my peers.<br>
Then only one side should create a data channel.<br>
<br>
If I end up with two data channels for protocol X, can I send data on which=
ever I want to?<br>
Sure. These would be two independent bidirectional data channels.<br>
<br>
Possible for the same protocol - and that is what I would want to avoid. I =
want to send my protocol X messages on one data channel, and I would like t=
o receive the protocol X messages from my peer on the same data channel.<br=
>


<br>
Now, one can of course solve this by reseting one of the data channels, but=
 then the question is which endpoint should reset :)<br>
<br>
Regards,<br>
<br>
Christer<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>
_______________________________________________<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>
</div></div></blockquote></div><br></div></div>

--047d7b15a811d840f204f225b0a7--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Peter Thatcher' <pthatcher@google.com>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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From richard.ejzak@gmail.com  Tue Feb 11 12:05:11 2014
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From: Richard Ejzak <richard.ejzak@gmail.com>
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Subject: Re: [rtcweb] Data channel: JavaScript usage of data channel without the Data Channel Protocol?
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Hi Christer,
I didn't see any other response to your question, but please excuse me if I
missed it.

My understanding of this case is that if DCEP is not used, the "open" event
is triggered once the browser finishes creating the protocol machinery for
the data channel being created.  This is the only mechanism available to
acknowledge to the JS app that the data channel has been set up since, as
you say, the browser will neither send nor receive an open/ack message.  I
don't see any other reasonable alternative to this without API changes.

There is a slight problem with this since the peer may not have established
the protocol machinery necessary to actually communicate on the created
data channel.  Some external mechanism is necessary to verify that both
peers have "opened" the same data channel before either party sends data.
 See my draft on SDP for external negotiation of data channels for a
description of one way to do this.

Richard


On Thu, Feb 6, 2014 at 5:15 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> As was indicated earlier, the usage of the Data Channel Protocol (DCP) is
> optional with the Data Channel.
>
>
>
> However, if I understand the API correctly, only the DCP "open"/"ack"
> messages will trigger a JavaScript "open" event.
>
>
>
> If so, how will my JS App be able to use the data channel if I do NOT use
> the Data Channel Protocol? Can I use it without the "open" event, and/or is
> there something else that will trigger it?
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Hi Christer,<div>I didn&#39;t see any other response to yo=
ur question, but please excuse me if I missed it.</div><div><br></div><div>=
My understanding of this case is that if DCEP is not used, the &quot;open&q=
uot; event is triggered once the browser finishes creating the protocol mac=
hinery for the data channel being created. &nbsp;This is the only mechanism=
 available to acknowledge to the JS app that the data channel has been set =
up since, as you say, the browser will neither send nor receive an open/ack=
 message. &nbsp;I don&#39;t see any other reasonable alternative to this wi=
thout API changes.</div>
<div><br></div><div>There is a slight problem with this since the peer may =
not have established the protocol machinery necessary to actually communica=
te on the created data channel. &nbsp;Some external mechanism is necessary =
to verify that both peers have &quot;opened&quot; the same data channel bef=
ore either party sends data. &nbsp;See my draft on SDP for external negotia=
tion of data channels for a description of one way to do this.</div>
<div><br></div><div>Richard</div></div><div class=3D"gmail_extra"><br><br><=
div class=3D"gmail_quote">On Thu, Feb 6, 2014 at 5:15 AM, Christer Holmberg=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" ta=
rget=3D"_blank">christer.holmberg@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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">As was indicated earlier, the usage of the Data Chan=
nel Protocol (DCP) is optional with the Data Channel.
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">However, if I understand the API correctly, only the=
 DCP &quot;open&quot;/&quot;ack&quot; messages will trigger a JavaScript &q=
uot;open&quot; event.
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">If so, how will my JS App be able to use the data ch=
annel if I do NOT use the Data Channel Protocol? Can I use it without the &=
ldquo;open&rdquo; event, and/or is there something else that will trigger i=
t?<u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
</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>

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On 2/10/2014 10:20 AM, Peter Thatcher wrote:
> If you have to wait for the ACK, it's one more RTT you have to wait 
> before you can send data.  We've already got many RTTs form ICE and 
> DTLS and what's built into SCTP.  We originally didn't have an ACK 
> message at all, in large part because we didn't want more RTTs.  The 
> ACK was put in to handle the edge case of unordered data arriving 
> before the OPEN message.  But we added it in a way that it doesn't add 
> more RTTs.

Exactly.   Especially at call-start.

>
>
> On Mon, Feb 10, 2014 at 2:20 AM, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Hi,
>
>     As is defined in the data channel protocol, an entity can send
>     data once DATA_CHANNEL_OPEN has been sent, before the associated
>     DATA_CHANNEL_ACK is received.
>
>     What is the reason for allowing data to be sent before
>     DATA_CHANNEL_ACK? The sender may not even know (depends on
>     whatever external negotiation mechanisms are used) whether the
>     remote peer supports the protocol to begin with.
>
>     It think it would be good to allow the remote peer to accept
>     (DATA_CHANNEL_ACK) or reject (stream reset) the DATA_CHANNEL_OPEN
>     before data is sent.
>

"Rejection" of a channel can be done be either ignoring it (drop the 
channel reference on the floor), or actively closing it 
(channel.close(), then drop it on the floor).  This was agreed on a long 
time ago.

-- 
Randell Jesup
randell-ietf@jesup.org


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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 2/10/2014 10:20 AM, Peter Thatcher
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrXDUHBBt6L7tp4Ck426xQEkAQEmOnscmu2mBFUGR8YVHqoaw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:verdana,sans-serif">If you have to wait for
          the ACK, it's one more RTT you have to wait before you can
          send data.  We've already got many RTTs form ICE and DTLS and
          what's built into SCTP.  We originally didn't have an ACK
          message at all, in large part because we didn't want more
          RTTs.  The ACK was put in to handle the edge case of unordered
          data arriving before the OPEN message.  But we added it in a
          way that it doesn't add more RTTs.</div>
      </div>
    </blockquote>
    <br>
    Exactly.   Especially at call-start.<br>
    <br>
    <blockquote
cite="mid:CAJrXDUHBBt6L7tp4Ck426xQEkAQEmOnscmu2mBFUGR8YVHqoaw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Mon, Feb 10, 2014 at 2:20 AM,
          Christer Holmberg <span dir="ltr">&lt;<a
              moz-do-not-send="true"
              href="mailto:christer.holmberg@ericsson.com"
              target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
            <br>
            As is defined in the data channel protocol, an entity can
            send data once DATA_CHANNEL_OPEN has been sent, before the
            associated DATA_CHANNEL_ACK is received.<br>
            <br>
            What is the reason for allowing data to be sent before
            DATA_CHANNEL_ACK? The sender may not even know (depends on
            whatever external negotiation mechanisms are used) whether
            the remote peer supports the protocol to begin with.<br>
            <br>
            It think it would be good to allow the remote peer to accept
            (DATA_CHANNEL_ACK) or reject (stream reset) the
            DATA_CHANNEL_OPEN before data is sent.<br>
          </blockquote>
        </div>
      </div>
    </blockquote>
    <br>
    "Rejection" of a channel can be done be either ignoring it (drop the
    channel reference on the floor), or actively closing it
    (channel.close(), then drop it on the floor).  This was agreed on a
    long time 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>
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On 2/11/2014 5:47 AM, Michael Tuexen wrote:
> On Nov 11, 2013, at 2:54 PM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
>
>
>> 3. Section 3.2:
>>    U-C 7:  Proxy browsing, where a browser uses data channels of a
>>       PeerConnection to send and receive HTTP/HTTPS requests and data,
>>       for example to avoid local internet filtering or monitoring.
>>
>> Yes, this may be something that is possible, but to express it as a use
>> cases intended to be supported might not be the best. We are after all
>> likely talking about circumventing local security policies. I would also
>> note that "Internet" is with capital I.
> Change to Internet. The use case seems important, since there are already
> implementations doing this (possibly not using data channels, don't know).
> Randell: What do you think?

There are people actively planning to use it for this usecase in 
high-profile applications (using datachannels), or so I am told.

>
>> 6. Section 4:
>> [ TBD: how this is encoded and what the impact of this is. ]
>>
>> Didn't we have some direction decisions on how priority was to be
>> handled at least locally by the end-point. Please add this to the draft.
> I think we decided to have priorities for data channels, which are not strict
> priorities. That is all. The W3C document doesn't mention it.

We haven't discussed it yet at the W3 level.

>> I also hope that the way of representing priorities can get clearer from
>> an API perspective.
> I agree. However, I'm not a member of W3C...

Anyone can take part in the W3 discussions on this.  Please feel free to 
propose something.

>> 12. Section 5:
>> "  Since DTLS is typically implemented in user-land, the SCTP stack also
>>    needs to be a user-land stack."
>>
>> I wonder what the message of this sentence is? The reason is that I
> That you can't use a kernel SCTP stack, even if available.
>> wonder what it should be saying if one implements DTLS in a kernel, or
> I haven't seen this yet. I think you can do the encrypt/decrypt part,
> but doing the key management stuff in kernel land doesn't seem appropriate.
>> similar if one despite a DTLS user-land stack uses a SCTP stack in the
>> kernel through access to raw sockets.
> How can the lower layer run in userland when the upper layer runs in
> kernel land?

It can in theory, but likely not in any of the OS's we're concerned with 
currently.

>> 15. Section 5:
>>    Using a congestion control
>>    different from the standard one might improve the impact on the
>>    parallel SRTP media streams.  Since SCTP does not support the
>>    negotiation of a congestion control algorithm, alternate congestion
>>    controls SHOULD only require a different sender side behavior using
>>    existing information carried in the association.
>>
>> A: I wonder if adding a "yet" into the second sentence to make clear
>> that in the future there might be support for negotiating congestion
>> control.
>>
>> B: Also, what happens if one violate the SHOULD and require receiver
>> side information and the peer don't support it. That clearly can't be
>> acceptable? Can this be formulated in some other way that is tighter
>> without preventing sender side changes?
> OK. Changed to
> Since SCTP does not support the negotiation of a congestion control algorithm
> yet, alternate congestion controls SHOULD either only require a different
> sender side behavior using existing information carried in the association or
> need also specify a negotiation of of a congestion control algorithm.</t>

Sounds good.

>> 18. Section 6.2:
>> "  Additionally,
>>    the negotiation SHOULD include some type of congestion control
>>    selection. "
>>
>> Okay, this appears very fuzzy. And what name space or specifications are
>> you going to use to negotiate what the different options are?
> Randell? Salvatore? I don't think there is anything specified yet.

There is not thus far



-- 
Randell Jesup
randell-ietf@jesup.org


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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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On 2/11/14 1:06 PM, Peter Thatcher wrote:
> I believe the API already gives you all the tools you need for this use
> case.  Here's one way to create a data channel where both ends call
> createDataChannel() at the same time, with no "middle":
>
> 1.  Both sides call createDataChannel() with the same label.
> 2.  If an endpoint ends up with two data channels with the same label,
> close the one with the higher .id, and keep the one with the lower .id.
>
> Now both sides will use the same channel, and you can pretend like the
> other never existed.   If you're worried about data sent before this
> process completes, either accept data on both channels for a while, or
> don't send data until the process completes.
>
> Is there any reason this would not work?

This seems reasonable to me.

	Thanks,
	Paul

> On Mon, Feb 10, 2014 at 5:10 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     You two guys are talking past one another.
>
>     Christer's point is that the *application" distributed between the
>     two endpoints, needs a *single* instance of a channel for a
>     particular application specific purpose. But the two ends are
>     generally symmetric. There is no obvious choice of which side should
>     create the channel. So how do we use the mechanism to achieve this
>     result?
>
>     I think I can predict what Michael is going to say: it isn't a
>     protocol issue - the two ends should negotiate this some other way.
>     (In the normal rtcweb scenario, a *single* app in the middle can
>     arrange for one side to do the open.)
>
>     This is a problem with a sip app - there is no middle, just two ends.
>
>
>     Previously Christer said:
>
>         Couldnâ€™t we e.g. say that, by default, the DTLS client is
>         responsible for sending the DATA_CHANNEL_OPEN? Then there is no
>         risk for glare, and it could even use both an even or odd stream
>         id value.
>
>
>     That is a bad thing for data channel in general. But I suppose the
>     app can choose to follow this rule if it wants for particular channels.
>
>              Thanks,
>              Paul
>
>
>
>
>
>
>     On 2/10/14 12:29 PM, Christer Holmberg wrote:
>
>
>         Hi,
>
>                                 As is defined in the data channel
>                                 protocol, an entity can send data once
>                                 DATA_CHANNEL_OPEN has been sent, before
>                                 the associated DATA_CHANNEL_ACK is received.
>
>                                 The draft also says that, if both
>                                 endpoints send DATA_CHANNEL_OPEN at the
>                                 same time, using different stream id
>                                 values, the result may be TWO data
>                                 channels, with data sent on both.
>
>                                 However, as is also indicated, the draft
>                                 does not provide a mechanism to
>                                 prevent/handle such glare situation.
>
>                                 I personally think there shall be a way
>                                 to prevent such situation, or otherwise
>                                 I'm sure we are going to end up with
>                                 interoperability problems.
>
>                                 Couldn't we e.g. say that, by default,
>                                 the DTLS client is responsible for
>                                 sending the DATA_CHANNEL_OPEN? Then
>                                 there is no risk for glare, and it could
>                                 even use both an even or odd stream id
>                                 value.
>
>                             Section 4 reads:
>
>                             To avoid glare in opening Channels, each
>                             side MUST use either even
>                             or  odd Streams when sending a
>                             DATA_CHANNEL_OPEN message.  The method
>                             used to determine which side uses odd or
>                             even is based on the
>                             underlying DTLS connection role when used in
>                             WebRTC, with the side
>                             acting as the DTLS client using even stream
>                             identifiers.
>
>                             So there is no problem in setting up data
>                             channels.
>
>                             This should not be confused by both sides
>                             setting up a data channel with the same
>                             Label. This ends up in two data channels
>                             with the same label.
>                             You can even create more data channels with
>                             the same label.
>
>
>                         My point is that I don't want to end up with
>                         more than one channel.
>
>                     And you won't, see above.
>
>
>                 I am not sure I understand. What text do you refer to?
>
>             To avoid glare in opening Channels, each side MUST use
>             either even
>             or  odd Streams when sending a DATA_CHANNEL_OPEN message.
>               The method
>             used to determine which side uses odd or even is based on the
>             underlying DTLS connection role when used in WebRTC, with
>             the side
>             acting as the DTLS client using even stream identifiers.
>
>
>         That part I understand.
>
>         But, this means that both endpoints may send DATA_CHANNEL_OPEN,
>         for the same protocol, and THAT is what I would like to avoid. I
>         don't want to have two bi-directional data channels for the same
>         thing.
>
>         For that reason I suggested that we could also specify which
>         endpoint is responsisble for sending DATA_CHANNEL_OPEN.
>
>
>                     However, there is no mechanism to ensure uniqueness
>                     for labels. The WebRTC API already allows one side
>                     to create n data channels with the same label.
>
>
>                 I am not saying it should be forbidden to do so. But,
>                 for protocol X, I want to end up with ONE data channel
>                 between my peers.
>                 Then only one side should create a data channel.
>
>                 If I end up with two data channels for protocol X, can I
>                 send data on whichever I want to?
>
>             Sure. These would be two independent bidirectional data
>             channels.
>
>
>         Possible for the same protocol - and that is what I would want
>         to avoid. I want to send my protocol X messages on one data
>         channel, and I would like to receive the protocol X messages
>         from my peer on the same data channel.
>
>         Now, one can of course solve this by reseting one of the data
>         channels, but then the question is which endpoint should reset :)
>
>         Regards,
>
>         Christer
>
>         _________________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/__listinfo/rtcweb
>         <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>
>     _________________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/rtcweb
>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>


From pkyzivat@alum.mit.edu  Tue Feb 11 19:56:12 2014
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Subject: Re: [rtcweb] RTCWEB Data Channel Protocol: Label vs Protocol glare
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On 2/11/14 12:50 PM, Christer Holmberg wrote:
> Hi,
>
> The draft says:
>
> “Note: There is no attempt to resolve label glare; if both sides open
>
>                  a Channel labeled "x" at the same time, there will be
> two Channels
>
>                  labeled "x" - one on an even Stream pair, one on an odd
> pair.”
>
> I think the text shall be extended to also cover the Protocol, i.e. both
> ends can try to open a channel for protocol X at the same time.

And that should be fine.

Having two channels with the same protocol is analogous to having two 
audio rtp m-lines.

> Also, even if there is a label glare, it doesn’t have to be for the same
> protocol. As far as I know, endpoints can choose whatever label value
> they want, and they don’t even need to be for the same protcol (unless
> specific protocols define what label value(s) must be used for channels
> associated with that protocol).

IIUC the point is for applications to manage the label namespace as they 
wish.

	Thanks,
	Paul

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


From pkyzivat@alum.mit.edu  Tue Feb 11 20:03:56 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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I suggest that this just be called "Data Channel", not "WebRTC Data 
Channel". (Or pick some other neutral name - perhaps SCTP Data Channel.)

This won't be used solely for WebRTC. (It seems weird for two sip-based 
implementations of CLUE to be using a *WebRTC* Data Channel to talk to 
one another.)

(But it would still make sense for the JS API to have a WebRTC-specific 
name.)

	Thanks,
	Paul

On 2/11/14 1:15 PM, Christer Holmberg wrote:
> Hi,
>
> Section 6 describes the characteristics of the data channel.
>
> However, the section name is "The Usage of SCTP in the WebRTC Context".
>
> I think we should rename that to something like "The Usage of SCTP for the WebRTC Data Channel".
>
> Because, the definition of the channel is the same no matter in which context it is used :)
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: Re: [rtcweb] New version of use-case draft uploaded
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See inline



On 2014-02-11 18:18, Parthasarathi R wrote:
>>
>> Sorry, I fail to understand which of multiple possible angles you are
>> arguing here.
>>
>> A) That the use case document is missing a whole requirement related to
>> WebRTC Servers. WebRTC server, is something I am uncertain to what you
>> refer to, a Web server or a media plan related server dealing with peer
>> connections?
>>
> <Partha> WebRTC server/gateway is a server which is a combination of Web
> Server which handles WebRTC signaling and Media Plane server which deals
> with peer-connection. Sec 3.4 of
> draft-ietf-rtcweb-use-cases-and-requirements-13 describes 3 usecases for the
> same. Unfortunately, draft-ietf-rtcweb-use-cases-and-requirements-13 is
> missing NAT/Firewall traversal requirement related to WebRTC server/gateway.
> </Partha>

Well, the Media plane functions are logically separate from the Web
Server providing the JS and providing the signalling protocol
functionality. Thus, I think this is unnecessary bunching together
functionality.

When it comes to NAT/FW requirements we have them on the whole solution.
Why aren't these sufficient?

> 
>> B) That F31 and F32 would lead to an assumption of mandating something,
>> which currently is the only solution described?
>>
> <Partha> 
> IMO, WebRTC server/gateway does not require TURN server and ICE-TCP serve
> the purpose (Maximum of single WebRTC endpoint behind the firewall scenario
> in a two party WebRTC session). As F31 and F32 mandates for TURN and there
> is no NAT/Firewall explicit requirement for WebRTC server/gateway, it leads
> to the conclusion of TURN server as an only solution by some of WebRTC
> gateway providers. Hope this clarifies.
> </Partha>

If we look at the media functionality side, as the Webserver NAT/FW
traversal is basic requirement to even get the signaling to establish a
peer connection working. Then we have general requirements. We have
requirements that NAT/FW traversal is required. We have F31 and F32
which are requirement that was created after we taken a decision on at
least requiring ICE with STUN and TURN support for WebRTC endpoints.
Thus, F31 and F32 applies assuming that you will use these. I would
prefer a requirements formulation that isn't technology specific. But
you edits doesn't resole that concern.

> 
>> C) That there needs to be text describing some aspect of the global
>> presence?
>>
> <Partha> It is good to have the text which describe NAT/firewall traversal
> for WebRTC server/gatway perspective as well. I could not understand why you
> are not allowing the changing F31 and F32 or adding the text about WebRTC
> server NAT/firewall traversal in sec 3.4. Could you please clarify.
> </Partha>

First of all, I do believe we have requirements on the functionality you
discuss. Secondly, your proposal to change does not appear to achieve
any consensus. My personal view, which I just count as one view here is
that your proposal for change does not improve the situation, it only
confuses it more. If you had a proposal which there appear to exist
consensus around then I would request the authors to change the document.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From christer.holmberg@ericsson.com  Wed Feb 12 02:23:49 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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Hi,

> I suggest that this just be called "Data Channel", not "WebRTC Data Chann=
el". (Or pick some other neutral name - perhaps SCTP Data Channel.)

Agree. A more generic name would be useful.

"Simple Data Channel"
"Simple SCTP Data Channel"
Etc...

> This won't be used solely for WebRTC. (It seems weird for two sip-based i=
mplementations of CLUE to be using a *WebRTC* Data Channel to talk to one a=
nother.)
>
> (But it would still make sense for the JS API to have a WebRTC-specific n=
ame.)

Agree.

Regards

Christer


On 2/11/14 1:15 PM, Christer Holmberg wrote:
> Hi,
>
> Section 6 describes the characteristics of the data channel.
>
> However, the section name is "The Usage of SCTP in the WebRTC Context".
>
> I think we should rename that to something like "The Usage of SCTP for th=
e WebRTC Data Channel".
>
> Because, the definition of the channel is the same no matter in which=20
> context it is used :)
>
> Regards,
>
> Christer
>
> _______________________________________________
> 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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] Data channel: JavaScript usage of data channel without the Data Channel Protocol?
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Subject: Re: [rtcweb] Data channel: JavaScript usage of data channel without the Data Channel Protocol?
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Hi Richard,

Specific applications can decide who opens the data channel, when to start =
sending data, etc.

Regards,

Christer


From: Richard Ejzak [mailto:richard.ejzak@gmail.com]
Sent: 11. helmikuuta 2014 22:05
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Data channel: JavaScript usage of data channel withou=
t the Data Channel Protocol?

Hi Christer,
I didn't see any other response to your question, but please excuse me if I=
 missed it.

My understanding of this case is that if DCEP is not used, the "open" event=
 is triggered once the browser finishes creating the protocol machinery for=
 the data channel being created.  This is the only mechanism available to a=
cknowledge to the JS app that the data channel has been set up since, as yo=
u say, the browser will neither send nor receive an open/ack message.  I do=
n't see any other reasonable alternative to this without API changes.

There is a slight problem with this since the peer may not have established=
 the protocol machinery necessary to actually communicate on the created da=
ta channel.  Some external mechanism is necessary to verify that both peers=
 have "opened" the same data channel before either party sends data.  See m=
y draft on SDP for external negotiation of data channels for a description =
of one way to do this.

Richard

On Thu, Feb 6, 2014 at 5:15 AM, Christer Holmberg <christer.holmberg@ericss=
on.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

As was indicated earlier, the usage of the Data Channel Protocol (DCP) is o=
ptional with the Data Channel.

However, if I understand the API correctly, only the DCP "open"/"ack" messa=
ges will trigger a JavaScript "open" event.

If so, how will my JS App be able to use the data channel if I do NOT use t=
he Data Channel Protocol? Can I use it without the "open" event, and/or is =
there something else that will trigger it?

Regards,

Christer

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


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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">Hi Richard,<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>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Specific applications can=
 decide who opens the data channel, when to start sending data, etc.<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"><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">Christer<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"><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;"> Richard =
Ejzak [mailto:richard.ejzak@gmail.com]
<br>
<b>Sent:</b> 11. helmikuuta 2014 22:05<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Data channel: JavaScript usage of data channel=
 without the Data Channel Protocol?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Hi Christer,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">I didn't see any other response to your question, bu=
t please excuse me if I missed it.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">My understanding of this case is that if DCEP is not=
 used, the &quot;open&quot; event is triggered once the browser finishes cr=
eating the protocol machinery for the data channel being created. &nbsp;Thi=
s is the only mechanism available to acknowledge
 to the JS app that the data channel has been set up since, as you say, the=
 browser will neither send nor receive an open/ack message. &nbsp;I don't s=
ee any other reasonable alternative to this without API changes.<o:p></o:p>=
</p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">There is a slight problem with this since the peer m=
ay not have established the protocol machinery necessary to actually commun=
icate on the created data channel. &nbsp;Some external mechanism is necessa=
ry to verify that both peers have &quot;opened&quot;
 the same data channel before either party sends data. &nbsp;See my draft o=
n SDP for external negotiation of data channels for a description of one wa=
y to do this.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Richard<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Thu, Feb 6, 2014 at 5:15 AM, Christer Holmberg &l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">As was indicated earlier, the usage of the Data Channel Protocol (=
DCP) is optional with the Data Channel.
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">However, if I understand the API correctly, only the DCP &quot;ope=
n&quot;/&quot;ack&quot; messages will trigger a JavaScript &quot;open&quot;=
 event.
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">If so, how will my JS App be able to use the data channel if I do =
NOT use the Data Channel Protocol? Can I use it without the &#8220;open&#82=
21; event, and/or is there something else that will
 trigger it?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><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><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D16E732ESESSMB209erics_--


From christer.holmberg@ericsson.com  Wed Feb 12 02:33:01 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB Data Channel Protocol: Label vs Protocol glare
Thread-Index: Ac8nUY8MdmECwmXERwuUTRwOK9mRzAATGveAAA/PlKA=
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Subject: Re: [rtcweb] RTCWEB Data Channel Protocol: Label vs Protocol glare
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Hi,

>> The draft says:
>>
>> "Note: There is no attempt to resolve label glare; if both sides open
>>
>>                  a Channel labeled "x" at the same time, there will be=20
>> two Channels
>>
>>                  labeled "x" - one on an even Stream pair, one on an=20
>> odd pair."
>>
>> I think the text shall be extended to also cover the Protocol, i.e.=20
>> both ends can try to open a channel for protocol X at the same time.
>
> And that should be fine.
>
> Having two channels with the same protocol is analogous to having two aud=
io rtp m-lines.

...and sometimes that is what you want. But, in case you only want one chan=
nel for protocol X, the basic mechanism doesn't prevent a glare.

(Then, as has been discussed, one can do other things from preventing multi=
ple channels for the same protocol to be created.)

>> Also, even if there is a label glare, it doesn't have to be for the=20
>> same protocol. As far as I know, endpoints can choose whatever label=20
>> value they want, and they don't even need to be for the same protcol=20
>> (unless specific protocols define what label value(s) must be used for=20
>> channels associated with that protocol).
>
> IIUC the point is for applications to manage the label namespace as they =
wish.

Assume that you are using application X, and I am using application Y. Both=
 X and Y support protocol Z.

Now, unless protocol Z specifies what label value to use, X and Y may use d=
ifferent values. I don't know whether it really matters, but my point was t=
hat same label value doesn't by default mean same protocol :)

Regards,

Christer


From internet-drafts@ietf.org  Wed Feb 12 09:30:09 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-14.txt
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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           : Web Real-Time Communication Use-cases and Requirements
        Authors         : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-14.txt
	Pages           : 31
	Date            : 2014-02-12

Abstract:
   This document describes web based real-time communication use-cases.
   Requirements on the browser functionality are derived from the use-
   cases.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-use-cases-and-requirements-14


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until the htmlized version and diff are available at tools.ietf.org.

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ftp://ftp.ietf.org/internet-drafts/


From stefan.lk.hakansson@ericsson.com  Wed Feb 12 09:33:53 2014
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New for this version:

* The requirements are spelled out as they are derived in each use-case
* The requirements are put in categories
* Updated the note about that other mechanisms than ICE/STUN/TURN are allow=
ed for nat/fw traversal

I'm not sure if there is anything more that needs to be fixed.

Stefan

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of internet-drafts@=
ietf.org
Sent: den 12 februari 2014 18:30
To: i-d-announce@ietf.org
Cc: rtcweb@ietf.org
Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-=
14.txt


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 Use-cases and Require=
ments
        Authors         : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-14.txt
	Pages           : 31
	Date            : 2014-02-12

Abstract:
   This document describes web based real-time communication use-cases.
   Requirements on the browser functionality are derived from the use-
   cases.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requiremen=
ts/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-use-cases-and-requirem=
ents-14


Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at tools.ietf.org.

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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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "'Peter Thatcher'" <pthatcher@google.com>
Thread-Topic: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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Date: Wed, 12 Feb 2014 18:51:20 +0000
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Subject: Re: [rtcweb] Data Channel Protocol: Prevent DATA_CHANNEL_OPEN glare
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From Raju.Makaraju@alcatel-lucent.com  Wed Feb 12 10:57:05 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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> Hi,
>=20
> > I suggest that this just be called "Data Channel", not "WebRTC Data
> Channel". (Or pick some other neutral name - perhaps SCTP Data Channel.)
>=20
> Agree. A more generic name would be useful.
>=20
> "Simple Data Channel"
> "Simple SCTP Data Channel"
> Etc...

[Raju] How about "Application Data Channel"?

> (But it would still make sense for the JS API to have a WebRTC-specific n=
ame.)
+1

-raju


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From: Suhas Nandakumar <suhasietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Fwd: FW: New Version Notification for draft-nandakumar-rtcweb-sdp-04.txt
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--047d7bfceb1a31cd9704f23d1510
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Hello All

A new version of WebRTC SDP Examples draft has been submitted. The attempt
with this version is to closely align with the JSEP document.

URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-rtcweb-sdp-04.txt

Please let us know your inputs and comments.

Cheers
Suhas

--047d7bfceb1a31cd9704f23d1510
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<div dir=3D"ltr">Hello All<div><br></div><div><div class=3D"gmail_quote">
A new version of WebRTC SDP Examples draft=A0has been submitted. The attemp=
t with this version is to closely align with the JSEP document.</div><div c=
lass=3D"gmail_quote"><br>URL: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.=
ietf.org/internet-drafts/draft-nandakumar-rtcweb-sdp-04.txt" target=3D"_bla=
nk">http://www.ietf.org/internet-drafts/draft-nandakumar-rtcweb-sdp-04.txt<=
/a><br>
<br></div></div><div class=3D"gmail_quote">Please let us know your inputs a=
nd comments.</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_=
quote">Cheers</div><div class=3D"gmail_quote">Suhas</div><div class=3D"gmai=
l_quote">
<br></div></div>

--047d7bfceb1a31cd9704f23d1510--


From cb.list6@gmail.com  Wed Feb 12 22:07:01 2014
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Subject: [rtcweb] UDP transport problem
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Folks,

For about a year now, i have been very concerned about IPv4 UDP.  It
has been increasingly associated with DDoS traffic [1], to the point
that the protocol's reputation is akin to IPv4 PING in 2003 (widely
blocked).  The material point here is that IPv4 UDP is being
indiscriminately blocked and policed in access networks.  This is not
something access networks are wantonly electing to do. This decision
is made in the heat of the moment to restore service to customers and
networks who are overrun with attack traffic [2].  And it is not
simply DNS, NTP, and chargen for simple filtering.

My suggestion is that draft-ietf-rtcweb-transports be updated to
include SCTP as a transport type option.  I understand that SCTP is
not supported on Windows (tm) and most NATs.  In general, we may find
that TURN / TRAM is a required local trust anchor in access networks,
just as local ISP-administered SMTP servers in access networks are
frequently required relay points since access networks block SMTP. [3]

This UDP concern is not an issue for IPv6, since IPv6 is not, as of
yet, been associated with these massive attacks.  And when i say
massive, i mean massive [4].  And because of the scale and risk,
access networks are simply clamping IPv4 UDP throughput and moving on
and not looking back.

I  understand predicting the demise of UDP is not a popular opinion in
the IETF.  But as a network operator, and stakeholder in the internet
/ ietf / webrtc, i am hoping to share this information so that you all
can make well informed protocol decisions.



[1] http://www.us-cert.gov/ncas/alerts/TA14-017A
[2] http://www.gossamer-threads.com/lists/nanog/users/168576
[3] http://customer.comcast.com/help-and-support/internet/email-port-25-no-longer-supported/
[4] http://rt.com/news/biggest-ddos-us-cloudflare-557/


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-stun-consent-freshness-00
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Hi,
(As individual)

removing the parts that appears resolved.

On 2014-02-05 07:19, Ram Mohan R (rmohanr) wrote:

>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> On 2013-11-14 18:23, Ram Mohan R (rmohanr) wrote:
>>> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
> 
> 
> 
>>
>>
>>>>
>>>> 8. Section 4:
>>>>   If a valid STUN Binding Response is not received after 500ms, the
>>>>   STUN Binding Request is retransmitted (with the same Transaction ID
>>>>   and all other fields).  As long as a valid STUN Binding Response is
>>>>   not received, this retransmission is repeated every 500ms until Tf
>>>>   seconds have elapsed or a valid response is received.
>>>>
>>>> So no exponential back-off on the retransmission timer. I think that
>>>> is fine. However, I think it do require you to expand a bit on what
>>>> happens when one gets multiple response on the same Transaction ID.
>>>> For example additional responses do they or do they not reset the Tc
>>>> timer?
>>>
>>> Additional responses MUST reset the Tc timer.
>>>
>>
>> Okay, that appears fine.
>>
>>>
>>>>
>>>> Then I wonder about the cases where the RTT is above 500 ms, should
>>>> one then scale this factor to be once per RTT?
>>
>> What about this question?
> 
> The ICE agent can learn the RTT of a given path from RTCP reports (if
> present). If there is no RTCP reports then it can use STUN BindReq/Res to
> learn RTT.
> 
> STUN allows you to learn the RTT of a path when you sending Binding
> request and get a Binding response for a candidate pair. Before consent
> timer
> is started the browser¹s ICE agent would have done ICE checks and
> concluded on a candidate for each media stream.  So an ICE agent can use
> the RTT learnt here.
> Thoughts on this ?

I think using the STUN connectivity checks to gather an RTT estimate is
fine for this and likely the easiest. That avoids the need to have the
RTCP stack closely integrated with the consent mechanism.

And if I understand each new request has a new Transaction ID, that way
one can keep a running filtered RTT estimate to use for this. But, I do
believe that we should ha a mechanism in place to protect against RTTs >
500 ms. In buffer bloat case, as well as certain paths they do occur.

> 
> 
>>
>>>>
>>>> 9. Section 4:
>>>> "with the same Transaction ID"
>>>>
>>>> Why not use a new transaction ID for each sent packet? It would allow
>>>> tracking loss and determine RTT more reliable. All for the small cost
>>>> of keeping track of slightly more Transaction IDs?
>>>
>>> Yes, new transaction ID will help to determine the packet loss and RTT.
>>>
>>
>> Yes, so what is preferable here? Doing cloned retransmission, or
>> generating new TIDs for each outgoing request?
> 
> Using same Transaction ID would lead the peer ICE agent think this request
> as a retransmission and it may not reset its Consent timer.  So I think
> using new ID is better.

Good, lets use a new TID.

> 
> 
> 
>>
>>>
>>>
>>>>
>>>> 10. Section 4.
>>>>   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.).  Note the
>>>>   definition of a received packet is liberal, and includes an SRTP
>>>>   packet that fails authentication, a STUN Binding Request with an
>>>>   invalid USERNAME or PASSWORD, or any other packet.
>>>>
>>>> I think this requires some considerations in regards to RTT. If the
>>>> RTT is 700 ms, high for a real-time app but not unheard of at all.
>>>> Then a Tr = 1 second would fire on a single loss.
>>>
>>> Ok.  will start a discussion for this item in a separate email.
>>>
> 
> As discussed above the ICE agent can learn the RTT of a given path from
> RTCP reports (if present) or compute the RTT from STUN BindReq/Response.
> It can then use the learnt value
> to influence the Tr value.

I will leave it to you to consider how to use and specify a proposal for
this.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Thu Feb 13 00:41:36 2014
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-data-channel-06
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Hi,

Just responding to one thing. I will have a goal to review the new
versions prior to the WG session to see how well the new version
resolves my previous comments.



On 2014-02-12 03:05, Randell Jesup wrote:
> On 2/11/2014 5:47 AM, Michael Tuexen wrote:
>> On Nov 11, 2013, at 2:54 PM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>
>>
>>> 3. Section 3.2:
>>>    U-C 7:  Proxy browsing, where a browser uses data channels of a
>>>       PeerConnection to send and receive HTTP/HTTPS requests and data,
>>>       for example to avoid local internet filtering or monitoring.
>>>
>>> Yes, this may be something that is possible, but to express it as a use
>>> cases intended to be supported might not be the best. We are after all
>>> likely talking about circumventing local security policies. I would also
>>> note that "Internet" is with capital I.
>> Change to Internet. The use case seems important, since there are already
>> implementations doing this (possibly not using data channels, don't
>> know).
>> Randell: What do you think?
> 
> There are people actively planning to use it for this usecase in
> high-profile applications (using datachannels), or so I am told.

Okay, this is going to be causing issue, independent on if you include
the use case explicitly or not. Please discuss this security concerns in
the security consideration section. From my perspective this has several
implications.

1. Violating a sites security policy. Can this be detected, controlled
or prevented somehow without generally blocking the data channel or even
DTLS in general?

2. What privacy implications does this have? You are putting your
proxied traffic into another browser instances JS code space. Thus isn't
that browser going to have full insight in what the first browser is
doing, including content coming over a HTTPS from the proxy to the. This
has significant trust issues.

Cheers

Magnus Westerlund

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On 2014-02-12 18:33, Stefan Håkansson LK wrote:
> New for this version:
> 
> * The requirements are spelled out as they are derived in each use-case
> * The requirements are put in categories
> * Updated the note about that other mechanisms than ICE/STUN/TURN are allowed for nat/fw traversal
> 
> I'm not sure if there is anything more that needs to be fixed.

Thanks, I think especially the first is helping readability significant.

I intended to publication request this next week (On the 20th of Feb)
unless a consensus appear to emerge around the change Partha requested
or some significant issue is raised.

Cheers

Magnus Westerlund

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Cc: Cullen Jennings <fluffy@cisco.com>, Harald Alvestrand <hta@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2014-02-04 01:05, Justin Uberti wrote:

> 
> Agree that msid should map m-line to MST, but it would seem recv-msid
> could map RTP flow to m=line as well as indicate whether the receiver is
> in fact interested in receiving the MST associated with said m= line.
> Note that recv-msid doesn't specify an actual MSID; its purpose is to
> indicate that the receiver wants to receive data from the remote side's
> MST, and provide a mechanism for the sender to mark the data
> accordingly. (Which ends up seeming a lot like appid.)
> 

A question here, what are you really trying to reject?

Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this
media description (m= block)  in a particular direction (i.e. SDP internal)?

I think the solution differs quite a lot depending on what you really
try to accomplish.

cheers

Magnus Westerlund

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From ted.ietf@gmail.com  Thu Feb 13 07:50:40 2014
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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Adam Roach <adam@nostrum.com>,  Cullen Jennings <fluffy@cisco.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Video draft author
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Howdy,

As folks know, we have suspended discussion of mandatory to implement video
codecs for six months or so.  We do, however, a milestone for a video draft
that comes up pretty shortly after the period we've set aside comes to
attend, and there are issues which are general enough that the working
group chairs felt that it would valuable to begin working on those.  We've
asked Adam Roach to serve as author of the draft to fulfil that milestone
requirement, and he has been kind enough to agree.  We don't expect this to
be a primary focus of discussion in London, but if Adam has a chance to
create the initial text, we will ask for review at that time.

regards,

Ted

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Howdy,<br><br>As folks know, we have suspended discussion of mandato=
ry to implement video codecs for six months or so.=A0 We do, however, a mil=
estone for a video draft that comes up pretty shortly after the period we&#=
39;ve set aside comes to attend, and there are issues which are general eno=
ugh that the working group chairs felt that it would valuable to begin work=
ing on those.=A0 We&#39;ve asked Adam Roach to serve as author of the draft=
 to fulfil that milestone requirement, and he has been kind enough to agree=
.=A0 We don&#39;t expect this to be a primary focus of discussion in London=
, but if Adam has a chance to create the initial text, we will ask for revi=
ew at that time.=A0 <br>
<br>regards,<br><br>Ted<br></div></div>

--047d7bd74b10865b6904f24ba621--


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Draft Agenda RTCWeb IETF89 - Version 1 
Thread-Index: AQHPKNroA1BbAkEL7UmC4afRmUQA4g==
Date: Thu, 13 Feb 2014 16:44:49 +0000
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Subject: [rtcweb] Draft Agenda RTCWeb IETF89 - Version 1
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Here is very rough draft of what we currently have on the agenda.=20

Note that the first hour of wed conflicts with some transport stuff.=20

Thanks, the chairs


=97=97=97=97=97=97=97=97=97=97=97=97=97=97=97=97


Draft Agenda Version 1

Tuesday 9 -11:30

Chair Admin: (chairs) - 10 min=20

Data Channel:  (who ???) - 80 min=20
draft-ietf-rtcweb-data-channel
draft-ietf-rtcweb-data-protocol
<Need Open Issues Here>

Transports - Harald  - 45 min
draft-ietf-rtcweb-transports
- The appropriate form of the reference to draft-dhesikan
- MUST support ALTERNATE-SERVER
- MUST, SHOULD or MAY on TURN TCP candidates
- MUST, SHOULD or MAY on ICE TCP candidates
- Whether this draft should have normative references on anything SDP-relat=
ed (which, conceptually, are above the transport level).

Stun/DTLS heartbeat discussion: Dan Wing - 15 min
draft-ietf-rtcweb-stun-consent-freshness

Wed 9-11:30=20

Chair Admin: (chairs) - 5 min=20

JSEP: Justin  - 90 min
draft-ietf-rtcweb-jsep=20

Update from RMCAT - RMCAT Chairs - 10 min=20

Sec / Sec Arch: Eric Rescorla - 30 min=20
draft-ietf-rtcweb-security
draft-ietf-rtcweb-security-arch

RTP usage:  Colin -15 min
draft-ietf-rtcweb-rtp-usage
- Prepared for WG Last Call - Issues with latest changes?


NAT-Firewall: Still trying to resolve what parts of this are in this WG -  =
Andy  ( ?? minutes)=20
draft-hutton-rtcweb-nat-firewall-considerations-03 =


From magnus.westerlund@ericsson.com  Thu Feb 13 08:46:34 2014
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Subject: [rtcweb] Updated Potential dates for WebRTC/RTCWEB joint Interim
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WG,

We have received some feedback on the dates. After having done some
checking we propose to shift the interim meeting one day earlier to
reduce the overlap with the 3GPP meeting. Thus, the proposed dates for
the interim is 19-21 of May (Mon-Wed).

If you are willing to host, please contact the WG chairs.

Cheers

Magnus, Ted and Cullen


On 2014-01-29 08:59, Magnus Westerlund wrote:
> WG,
> 
> The chairs of this WG and the W3C are considering an joint interim
> meeting. The dates we chairs have arrived as feasible are May 20-22
> (Tue-Thu). We are considering to split up the time between the two WGs
> across all three days. So please take note of these dates in your
> calendar now.
> 
> The target region for this interim would be East Coast of North America,
> but we have at this stage no host or set location. If you are willing to
> host, please contact the chairs.
> 
> We haven't decided on this interim yet, and appreciate your input into
> holding one. The goals of the interim would be to advanced the state of
> the core documents we need to finish up. We expect that we will have
> some document that may have WG last call issues needing face to face
> time to speed up resolving them. We expect to have documents that are
> under development, like JSEP that will have thorny issues to deal with.
> Especially issues that interact with the API where we can benefit from
> having the W3C WG also present to come to joint conclusions.
> 
> Cheers
> 
> Magnus Westerlund
> (For the RTCWEB Chairs)
> 
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From nobody Thu Feb 13 09:47:39 2014
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On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
> For about a year now, i have been very concerned about IPv4 UDP.  It
> has been increasingly associated with DDoS traffic [1],

Is your concern that WebRTC will increase the potential for DoS (which
would presume the DoS mitigation measures in ICE [RFC 5245] are
insufficient), or is it just that UDP is so toxic to network operators
that you predict it will be turned off?


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On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>> For about a year now, i have been very concerned about IPv4 UDP.  It
>> has been increasingly associated with DDoS traffic [1],
>
> Is your concern that WebRTC will increase the potential for DoS (which
> would presume the DoS mitigation measures in ICE [RFC 5245] are
> insufficient), or is it just that UDP is so toxic to network operators
> that you predict it will be turned off?

My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
wise to start SCTP in the standard from the start.

Cameron


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From: Harald Alvestrand <hta@google.com>
Date: Thu, 13 Feb 2014 19:11:47 +0100
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Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP
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--089e0117696f994d5b04f24da06a
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What's being asked for is a way to control MediaStreamTracks.

The RTP streams are only incidentally impacted.

I'll put a strawman proposal into -msid- today.


On Thu, Feb 13, 2014 at 10:05 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-02-04 01:05, Justin Uberti wrote:
>
> >
> > Agree that msid should map m-line to MST, but it would seem recv-msid
> > could map RTP flow to m=3Dline as well as indicate whether the receiver=
 is
> > in fact interested in receiving the MST associated with said m=3D line.
> > Note that recv-msid doesn't specify an actual MSID; its purpose is to
> > indicate that the receiver wants to receive data from the remote side's
> > MST, and provide a mechanism for the sender to mark the data
> > accordingly. (Which ends up seeming a lot like appid.)
> >
>
> A question here, what are you really trying to reject?
>
> Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this
> media description (m=3D block)  in a particular direction (i.e. SDP
> internal)?
>
> I think the solution differs quite a lot depending on what you really
> try to accomplish.
>
> cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

--089e0117696f994d5b04f24da06a
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">What&#39;s being asked for is a way to control MediaStream=
Tracks.<div><br></div><div>The RTP streams are only incidentally impacted.<=
/div><div><br></div><div>I&#39;ll put a strawman proposal into -msid- today=
.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Feb 13, 2014 at 10:05 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=
=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerl=
und@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"">On 2014-02-04 01:05, Justin =
Uberti wrote:<br>
<br>
&gt;<br>
&gt; Agree that msid should map m-line to MST, but it would seem recv-msid<=
br>
&gt; could map RTP flow to m=3Dline as well as indicate whether the receive=
r is<br>
&gt; in fact interested in receiving the MST associated with said m=3D line=
.<br>
&gt; Note that recv-msid doesn&#39;t specify an actual MSID; its purpose is=
 to<br>
&gt; indicate that the receiver wants to receive data from the remote side&=
#39;s<br>
&gt; MST, and provide a mechanism for the sender to mark the data<br>
&gt; accordingly. (Which ends up seeming a lot like appid.)<br>
&gt;<br>
<br>
</div>A question here, what are you really trying to reject?<br>
<br>
Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this<br=
>
media description (m=3D block) =C2=A0in a particular direction (i.e. SDP in=
ternal)?<br>
<br>
I think the solution differs quite a lot depending on what you really<br>
try to accomplish.<br>
<br>
cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</blockquote></div><br></div>

--089e0117696f994d5b04f24da06a--


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On 2/13/2014 9:56 AM, Cb B wrote:
> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>> has been increasingly associated with DDoS traffic [1],
>> Is your concern that WebRTC will increase the potential for DoS (which
>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>> insufficient), or is it just that UDP is so toxic to network operators
>> that you predict it will be turned off?
> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
> wise to start SCTP in the standard from the start.
>

Why SCTP? Why not "Just like UDP only restricted to having SRTP inside" 
or even "just like UDP only a different protocol number that we like 
better"?

(Assuming that this is a good idea at all)

Matthew Kaufman


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On Thu, Feb 13, 2014 at 10:45 AM, Matthew Kaufman <matthew@matthew.at> wrote:
> On 2/13/2014 9:56 AM, Cb B wrote:
>>
>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>> <martin.thomson@gmail.com> wrote:
>>>
>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>>
>>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>>> has been increasingly associated with DDoS traffic [1],
>>>
>>> Is your concern that WebRTC will increase the potential for DoS (which
>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>> insufficient), or is it just that UDP is so toxic to network operators
>>> that you predict it will be turned off?
>>
>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
>> wise to start SCTP in the standard from the start.
>>
>
> Why SCTP? Why not "Just like UDP only restricted to having SRTP inside" or
> even "just like UDP only a different protocol number that we like better"?
>
> (Assuming that this is a good idea at all)
>

UDP2 using a different protocol number would be sufficient to
differentiate from the "bad" traffic.

Cameron

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Updated Potential dates for WebRTC/RTCWEB joint Interim
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Hi Chairs,

Thank You very much for the shift! I appreciate very much that you took the=
 3GPP meeting in consideration when planning the meeting.

Regards,

Christer



-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: rtcweb [mailto:rtcweb-bounces@ietf.org] Puolesta Magnus We=
sterlund
L=E4hetetty: 13. helmikuuta 2014 18:46
Vastaanottaja: rtcweb@ietf.org
Aihe: [rtcweb] Updated Potential dates for WebRTC/RTCWEB joint Interim

WG,

We have received some feedback on the dates. After having done some checkin=
g we propose to shift the interim meeting one day earlier to reduce the ove=
rlap with the 3GPP meeting. Thus, the proposed dates for the interim is 19-=
21 of May (Mon-Wed).

If you are willing to host, please contact the WG chairs.

Cheers

Magnus, Ted and Cullen


On 2014-01-29 08:59, Magnus Westerlund wrote:
> WG,
>=20
> The chairs of this WG and the W3C are considering an joint interim=20
> meeting. The dates we chairs have arrived as feasible are May 20-22=20
> (Tue-Thu). We are considering to split up the time between the two WGs=20
> across all three days. So please take note of these dates in your=20
> calendar now.
>=20
> The target region for this interim would be East Coast of North=20
> America, but we have at this stage no host or set location. If you are=20
> willing to host, please contact the chairs.
>=20
> We haven't decided on this interim yet, and appreciate your input into=20
> holding one. The goals of the interim would be to advanced the state=20
> of the core documents we need to finish up. We expect that we will=20
> have some document that may have WG last call issues needing face to=20
> face time to speed up resolving them. We expect to have documents that=20
> are under development, like JSEP that will have thorny issues to deal wit=
h.
> Especially issues that interact with the API where we can benefit from=20
> having the W3C WG also present to come to joint conclusions.
>=20
> Cheers
>=20
> Magnus Westerlund
> (For the RTCWEB Chairs)
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


--=20

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
F=E4r=F6gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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


From nobody Thu Feb 13 12:36:08 2014
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On Feb 13, 2014, at 7:50 PM, Cb B <cb.list6@gmail.com> wrote:

> On Thu, Feb 13, 2014 at 10:45 AM, Matthew Kaufman <matthew@matthew.at> =
wrote:
>> On 2/13/2014 9:56 AM, Cb B wrote:
>>>=20
>>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>>> <martin.thomson@gmail.com> wrote:
>>>>=20
>>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>>>=20
>>>>> For about a year now, i have been very concerned about IPv4 UDP.  =
It
>>>>> has been increasingly associated with DDoS traffic [1],
>>>>=20
>>>> Is your concern that WebRTC will increase the potential for DoS =
(which
>>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>>> insufficient), or is it just that UDP is so toxic to network =
operators
>>>> that you predict it will be turned off?
>>>=20
>>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may =
be
>>> wise to start SCTP in the standard from the start.
>>>=20
>>=20
>> Why SCTP? Why not "Just like UDP only restricted to having SRTP =
inside" or
>> even "just like UDP only a different protocol number that we like =
better"?
>>=20
>> (Assuming that this is a good idea at all)
>>=20
>=20
> UDP2 using a different protocol number would be sufficient to
> differentiate from the "bad" traffic.
... and would make sure that it does not pass NATs...
I guess this is the same problem as using SCTP/IP. For
NAT traversal through NATs not supporting SCTP one would
use SCTP/UDP/IP.

Best regards
Michael
>=20
> Cameron
>=20
>> Matthew Kaufman
>>=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
>=20


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On 02/13/2014 06:56 PM, Cb B wrote:
> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>> has been increasingly associated with DDoS traffic [1],
>> Is your concern that WebRTC will increase the potential for DoS (which
>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>> insufficient), or is it just that UDP is so toxic to network operators
>> that you predict it will be turned off?
> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
> wise to start SCTP in the standard from the start.

The bad guys will follow wherever the ports are open (and are usually
faster at writing code than the standards guys are at writing specs); so
will the traversal artists.

WebRTC over port 53, anyone?

(DNS is the one UDP-based service that's so important to the Internet,
it *cannot* be turned off unconditionally - so I expect that if UDP in
general gets blocked, port 53 will be the port 80 of UDP-land.)

-- 
Surveillance is pervasive. Go Dark.


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On Thu, Feb 13, 2014 at 12:48 PM, Harald Alvestrand
<harald@alvestrand.no> wrote:
> On 02/13/2014 06:56 PM, Cb B wrote:
>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>> <martin.thomson@gmail.com> wrote:
>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>>> has been increasingly associated with DDoS traffic [1],
>>> Is your concern that WebRTC will increase the potential for DoS (which
>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>> insufficient), or is it just that UDP is so toxic to network operators
>>> that you predict it will be turned off?
>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
>> wise to start SCTP in the standard from the start.
>
> The bad guys will follow wherever the ports are open (and are usually
> faster at writing code than the standards guys are at writing specs); so
> will the traversal artists.
>

Harald, the issue is not open ports. The issue is the entire transport
type is polluted.

> WebRTC over port 53, anyone?
>
> (DNS is the one UDP-based service that's so important to the Internet,
> it *cannot* be turned off unconditionally - so I expect that if UDP in
> general gets blocked, port 53 will be the port 80 of UDP-land.)
>

DNS runs over TCP as well.

But that's not the point.  The question is can we include native SCTP
as an option in  draft-ietf-rtcweb-transports?  What is the downside?

CB



> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 13/02/2014 4:20 PM, Cb B wrote:
> DNS runs over TCP as well. But that's not the point. The question is 
> can we include native SCTP as an option in 
> draft-ietf-rtcweb-transports? What is the downside? CB

The biggest downside, as I see it, is that 90%+ of the world runs 
Windows and that platform does not support SCTP natively (a fact that 
you have mentioned before). I am all in favor of SCTP, but I think you 
need to solve the Windows problem before this solution become meaningful.

Gili


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On Thu, Feb 13, 2014 at 1:20 PM, Cb B <cb.list6@gmail.com> wrote:

>
> But that's not the point.  The question is can we include native SCTP
> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>
> CB
>

Howdy,

Given previous decisions of the working group, you would need to describe
how to provide confidentiality.  At the moment, I know only of TLS over
SCTP (which would reverse the current mapping) and SCTP over IPSec (which
would likely actually be IPSec/UDP); do you know of other methods which
would keep SCTP at the top layer of the protocol stack and still provide
confidentiality?

regarsd,

Ted

--20cf3010eb89f0a2af04f2514335
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Feb 13, 2014 at 1:20 PM=
, Cb B <span dir=3D"ltr">&lt;<a href=3D"mailto:cb.list6@gmail.com" target=
=3D"_blank">cb.list6@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>
But that&#39;s not the point. =A0The question is can we include native SCTP=
<br>
as an option in =A0draft-ietf-rtcweb-transports? =A0What is the downside?<b=
r>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
CB</font></span><br></blockquote><div><br><div class=3D"gmail_default" styl=
e=3D"font-family:georgia,serif;display:inline">Howdy,<br><br></div><div cla=
ss=3D"gmail_default" style=3D"font-family:georgia,serif">Given previous dec=
isions of the working group, you would need to describe how to provide conf=
identiality.=A0 At the moment, I know only of TLS over SCTP (which would re=
verse the current mapping) and SCTP over IPSec (which would likely actually=
 be IPSec/UDP); do you know of other methods which would keep SCTP at the t=
op layer of the protocol stack and still provide confidentiality?<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
regarsd,<br><br>Ted<br></div><br></div></div></div></div>

--20cf3010eb89f0a2af04f2514335--


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On Thu, Feb 13, 2014 at 2:32 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
> On Thu, Feb 13, 2014 at 1:20 PM, Cb B <cb.list6@gmail.com> wrote:
>>
>>
>> But that's not the point.  The question is can we include native SCTP
>> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>>
>> CB
>
>
> Howdy,
>
> Given previous decisions of the working group, you would need to describe
> how to provide confidentiality.  At the moment, I know only of TLS over SCTP
> (which would reverse the current mapping) and SCTP over IPSec (which would
> likely actually be IPSec/UDP); do you know of other methods which would keep
> SCTP at the top layer of the protocol stack and still provide
> confidentiality?
>
> regarsd,
>
> Ted
>

I believe the cleanest would be to do TLS over SCTP.  I understand
that changes the top of the transport stack, and create variation.
But, i don't think current SCTP over DTLS plan has won any beauty
prizes.  If this is standards track work that must stand the test of
time, TLS over SCTP seems appropriate.

CB


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On 02/13/2014 10:20 PM, Cb B wrote:
> On Thu, Feb 13, 2014 at 12:48 PM, Harald Alvestrand
> <harald@alvestrand.no> wrote:
>> On 02/13/2014 06:56 PM, Cb B wrote:
>>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>>> <martin.thomson@gmail.com> wrote:
>>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>>>> has been increasingly associated with DDoS traffic [1],
>>>> Is your concern that WebRTC will increase the potential for DoS (which
>>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>>> insufficient), or is it just that UDP is so toxic to network operators
>>>> that you predict it will be turned off?
>>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
>>> wise to start SCTP in the standard from the start.
>> The bad guys will follow wherever the ports are open (and are usually
>> faster at writing code than the standards guys are at writing specs); so
>> will the traversal artists.
>>
> Harald, the issue is not open ports. The issue is the entire transport
> type is polluted.

And I think that notion is bogus. Faced with the choice of dealing with
the devil they know (UDP) and the devil they don't have a clue about
(SCTP), firewall operators are going to stick with building out the
rulesets around UDP, and continuing to refuse all the "new stuff".

So I don't think the plan has legs - adding SCTP without the UDP wrapper
to our options will not help anyone do anything.

But this is trying to forecast the thinking and action of humans that
are not me, which is something I frequently get wrong - so other
opinions would be welcome.

>
>> WebRTC over port 53, anyone?
>>
>> (DNS is the one UDP-based service that's so important to the Internet,
>> it *cannot* be turned off unconditionally - so I expect that if UDP in
>> general gets blocked, port 53 will be the port 80 of UDP-land.)
>>
> DNS runs over TCP as well.

For AXFR, yes. For daily lookups .... better not tell any root DNS
operator you're advocating that they should have all their lookups over
TCP; you wouldn't like the expletives that result.

>
> But that's not the point.  The question is can we include native SCTP
> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>

Clutter.



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On Thu, Feb 13, 2014 at 2:37 PM, Cb B <cb.list6@gmail.com> wrote:

> On Thu, Feb 13, 2014 at 2:32 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
> > On Thu, Feb 13, 2014 at 1:20 PM, Cb B <cb.list6@gmail.com> wrote:
> >>
> >>
> >> But that's not the point.  The question is can we include native SCTP
> >> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
> >>
> >> CB
> >
> >
> > Howdy,
> >
> > Given previous decisions of the working group, you would need to describe
> > how to provide confidentiality.  At the moment, I know only of TLS over
> SCTP
> > (which would reverse the current mapping) and SCTP over IPSec (which
> would
> > likely actually be IPSec/UDP); do you know of other methods which would
> keep
> > SCTP at the top layer of the protocol stack and still provide
> > confidentiality?
> >
> > regarsd,
> >
> > Ted
> >
>
> I believe the cleanest would be to do TLS over SCTP.  I understand
> that changes the top of the transport stack, and create variation.
> But, i don't think current SCTP over DTLS plan has won any beauty
> prizes.  If this is standards track work that must stand the test of
> time, TLS over SCTP seems appropriate.
>
>
This requires a completely new method for providing channel abstractions.

If you would like the working group to consider a change of that magnitude,
a worked draft describing a solution that meets all of the use cases set
out in
draft-ietf-rtcweb-data-channel-07 seems to me the bare minimum to consider
overturning the established working group consensus on this point.

Speaking personally, I do not see sufficient SCTP/IP deployment to warrant
further discussion on this point.  The working group has invested
considerable
time and effort on this approach and the likely result of this change seems
to
me either a less deployable system or one which will deploy later because
of the
need to start discussions over.

regards,

Ted Hardie




> CB
>

--047d7bd8fbac6c387b04f251735b
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Feb 13, 2014 at 2:37 PM=
, Cb B <span dir=3D"ltr">&lt;<a href=3D"mailto:cb.list6@gmail.com" target=
=3D"_blank">cb.list6@gmail.com</a>&gt;</span> wrote:<br><div class=3D"gmail=
_quote"><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 class=3D""><div class=3D"h5">On Thu, Feb 13, 2014 at 2:32 PM, Ted Hard=
ie &lt;<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>&gt; wro=
te:<br>
&gt; On Thu, Feb 13, 2014 at 1:20 PM, Cb B &lt;<a href=3D"mailto:cb.list6@g=
mail.com">cb.list6@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; But that&#39;s not the point. =A0The question is can we include na=
tive SCTP<br>
&gt;&gt; as an option in =A0draft-ietf-rtcweb-transports? =A0What is the do=
wnside?<br>
&gt;&gt;<br>
&gt;&gt; CB<br>
&gt;<br>
&gt;<br>
&gt; Howdy,<br>
&gt;<br>
&gt; Given previous decisions of the working group, you would need to descr=
ibe<br>
&gt; how to provide confidentiality. =A0At the moment, I know only of TLS o=
ver SCTP<br>
&gt; (which would reverse the current mapping) and SCTP over IPSec (which w=
ould<br>
&gt; likely actually be IPSec/UDP); do you know of other methods which woul=
d keep<br>
&gt; SCTP at the top layer of the protocol stack and still provide<br>
&gt; confidentiality?<br>
&gt;<br>
&gt; regarsd,<br>
&gt;<br>
&gt; Ted<br>
&gt;<br>
<br>
</div></div>I believe the cleanest would be to do TLS over SCTP. =A0I under=
stand<br>
that changes the top of the transport stack, and create variation.<br>
But, i don&#39;t think current SCTP over DTLS plan has won any beauty<br>
prizes. =A0If this is standards track work that must stand the test of<br>
time, TLS over SCTP seems appropriate.<br>
<span class=3D""><font color=3D"#888888"><br></font></span></blockquote><di=
v><br><div class=3D"gmail_default" style=3D"font-family:georgia,serif">This=
 requires a completely new method for providing channel abstractions.=A0 <b=
r><br>
If you would like the working group to consider a change of that magnitude,=
<br>a worked draft describing a solution that meets all of the use cases se=
t out in<br>draft-ietf-rtcweb-data-channel-07 seems to me the bare minimum =
to consider<br>
overturning the established working group consensus on this point.<br><br><=
/div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">Speak=
ing personally, I do not see sufficient SCTP/IP deployment to warrant<br>
further discussion on this point.=A0 The working group has invested conside=
rable<br>time and effort on this approach and the likely result of this cha=
nge seems to<br>me either a less deployable system or one which will deploy=
 later because of the<br>
need to start discussions over.<br><br></div><div class=3D"gmail_default" s=
tyle=3D"font-family:georgia,serif">regards,<br><br>Ted Hardie<br></div><br>=
<br>=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">
<span class=3D""><font color=3D"#888888">
CB<br>
</font></span></blockquote></div><br></div></div>

--047d7bd8fbac6c387b04f251735b--


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On Thu, Feb 13, 2014 at 5:28 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
> On 13/02/2014 4:20 PM, Cb B wrote:
>>
>> DNS runs over TCP as well. But that's not the point. The question is can
>> we include native SCTP as an option in draft-ietf-rtcweb-transports? Wha=
t is
>> the downside? CB
>
>
> The biggest downside, as I see it, is that 90%+ of the world runs Windows
> and that platform does not support SCTP natively (a fact that you have
> mentioned before). I am all in favor of SCTP, but I think you need to sol=
ve
> the Windows problem before this solution become meaningful.

The biggest downside, as I see it, is targeting advancements in the state o=
f
the art, at windows. 98% of the world or so run non-windows based cell
phones and tablets, and in terms of total users, probably outnumber
the windows contingent at this point.

SCTP and MPTCP are quite feasible on android and IOS.

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



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


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On Thu, Feb 13, 2014 at 2:45 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
> On Thu, Feb 13, 2014 at 2:37 PM, Cb B <cb.list6@gmail.com> wrote:
>>
>> On Thu, Feb 13, 2014 at 2:32 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
>> > On Thu, Feb 13, 2014 at 1:20 PM, Cb B <cb.list6@gmail.com> wrote:
>> >>
>> >>
>> >> But that's not the point.  The question is can we include native SCTP
>> >> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>> >>
>> >> CB
>> >
>> >
>> > Howdy,
>> >
>> > Given previous decisions of the working group, you would need to
>> > describe
>> > how to provide confidentiality.  At the moment, I know only of TLS over
>> > SCTP
>> > (which would reverse the current mapping) and SCTP over IPSec (which
>> > would
>> > likely actually be IPSec/UDP); do you know of other methods which would
>> > keep
>> > SCTP at the top layer of the protocol stack and still provide
>> > confidentiality?
>> >
>> > regarsd,
>> >
>> > Ted
>> >
>>
>> I believe the cleanest would be to do TLS over SCTP.  I understand
>> that changes the top of the transport stack, and create variation.
>> But, i don't think current SCTP over DTLS plan has won any beauty
>> prizes.  If this is standards track work that must stand the test of
>> time, TLS over SCTP seems appropriate.
>>
>
> This requires a completely new method for providing channel abstractions.
>
> If you would like the working group to consider a change of that magnitude,
> a worked draft describing a solution that meets all of the use cases set out
> in
> draft-ietf-rtcweb-data-channel-07 seems to me the bare minimum to consider
> overturning the established working group consensus on this point.
>
> Speaking personally, I do not see sufficient SCTP/IP deployment to warrant
> further discussion on this point.  The working group has invested
> considerable
> time and effort on this approach and the likely result of this change seems
> to
> me either a less deployable system or one which will deploy later because of
> the
> need to start discussions over.
>
> regards,
>
> Ted Hardie
>

Acknowledged.  That is approximately the story arc i expected before I
initiated the conversation.

CB

>
>
>>
>> CB
>
>


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On 13/02/2014 5:46 PM, Dave Taht wrote:
> The biggest downside, as I see it, is targeting advancements in the state of
> the art, at windows. 98% of the world or so run non-windows based cell
> phones and tablets, and in terms of total users, probably outnumber
> the windows contingent at this point.
>
> SCTP and MPTCP are quite feasible on android and IOS.

It doesn't matter how many smartphones there are. What matters is how 
many of them will be used to do meaningful video chat. The screen real 
estate on these devices is way too small.

So yes, mobile is huge, but from the point of view of WebRTC, Windows is 
still king.

Gili


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On Thu, Feb 13, 2014 at 2:44 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
> On 02/13/2014 10:20 PM, Cb B wrote:
>> On Thu, Feb 13, 2014 at 12:48 PM, Harald Alvestrand
>> <harald@alvestrand.no> wrote:
>>> On 02/13/2014 06:56 PM, Cb B wrote:
>>>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>>>> <martin.thomson@gmail.com> wrote:
>>>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>>>>> has been increasingly associated with DDoS traffic [1],
>>>>> Is your concern that WebRTC will increase the potential for DoS (which
>>>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>>>> insufficient), or is it just that UDP is so toxic to network operators
>>>>> that you predict it will be turned off?
>>>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
>>>> wise to start SCTP in the standard from the start.
>>> The bad guys will follow wherever the ports are open (and are usually
>>> faster at writing code than the standards guys are at writing specs); so
>>> will the traversal artists.
>>>
>> Harald, the issue is not open ports. The issue is the entire transport
>> type is polluted.
>
> And I think that notion is bogus. Faced with the choice of dealing with
> the devil they know (UDP) and the devil they don't have a clue about
> (SCTP), firewall operators are going to stick with building out the
> rulesets around UDP, and continuing to refuse all the "new stuff".
>

I never mentioned firewall operators.  I mentioned access networks and
internet backbone operators.

As many network operators know, there is a big issue with UDP going on
that is crushing some networks

http://blog.cloudflare.com/technical-details-behind-a-400gbps-ntp-amplification-ddos-attack

https://puck.nether.net/pipermail/outages/2014-February/006651.html

This is the point in bring to the WG.

> So I don't think the plan has legs - adding SCTP without the UDP wrapper
> to our options will not help anyone do anything.
>

it's a path out.

> But this is trying to forecast the thinking and action of humans that
> are not me, which is something I frequently get wrong - so other
> opinions would be welcome.
>
>>
>>> WebRTC over port 53, anyone?
>>>
>>> (DNS is the one UDP-based service that's so important to the Internet,
>>> it *cannot* be turned off unconditionally - so I expect that if UDP in
>>> general gets blocked, port 53 will be the port 80 of UDP-land.)
>>>
>> DNS runs over TCP as well.
>
> For AXFR, yes. For daily lookups .... better not tell any root DNS
> operator you're advocating that they should have all their lookups over
> TCP; you wouldn't like the expletives that result.
>

cute. But, TCP works.  Maybe you can ask someone at Android to support
EDNS0 so that TCP is used less.  Because for now, android required TCP
for any response over 512 bytes.

>>
>> But that's not the point.  The question is can we include native SCTP
>> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>>
>
> Clutter.
>
>


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On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
> On 13/02/2014 5:46 PM, Dave Taht wrote:
>>
>> The biggest downside, as I see it, is targeting advancements in the stat=
e
>> of
>> the art, at windows. 98% of the world or so run non-windows based cell
>> phones and tablets, and in terms of total users, probably outnumber
>> the windows contingent at this point.
>>
>> SCTP and MPTCP are quite feasible on android and IOS.
>
>
> It doesn't matter how many smartphones there are. What matters is how man=
y
> of them will be used to do meaningful video chat. The screen real estate =
on
> these devices is way too small.

In my experience, everybody is using tablets and handheld devices for
video chat.
It is a natural extension of the usage of the device to extend it from
phone calls
to video calls.

The lack of a working camera on most desktops is a hindrance, and the place=
ment
of cameras on most laptops is not ideal.

>
> So yes, mobile is huge, but from the point of view of WebRTC, Windows is
> still king.

I am under the impression that this is actually backed by statistics, but a=
m
interested in recent statistics for things like google hangouts and facetim=
e.

> Gili



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


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On Thu, Feb 13, 2014 at 5:57 PM, Cb B <cb.list6@gmail.com> wrote:

> cute. But, TCP works.  Maybe you can ask someone at Android to support
> EDNS0 so that TCP is used less.  Because for now, android required TCP
> for any response over 512 bytes.

Android, in part, uses dnsmasq, which we are busily adding dnssec support t=
o.

https://www.internetsociety.org/deploy360/blog/2014/02/weekend-project-4/

EDNS0 is supported.

I would certainly like to see android's DNS handling to be improved.

>
>>>
>>> But that's not the point.  The question is can we include native SCTP
>>> as an option in  draft-ietf-rtcweb-transports?  What is the downside?
>>>
>>
>> Clutter.

>From a QoS standpoint I would not mind at all if it were possible to run
webrtc (and things like QUIC) over their own protocol numbers. I
realize this would
make webrtc more easily identifiable, but so long as the e2e encryption is =
good,
the ability for operators and edge devices to optimize for highly
interactive traffic
would be enhanced.

I don't have a lot of hope for SCTP, do have more hope for mptcp, and
the ipv6 rollout is seemingly accelerating.

https://www.internetsociety.org/deploy360/blog/2014/02/googles-ipv6-stats-p=
ass-3-less-than-5-months-after-passing-2/

I am under the impression ipv6 support for webrtc landed in chrome recently=
.

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



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


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Most important point of reality anchoring (DNS over UDP vs TCP) is at
the bottom.
Other comments are incidental.

On 02/13/2014 11:57 PM, Cb B wrote:
> On Thu, Feb 13, 2014 at 2:44 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
>> On 02/13/2014 10:20 PM, Cb B wrote:
>>> On Thu, Feb 13, 2014 at 12:48 PM, Harald Alvestrand
>>> <harald@alvestrand.no> wrote:
>>>> On 02/13/2014 06:56 PM, Cb B wrote:
>>>>> On Thu, Feb 13, 2014 at 9:47 AM, Martin Thomson
>>>>> <martin.thomson@gmail.com> wrote:
>>>>>> On 12 February 2014 22:06, Cb B <cb.list6@gmail.com> wrote:
>>>>>>> For about a year now, i have been very concerned about IPv4 UDP.  It
>>>>>>> has been increasingly associated with DDoS traffic [1],
>>>>>> Is your concern that WebRTC will increase the potential for DoS (which
>>>>>> would presume the DoS mitigation measures in ICE [RFC 5245] are
>>>>>> insufficient), or is it just that UDP is so toxic to network operators
>>>>>> that you predict it will be turned off?
>>>>> My concern is that IPv4 UDP is so toxic it will be blocked.  It may be
>>>>> wise to start SCTP in the standard from the start.
>>>> The bad guys will follow wherever the ports are open (and are usually
>>>> faster at writing code than the standards guys are at writing specs); so
>>>> will the traversal artists.
>>>>
>>> Harald, the issue is not open ports. The issue is the entire transport
>>> type is polluted.
>> And I think that notion is bogus. Faced with the choice of dealing with
>> the devil they know (UDP) and the devil they don't have a clue about
>> (SCTP), firewall operators are going to stick with building out the
>> rulesets around UDP, and continuing to refuse all the "new stuff".
>>
> I never mentioned firewall operators.  I mentioned access networks and
> internet backbone operators.
>
> As many network operators know, there is a big issue with UDP going on
> that is crushing some networks
>
> http://blog.cloudflare.com/technical-details-behind-a-400gbps-ntp-amplification-ddos-attack
That's an UDP port 123 (NTP) matter. We've done several of these
exercises (live-fire updates of critical Internet infrastructure) over
the last few years.

>
> https://puck.nether.net/pipermail/outages/2014-February/006651.html
>
> This is the point in bring to the WG.

Yes, thanks for bringing it.
I just happen to totally disagree with your conclusions.

>
>> So I don't think the plan has legs - adding SCTP without the UDP wrapper
>> to our options will not help anyone do anything.
>>
> it's a path out.

I don't see any property of this path that makes it a) traversable, or
b) "out".

>
>> But this is trying to forecast the thinking and action of humans that
>> are not me, which is something I frequently get wrong - so other
>> opinions would be welcome.
>>
>>>> WebRTC over port 53, anyone?
>>>>
>>>> (DNS is the one UDP-based service that's so important to the Internet,
>>>> it *cannot* be turned off unconditionally - so I expect that if UDP in
>>>> general gets blocked, port 53 will be the port 80 of UDP-land.)
>>>>
>>> DNS runs over TCP as well.
>> For AXFR, yes. For daily lookups .... better not tell any root DNS
>> operator you're advocating that they should have all their lookups over
>> TCP; you wouldn't like the expletives that result.
>>
> cute. But, TCP works.  Maybe you can ask someone at Android to support
> EDNS0 so that TCP is used less.  Because for now, android required TCP
> for any response over 512 bytes.

I'm talking about
http://k.root-servers.org/statistics/ams-ix/ip_protocols.html

Note the relative width of the green (TCP) against the purple (UDP).
(I've been unable to find a green pixel on those graphs.)

-- 
Surveillance is pervasive. Go Dark.


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Date: Thu, 13 Feb 2014 18:50:38 -0500
From: cowwoc <cowwoc@bbs.darktech.org>
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On 13/02/2014 6:00 PM, Dave Taht wrote:
> On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>> On 13/02/2014 5:46 PM, Dave Taht wrote:
>>> The biggest downside, as I see it, is targeting advancements in the state
>>> of
>>> the art, at windows. 98% of the world or so run non-windows based cell
>>> phones and tablets, and in terms of total users, probably outnumber
>>> the windows contingent at this point.
>>>
>>> SCTP and MPTCP are quite feasible on android and IOS.
>>
>> It doesn't matter how many smartphones there are. What matters is how many
>> of them will be used to do meaningful video chat. The screen real estate on
>> these devices is way too small.
> In my experience, everybody is using tablets and handheld devices for
> video chat.
> It is a natural extension of the usage of the device to extend it from
> phone calls
> to video calls.
>
> The lack of a working camera on most desktops is a hindrance, and the placement
> of cameras on most laptops is not ideal.

All laptops and tablets come with decent cameras. I will agree that 
WebRTC on tablets will be strong, but smartphones is really pushing it. 
Most of the time I've seen people engage in video chat it was between 
family members; far less for business use. And in those cases, I've seen 
people jump on tablets and laptops instead of having to deal with a 
tiny, underpowered smartphone for video. These are just personal 
observations, not science, so please take them with a grain of salt.

Gili


From nobody Thu Feb 13 15:55:01 2014
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On Thu, Feb 13, 2014 at 6:50 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
> On 13/02/2014 6:00 PM, Dave Taht wrote:
>>
>> On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>>>
>>> On 13/02/2014 5:46 PM, Dave Taht wrote:
>>>>
>>>> The biggest downside, as I see it, is targeting advancements in the
>>>> state
>>>> of
>>>> the art, at windows. 98% of the world or so run non-windows based cell
>>>> phones and tablets, and in terms of total users, probably outnumber
>>>> the windows contingent at this point.
>>>>
>>>> SCTP and MPTCP are quite feasible on android and IOS.
>>>
>>>
>>> It doesn't matter how many smartphones there are. What matters is how
>>> many
>>> of them will be used to do meaningful video chat. The screen real estat=
e
>>> on
>>> these devices is way too small.
>>
>> In my experience, everybody is using tablets and handheld devices for
>> video chat.
>> It is a natural extension of the usage of the device to extend it from
>> phone calls
>> to video calls.
>>
>> The lack of a working camera on most desktops is a hindrance, and the
>> placement
>> of cameras on most laptops is not ideal.
>
>
> All laptops and tablets come with decent cameras. I will agree that WebRT=
C
> on tablets will be strong, but smartphones is really pushing it. Most of =
the
> time I've seen people engage in video chat it was between family members;
> far less for business use. And in those cases, I've seen people jump on
> tablets and laptops instead of having to deal with a tiny, underpowered
> smartphone for video. These are just personal observations, not science, =
so
> please take them with a grain of salt.

I will clarify what I said above is that what I observe among the under 20s
is huge tablet and handheld use for videoconferencing. For business use
I see primarily things like dedicated hardware, webex and to a smaller
extent hangouts,
and facetime.


> Gili



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


From nobody Thu Feb 13 17:20:19 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-05.txt
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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 Audio Codec and Processing Requirements
        Authors         : Jean-Marc Valin
                          Cary Bran
	Filename        : draft-ietf-rtcweb-audio-05.txt
	Pages           : 6
	Date            : 2014-02-13

Abstract:
   This document outlines the audio codec and processing requirements
   for WebRTC client application and endpoint devices.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-05


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Thu Feb 13 22:24:44 2014
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] WebRTC base P2P streaming
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Date: Fri, 14 Feb 2014 06:24:11 +0000
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--_000_51E6A56BD6A85142B9D172C87FC3ABBB4591F709nkgeml501mbschi_
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Hi all,

We have just published a draft in PPSP working group to discuss the current=
 emerging WebRTC based P2P streaming applications. If you're interested in =
this, comments and suggestions are appreciated.
http://tools.ietf.org/html/draft-huang-ppsp-p2p-webrtc-survey-00
This draft will be discussed in the PPSP WG in the coming London meeting.

Best regards,
Rachel


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	text-decoration:underline;}
span.EmailStyle17
	{mso-style-type:personal-compose;
	font-family:"Calibri","sans-serif";
	color:windowtext;}
.MsoChpDefault
	{mso-style-type:export-only;}
@page WordSection1
	{size:612.0pt 792.0pt;
	margin:72.0pt 90.0pt 72.0pt 90.0pt;}
div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi all,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We have just published a draft in PPSP working group=
 to discuss the current emerging WebRTC based P2P streaming applications. I=
f you&#8217;re interested in this, comments and suggestions are appreciated=
.<o:p></o:p></p>
<p class=3D"MsoNormal"><a href=3D"http://tools.ietf.org/html/draft-huang-pp=
sp-p2p-webrtc-survey-00">http://tools.ietf.org/html/draft-huang-ppsp-p2p-we=
brtc-survey-00</a><o:p></o:p></p>
<p class=3D"MsoNormal">This draft will be discussed in the PPSP WG in the c=
oming London meeting.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-align:justify;text-justify:inter-ideog=
raph"><span style=3D"font-size:10.5pt">Best regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-align:justify;text-justify:inter-ideog=
raph"><span style=3D"font-size:10.5pt">Rachel
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_51E6A56BD6A85142B9D172C87FC3ABBB4591F709nkgeml501mbschi_--


From nobody Thu Feb 13 23:20:42 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: Cb B <cb.list6@gmail.com>
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Subject: Re: [rtcweb] UDP transport problem
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On Feb 13, 2014, at 11:37 PM, Cb B <cb.list6@gmail.com> wrote:

> On Thu, Feb 13, 2014 at 2:32 PM, Ted Hardie <ted.ietf@gmail.com> =
wrote:
>> On Thu, Feb 13, 2014 at 1:20 PM, Cb B <cb.list6@gmail.com> wrote:
>>>=20
>>>=20
>>> But that's not the point.  The question is can we include native =
SCTP
>>> as an option in  draft-ietf-rtcweb-transports?  What is the =
downside?
>>>=20
>>> CB
>>=20
>>=20
>> Howdy,
>>=20
>> Given previous decisions of the working group, you would need to =
describe
>> how to provide confidentiality.  At the moment, I know only of TLS =
over SCTP
>> (which would reverse the current mapping) and SCTP over IPSec (which =
would
>> likely actually be IPSec/UDP); do you know of other methods which =
would keep
>> SCTP at the top layer of the protocol stack and still provide
>> confidentiality?
>>=20
>> regarsd,
>>=20
>> Ted
>>=20
>=20
> I believe the cleanest would be to do TLS over SCTP.  I understand
> that changes the top of the transport stack, and create variation.
> But, i don't think current SCTP over DTLS plan has won any beauty
> prizes.  If this is standards track work that must stand the test of
> time, TLS over SCTP seems appropriate.
Using TLS over SCTP as defined in
http://tools.ietf.org/search/rfc3436
doesn't support all features of SCTP, for example partial reliability,
and it doesn't scale well (It needs a TLS connection per bidirectional
stream, so per data channel in our case).
Using DTLS over SCTP resolves these issues:
http://tools.ietf.org/search/rfc6083

DTLS over SCTP is supported by OpenSSL.

Best regards
Michael
>=20
> CB
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Fri Feb 14 01:15:55 2014
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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           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-jsep-06.txt
	Pages           : 51
	Date            : 2014-02-14

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-06


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb 14 02:20:58 2014
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On 2/12/2014 10:06 PM, Cb B wrote:
> For about a year now, i have been very concerned about IPv4 UDP.  It
> has been increasingly associated with DDoS traffic [1], to the point
> that the protocol's reputation is akin to IPv4 PING in 2003 (widely
> blocked).  The material point here is that IPv4 UDP is being
> indiscriminately blocked and policed in access networks.  This is not
> something access networks are wantonly electing to do. This decision
> is made in the heat of the moment to restore service to customers and
> networks who are overrun with attack traffic [2].  And it is not
> simply DNS, NTP, and chargen for simple filtering.
>
> My suggestion is that draft-ietf-rtcweb-transports be updated to
> include SCTP as a transport type option.  I understand that SCTP is
> not supported on Windows (tm) and most NATs.

Side note: it would be ironic to end up using 
DataChannel-over-SCTP-over-DTLS-over-SCTP

> In general, we may find
> that TURN / TRAM is a required local trust anchor in access networks,
> just as local ISP-administered SMTP servers in access networks are
> frequently required relay points since access networks block SMTP. [3]

I certainly understand the reason for concern; but the idea that UDP 
would become limited to TURN-mediated is frightening and depressing (and 
whose TURN and how does the ISP know it's TURN-mediated unless it's an 
ISP's TURN?)

One significant piece of the value proposition of WebRTC is that through 
use of ICE (and TURN) it will usually bypass the need for relays, 
significantly improving media quality (delay), dramatically cutting the 
cost of operation for services, easing deployment, and avoiding a number 
of levels of control/ability-to-block/etc.

It's especially depressing in that we put significant effort into 
reducing the likelihood that WebRTC could be used for DDoS attacks.

I will note that blocking UDP (or massively-rate-limiting it) will have 
all sorts of nasty effects on all forms of VoIP.  TCP-entrained VoIP can 
evade that, but at a serious cost to call quality.  Surely the operators 
know this.

> This UDP concern is not an issue for IPv6, since IPv6 is not, as of
> yet, been associated with these massive attacks.  And when i say
> massive, i mean massive [4].  And because of the scale and risk,
> access networks are simply clamping IPv4 UDP throughput and moving on
> and not looking back.

Can the DNS amplification attack be used to hit any ports other than 
53?  Can clamps be limited to port 53

> I  understand predicting the demise of UDP is not a popular opinion in
> the IETF.  But as a network operator, and stakeholder in the internet
> / ietf / webrtc, i am hoping to share this information so that you all
> can make well informed protocol decisions.

Thank you


-- 
Randell Jesup
randell-ietf@jesup.org


From nobody Fri Feb 14 06:22:55 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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Hi,

Section 4.1.1. defines, for the PeerConnection constructor, the possibility=
 to specify BUNDLE policy.

However, I am missing the possibility to indicate whether, within a BUNDLE =
group, the same port number shall be used or not.=20

Section 5.2.1 does say:

	"o  The port value is set to the port of the default ICE candidate for
      	this m=3D section; if this m=3D section is not being bundled into
      	another m=3D section, the port value MUST be unique."

...but, it doesn't talk about the case when the port IS being bundled into =
another m=3D section. Normally, in the initial Offer, the port value will b=
e unique also in that case, but if the PeerConnection is created due to for=
king, and it is known that the remote peer supports BUNDLE, then identical =
port values can be used already in the initial offer.

Regards,

Christer


From nobody Fri Feb 14 06:26:49 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - SDP sctpmap attribute usage
Thread-Index: Ac8pkGsqajJyywweSF21x5yZu3cFWA==
Date: Fri, 14 Feb 2014 14:26:42 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - SDP sctpmap attribute usage
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Hi,

Section 5.2.1 says:

	"Within the data m=3D section, the "a=3Dmid", "a=3Dice-ufrag", "a=3Dice-
   	passwd", "a=3Dice-options", "a=3Dcandidate", "a=3Dfingerprint", and
   	"a=3Dsetup" lines MUST be included as mentioned above, along with an
   	"a=3Dsctpmap" line referencing the SCTP port number and specifying the
   	application protocol indicated in [I-D.ietf-rtcweb-data-protocol]."

The statement that the sctpmap attribute specifies the application protocol=
 indicate in the DCP is not aligned with the sctp-sdp draft. The sctp-sdp d=
raft specifies that the attribute specifies the application running on top =
of the SCTP association - which in this case is the WebRTC Data Channel. Th=
e protocol running on top of that data channel is then specified in the DCP=
.

Regards,

Christer


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From: Cb B <cb.list6@gmail.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] UDP transport problem
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On Feb 14, 2014 2:20 AM, "Randell Jesup" <randell-ietf@jesup.org> wrote:
>
> On 2/12/2014 10:06 PM, Cb B wrote:
>>
>> For about a year now, i have been very concerned about IPv4 UDP.  It
>> has been increasingly associated with DDoS traffic [1], to the point
>> that the protocol's reputation is akin to IPv4 PING in 2003 (widely
>> blocked).  The material point here is that IPv4 UDP is being
>> indiscriminately blocked and policed in access networks.  This is not
>> something access networks are wantonly electing to do. This decision
>> is made in the heat of the moment to restore service to customers and
>> networks who are overrun with attack traffic [2].  And it is not
>> simply DNS, NTP, and chargen for simple filtering.
>>
>> My suggestion is that draft-ietf-rtcweb-transports be updated to
>> include SCTP as a transport type option.  I understand that SCTP is
>> not supported on Windows (tm) and most NATs.
>
>
> Side note: it would be ironic to end up using
DataChannel-over-SCTP-over-DTLS-over-SCTP
>

The history of p2p applications such as skype have shown history favors the
nimble.

>
>> In general, we may find
>> that TURN / TRAM is a required local trust anchor in access networks,
>> just as local ISP-administered SMTP servers in access networks are
>> frequently required relay points since access networks block SMTP. [3]
>
>
> I certainly understand the reason for concern; but the idea that UDP
would become limited to TURN-mediated is frightening and depressing (and
whose TURN and how does the ISP know it's TURN-mediated unless it's an
ISP's TURN?)
>
> One significant piece of the value proposition of WebRTC is that through
use of ICE (and TURN) it will usually bypass the need for relays,
significantly improving media quality (delay), dramatically cutting the
cost of operation for services, easing deployment, and avoiding a number of
levels of control/ability-to-block/etc.
>
> It's especially depressing in that we put significant effort into
reducing the likelihood that WebRTC could be used for DDoS attacks.
>
> I will note that blocking UDP (or massively-rate-limiting it) will have
all sorts of nasty effects on all forms of VoIP.  TCP-entrained VoIP can
evade that, but at a serious cost to call quality.  Surely the operators
know this.
>

Agreed on all points. My view is one related to the basic requirement of
keeping the network up.  I hope i have provided enough reference points to
make the magnitude of the problem clear as well as how history has shown
protocols get blocked (smtp)

>
>> This UDP concern is not an issue for IPv6, since IPv6 is not, as of
>> yet, been associated with these massive attacks.  And when i say
>> massive, i mean massive [4].  And because of the scale and risk,
>> access networks are simply clamping IPv4 UDP throughput and moving on
>> and not looking back.
>
>
> Can the DNS amplification attack be used to hit any ports other than 53?
 Can clamps be limited to port 53
>

Right. The source port will be a function of the amp server such as 53,
123, 19... all very common. The destination port is of the spoofers
choosing.

But, as the us-cert advisory notes, there are several impacted appilication
ports. Well-known webrtc fixed ports would help for a white list.

Creating a blacklist of just 53, 123, 19 makes us still behind the curve on
whatever is next.

CB

>
>> I  understand predicting the demise of UDP is not a popular opinion in
>> the IETF.  But as a network operator, and stakeholder in the internet
>> / ietf / webrtc, i am hoping to share this information so that you all
>> can make well informed protocol decisions.
>
>
> Thank you
>
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--f46d0434c0a09dc9b804f25ed11a
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<p dir=3D"ltr"><br>
On Feb 14, 2014 2:20 AM, &quot;Randell Jesup&quot; &lt;<a href=3D"mailto:ra=
ndell-ietf@jesup.org">randell-ietf@jesup.org</a>&gt; wrote:<br>
&gt;<br>
&gt; On 2/12/2014 10:06 PM, Cb B wrote:<br>
&gt;&gt;<br>
&gt;&gt; For about a year now, i have been very concerned about IPv4 UDP. =
=A0It<br>
&gt;&gt; has been increasingly associated with DDoS traffic [1], to the poi=
nt<br>
&gt;&gt; that the protocol&#39;s reputation is akin to IPv4 PING in 2003 (w=
idely<br>
&gt;&gt; blocked). =A0The material point here is that IPv4 UDP is being<br>
&gt;&gt; indiscriminately blocked and policed in access networks. =A0This i=
s not<br>
&gt;&gt; something access networks are wantonly electing to do. This decisi=
on<br>
&gt;&gt; is made in the heat of the moment to restore service to customers =
and<br>
&gt;&gt; networks who are overrun with attack traffic [2]. =A0And it is not=
<br>
&gt;&gt; simply DNS, NTP, and chargen for simple filtering.<br>
&gt;&gt;<br>
&gt;&gt; My suggestion is that draft-ietf-rtcweb-transports be updated to<b=
r>
&gt;&gt; include SCTP as a transport type option. =A0I understand that SCTP=
 is<br>
&gt;&gt; not supported on Windows (tm) and most NATs.<br>
&gt;<br>
&gt;<br>
&gt; Side note: it would be ironic to end up using DataChannel-over-SCTP-ov=
er-DTLS-over-SCTP<br>
&gt;</p>
<p dir=3D"ltr">The history of p2p applications such as skype have shown his=
tory favors the nimble.</p>
<p dir=3D"ltr">&gt;<br>
&gt;&gt; In general, we may find<br>
&gt;&gt; that TURN / TRAM is a required local trust anchor in access networ=
ks,<br>
&gt;&gt; just as local ISP-administered SMTP servers in access networks are=
<br>
&gt;&gt; frequently required relay points since access networks block SMTP.=
 [3]<br>
&gt;<br>
&gt;<br>
&gt; I certainly understand the reason for concern; but the idea that UDP w=
ould become limited to TURN-mediated is frightening and depressing (and who=
se TURN and how does the ISP know it&#39;s TURN-mediated unless it&#39;s an=
 ISP&#39;s TURN?)<br>

&gt;<br>
&gt; One significant piece of the value proposition of WebRTC is that throu=
gh use of ICE (and TURN) it will usually bypass the need for relays, signif=
icantly improving media quality (delay), dramatically cutting the cost of o=
peration for services, easing deployment, and avoiding a number of levels o=
f control/ability-to-block/etc.<br>

&gt;<br>
&gt; It&#39;s especially depressing in that we put significant effort into =
reducing the likelihood that WebRTC could be used for DDoS attacks.<br>
&gt;<br>
&gt; I will note that blocking UDP (or massively-rate-limiting it) will hav=
e all sorts of nasty effects on all forms of VoIP. =A0TCP-entrained VoIP ca=
n evade that, but at a serious cost to call quality. =A0Surely the operator=
s know this.<br>

&gt;</p>
<p dir=3D"ltr">Agreed on all points. My view is one related to the basic re=
quirement of keeping the network up.=A0 I hope i have provided enough refer=
ence points to make the magnitude of the problem clear as well as how histo=
ry has shown protocols get blocked (smtp)</p>

<p dir=3D"ltr">&gt;<br>
&gt;&gt; This UDP concern is not an issue for IPv6, since IPv6 is not, as o=
f<br>
&gt;&gt; yet, been associated with these massive attacks. =A0And when i say=
<br>
&gt;&gt; massive, i mean massive [4]. =A0And because of the scale and risk,=
<br>
&gt;&gt; access networks are simply clamping IPv4 UDP throughput and moving=
 on<br>
&gt;&gt; and not looking back.<br>
&gt;<br>
&gt;<br>
&gt; Can the DNS amplification attack be used to hit any ports other than 5=
3? =A0Can clamps be limited to port 53<br>
&gt;</p>
<p dir=3D"ltr">Right. The source port will be a function of the amp server =
such as 53, 123, 19... all very common. The destination port is of the spoo=
fers choosing. </p>
<p dir=3D"ltr">But, as the us-cert advisory notes, there are several impact=
ed appilication ports. Well-known webrtc fixed ports would help for a white=
 list. </p>
<p dir=3D"ltr">Creating a blacklist of just 53, 123, 19 makes us still behi=
nd the curve on whatever is next. </p>
<p dir=3D"ltr">CB<br></p>
<p dir=3D"ltr">&gt;<br>
&gt;&gt; I =A0understand predicting the demise of UDP is not a popular opin=
ion in<br>
&gt;&gt; the IETF. =A0But as a network operator, and stakeholder in the int=
ernet<br>
&gt;&gt; / ietf / webrtc, i am hoping to share this information so that you=
 all<br>
&gt;&gt; can make well informed protocol decisions.<br>
&gt;<br>
&gt;<br>
&gt; Thank you<br>
&gt;<br>
&gt;<br>
&gt; -- <br>
&gt; Randell Jesup<br>
&gt; <a href=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a><b=
r>
&gt;<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">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
=A0 </p>

--f46d0434c0a09dc9b804f25ed11a--


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On 14/02/2014 9:42 AM, Cb B wrote:
> Right. The source port will be a function of the amp server such as 
> 53, 123, 19... all very common. The destination port is of the 
> spoofers choosing.
>
> But, as the us-cert advisory notes, there are several impacted 
> appilication ports. Well-known webrtc fixed ports would help for a 
> white list.
>
> Creating a blacklist of just 53, 123, 19 makes us still behind the 
> curve on whatever is next.
>

Which is why we will never be able to stop DDoS attacks:

 1. With dumb firewall rules that rely upon port numbers alone.
 2. Unless the destination address is able to communicate back to the
    source (or as close as possible to the source) that it does not wish
    to continue receiving (any) packets from it, and in so doing pushes
    the problem back as far as possible to the source.

Gili


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    <div class="moz-cite-prefix">On 14/02/2014 9:42 AM, Cb B wrote:<br>
    </div>
    <blockquote
cite="mid:CAD6AjGRSVHTK7apQ1x3j0pE=dkeFeXBKc0U3z4GkCTywVvckTA@mail.gmail.com"
      type="cite">Right. The source port will be a function of the amp
      server such as 53, 123, 19... all very common. The destination
      port is of the spoofers choosing.
      <p dir="ltr">But, as the us-cert advisory notes, there are several
        impacted appilication ports. Well-known webrtc fixed ports would
        help for a white list. </p>
      <p dir="ltr">Creating a blacklist of just 53, 123, 19 makes us
        still behind the curve on whatever is next. </p>
    </blockquote>
    <br>
    Which is why we will never be able to stop DDoS attacks:<br>
    <ol>
      <li>With dumb firewall rules that rely upon port numbers alone.</li>
      <li>Unless the destination address is able to communicate back to
        the source (or as close as possible to the source) that it does
        not wish to continue receiving (any) packets from it, and in so
        doing pushes the problem back as far as possible to the source.</li>
    </ol>
    <p>Gili<br>
    </p>
  </body>
</html>

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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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I created a proposal for how one could signal the receiver's desires for
handling of a MediaStream as an extension to the MSID proposal.

The result is in draft-ietf-mmusic-msid-04 section 5.

There will be discussion in MMUSIC; comments on whether the mechanism
fits the RTCWEB need may be appropriate here.

         Harald

-------- Original Message --------
Subject: 	[MMUSIC] msid-04 submitted
Date: 	Fri, 14 Feb 2014 08:10:54 +0100
From: 	Harald Alvestrand <harald@alvestrand.no>
To: 	mmusic <mmusic@ietf.org>



I have submitted a new version of the MSID draft, -04.

It tries to make it crystal clear that the msid-semantic:wms is only
intended for signalling MediaStreamTracks and MediaStreams.

It also adds a new functionality: msid-control, which can be used by a
receiver of MediaStreamTracks to signal the sender of a MediaStreamTrack
what the receiver wishes with regard to that MediaStreamTrack; this need
has been identified within the WebRTC and RTCWEB WGs.

It seemed logical to extend this draft with that functionality, since it
is closely allied functionality.

Comments are welcome.

          Harald

-- 
Surveillance is pervasive. Go Dark.

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




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    I created a proposal for how one could signal the receiver's desires
    for handling of a MediaStream as an extension to the MSID proposal.<br>
    <br>
    The result is in draft-ietf-mmusic-msid-04 section 5.<br>
    <br>
    There will be discussion in MMUSIC; comments on whether the
    mechanism fits the RTCWEB need may be appropriate here.<br>
    <div class="moz-forward-container"><br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<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>[MMUSIC] msid-04 submitted</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Fri, 14 Feb 2014 08:10:54 +0100</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>mmusic <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>I have submitted a new version of the MSID draft, -04.

It tries to make it crystal clear that the msid-semantic:wms is only
intended for signalling MediaStreamTracks and MediaStreams.

It also adds a new functionality: msid-control, which can be used by a
receiver of MediaStreamTracks to signal the sender of a MediaStreamTrack
what the receiver wishes with regard to that MediaStreamTrack; this need
has been identified within the WebRTC and RTCWEB WGs.

It seemed logical to extend this draft with that functionality, since it
is closely allied functionality.

Comments are welcome.

          Harald

-- 
Surveillance is pervasive. Go Dark.

_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
      <br>
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On 02/14/2014 03:42 PM, Cb B wrote:
>
> > It's especially depressing in that we put significant effort into
> reducing the likelihood that WebRTC could be used for DDoS attacks.
> >
> > I will note that blocking UDP (or massively-rate-limiting it) will
> have all sorts of nasty effects on all forms of VoIP.  TCP-entrained
> VoIP can evade that, but at a serious cost to call quality.  Surely
> the operators know this.
> >
>
> Agreed on all points. My view is one related to the basic requirement
> of keeping the network up.  I hope i have provided enough reference
> points to make the magnitude of the problem clear as well as how
> history has shown protocols get blocked (smtp)
>
This SMTP example doesn't match what I've seen happen.

SMTP is not blocked by any backbone service provider I know of.

Outgoing SMTP on port 25 is commonly blocked by firewalls that think
they don't have servers behind them (hotels are notorious in this
aspect). That's why the submit port is popular (and deployed with
authentication). The main concern isn't DDOS attacks, it's being blamed
for spam.

I've not seen any report of a DDOS attack that used port 25 for a
traffic-volume-based attack (although intentional or unintentional DDOS
attacks on mail servers are too common to care about).

Given that I still haven't seen a report that leads me to belive we'll
ever see a proposal that seriously proposes blocking all UDP traffic on
the Internet - I continue to disbelieve the premise, so it's not
surprising that I disagree with the conclusion.

-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Feb 14 10:42:04 2014
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Thread-Topic: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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Hi Harald,

When reading the text, my take is that this is more than controlling a MS a=
nd MST.

You are actaually OVERRIDING the rules for sending RTCP. The rules say that=
 RTCP is always sent, as long as the m- line is enabled (non-zero port valu=
e).

Regarding "msid-control: enable", I don't think this is needed at all. The =
DEFAULT is whatever is the value of the existing direction attributes.

Regarding "msid-control: disable", what is the difference compared to using=
 "sendonly"?

Regarding "msid-control: reject", what couldn't you use "msid-control: stop=
" instead? If the stream has never been enabled, doesn't stop and reject me=
an the same thing?

Regarding "msid-control: stop", the text says that one guarantees that the =
track will not be used again. But, I assume the m- line could still be used=
 for sending media, for another track (assuming we allow associating an m- =
line with another track, that is)?

Regards,

Christer
________________________________________
From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Harald Alvestrand [hara=
ld@alvestrand.no]
Sent: Friday, 14 February 2014 8:06 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Stream control: [MMUSIC] msid-04 submitted

I created a proposal for how one could signal the receiver's desires for ha=
ndling of a MediaStream as an extension to the MSID proposal.

The result is in draft-ietf-mmusic-msid-04 section 5.

There will be discussion in MMUSIC; comments on whether the mechanism fits =
the RTCWEB need may be appropriate here.

         Harald

-------- Original Message --------
Subject:        [MMUSIC] msid-04 submitted
Date:   Fri, 14 Feb 2014 08:10:54 +0100
From:   Harald Alvestrand <harald@alvestrand.no><mailto:harald@alvestrand.n=
o>
To:     mmusic <mmusic@ietf.org><mailto:mmusic@ietf.org>



I have submitted a new version of the MSID draft, -04.

It tries to make it crystal clear that the msid-semantic:wms is only
intended for signalling MediaStreamTracks and MediaStreams.

It also adds a new functionality: msid-control, which can be used by a
receiver of MediaStreamTracks to signal the sender of a MediaStreamTrack
what the receiver wishes with regard to that MediaStreamTrack; this need
has been identified within the WebRTC and RTCWEB WGs.

It seemed logical to extend this draft with that functionality, since it
is closely allied functionality.

Comments are welcome.

          Harald

--
Surveillance is pervasive. Go Dark.

_______________________________________________
mmusic mailing list
mmusic@ietf.org<mailto:mmusic@ietf.org>
https://www.ietf.org/mailman/listinfo/mmusic




From nobody Fri Feb 14 10:54:18 2014
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On Fri, Feb 14, 2014 at 10:24 AM, Harald Alvestrand
<harald@alvestrand.no> wrote:
> On 02/14/2014 03:42 PM, Cb B wrote:
>>
>> > It's especially depressing in that we put significant effort into
>> reducing the likelihood that WebRTC could be used for DDoS attacks.
>> >
>> > I will note that blocking UDP (or massively-rate-limiting it) will
>> have all sorts of nasty effects on all forms of VoIP.  TCP-entrained
>> VoIP can evade that, but at a serious cost to call quality.  Surely
>> the operators know this.
>> >
>>
>> Agreed on all points. My view is one related to the basic requirement
>> of keeping the network up.  I hope i have provided enough reference
>> points to make the magnitude of the problem clear as well as how
>> history has shown protocols get blocked (smtp)
>>
> This SMTP example doesn't match what I've seen happen.
>
> SMTP is not blocked by any backbone service provider I know of.
>
> Outgoing SMTP on port 25 is commonly blocked by firewalls that think
> they don't have servers behind them (hotels are notorious in this
> aspect). That's why the submit port is popular (and deployed with
> authentication). The main concern isn't DDOS attacks, it's being blamed
> for spam.
>

Spam is a type of attack.  It is an L7 attack as opposed to a volume attack.


The largest broadband ISP in the USA blocks port 25, i provided this
info in my first email.

http://customer.comcast.com/help-and-support/internet/email-port-25-no-longer-supported/

> I've not seen any report of a DDOS attack that used port 25 for a
> traffic-volume-based attack (although intentional or unintentional DDOS
> attacks on mail servers are too common to care about).
>
> Given that I still haven't seen a report that leads me to belive we'll
> ever see a proposal that seriously proposes blocking all UDP traffic on
> the Internet - I continue to disbelieve the premise, so it's not
> surprising that I disagree with the conclusion.
>

That's fine.  It is not my goal to block UDP or save WebRTC.  I am
just submitting information that i have and connecting the dots that
are in front of me.  The future may show that i am too pessimistic or
not enough.

CB

> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Fri Feb 14 11:52:27 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-12.txt
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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           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-12.txt
	Pages           : 41
	Date            : 2014-02-14

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-12

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-12


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb 14 12:44:58 2014
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On 02/14/2014 07:41 PM, Christer Holmberg wrote:
> Hi Harald,
>
> When reading the text, my take is that this is more than controlling a MS and MST.
>
> You are actaually OVERRIDING the rules for sending RTCP. The rules say that RTCP is always sent, as long as the m- line is enabled (non-zero port value).
Yep, I think I'm repeating that. The question is not if RTCP is sent,
it's if this SSID is included in the RTCP sender report or not - which
the rules for RTCP do *not* dictate.
>
> Regarding "msid-control: enable", I don't think this is needed at all. The DEFAULT is whatever is the value of the existing direction attributes.
I don't think it's needed either. I included it for completeness.
>
> Regarding "msid-control: disable", what is the difference compared to using "sendonly"?
>
> Regarding "msid-control: reject", what couldn't you use "msid-control: stop" instead? If the stream has never been enabled, doesn't stop and reject mean the same thing?
As I said in the draft, I wonder about that too. People have talked
about rejecting a track at the API level; if they come out with a
sensible proposal, this should not prevent them.
Personally, my API preference is a model where tracks occur, and if you
don't want them, you have to stop them; I don't see a need for "reject".

But others might.

>
> Regarding "msid-control: stop", the text says that one guarantees that the track will not be used again. But, I assume the m- line could still be used for sending media, for another track (assuming we allow associating an m- line with another track, that is)?

Do we allow that? I don't know, and that's a problem.

>
> Regards,
>
> Christer
> ________________________________________
> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Harald Alvestrand [harald@alvestrand.no]
> Sent: Friday, 14 February 2014 8:06 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
>
> I created a proposal for how one could signal the receiver's desires for handling of a MediaStream as an extension to the MSID proposal.
>
> The result is in draft-ietf-mmusic-msid-04 section 5.
>
> There will be discussion in MMUSIC; comments on whether the mechanism fits the RTCWEB need may be appropriate here.
>
>          Harald
>
> -------- Original Message --------
> Subject:        [MMUSIC] msid-04 submitted
> Date:   Fri, 14 Feb 2014 08:10:54 +0100
> From:   Harald Alvestrand <harald@alvestrand.no><mailto:harald@alvestrand.no>
> To:     mmusic <mmusic@ietf.org><mailto:mmusic@ietf.org>
>
>
>
> I have submitted a new version of the MSID draft, -04.
>
> It tries to make it crystal clear that the msid-semantic:wms is only
> intended for signalling MediaStreamTracks and MediaStreams.
>
> It also adds a new functionality: msid-control, which can be used by a
> receiver of MediaStreamTracks to signal the sender of a MediaStreamTrack
> what the receiver wishes with regard to that MediaStreamTrack; this need
> has been identified within the WebRTC and RTCWEB WGs.
>
> It seemed logical to extend this draft with that functionality, since it
> is closely allied functionality.
>
> Comments are welcome.
>
>           Harald
>
> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org<mailto:mmusic@ietf.org>
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>


-- 
Surveillance is pervasive. Go Dark.


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On 02/14/2014 07:54 PM, Cb B wrote:
> On Fri, Feb 14, 2014 at 10:24 AM, Harald Alvestrand
> <harald@alvestrand.no> wrote:
>> On 02/14/2014 03:42 PM, Cb B wrote:
>>>> It's especially depressing in that we put significant effort into
>>> reducing the likelihood that WebRTC could be used for DDoS attacks.
>>>> I will note that blocking UDP (or massively-rate-limiting it) will
>>> have all sorts of nasty effects on all forms of VoIP.  TCP-entrained
>>> VoIP can evade that, but at a serious cost to call quality.  Surely
>>> the operators know this.
>>> Agreed on all points. My view is one related to the basic requirement
>>> of keeping the network up.  I hope i have provided enough reference
>>> points to make the magnitude of the problem clear as well as how
>>> history has shown protocols get blocked (smtp)
>>>
>> This SMTP example doesn't match what I've seen happen.
>>
>> SMTP is not blocked by any backbone service provider I know of.
>>
>> Outgoing SMTP on port 25 is commonly blocked by firewalls that think
>> they don't have servers behind them (hotels are notorious in this
>> aspect). That's why the submit port is popular (and deployed with
>> authentication). The main concern isn't DDOS attacks, it's being blamed
>> for spam.
>>
> Spam is a type of attack.  It is an L7 attack as opposed to a volume attack.
>
>
> The largest broadband ISP in the USA blocks port 25, i provided this
> info in my first email.
>
> http://customer.comcast.com/help-and-support/internet/email-port-25-no-longer-supported/

I believe this is consistent with what I said. They block it *for
residential customers*.

That's why I said "by backbone providers" above.

If port 25 was blocked by backbone providers, we would not be having
this conversation in this medium; from your message:

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The occurences of the word "SMTP" in those header fields should tell you
something.

>
>> I've not seen any report of a DDOS attack that used port 25 for a
>> traffic-volume-based attack (although intentional or unintentional DDOS
>> attacks on mail servers are too common to care about).
>>
>> Given that I still haven't seen a report that leads me to belive we'll
>> ever see a proposal that seriously proposes blocking all UDP traffic on
>> the Internet - I continue to disbelieve the premise, so it's not
>> surprising that I disagree with the conclusion.
>>
> That's fine.  It is not my goal to block UDP or save WebRTC.  I am
> just submitting information that i have and connecting the dots that
> are in front of me.  The future may show that i am too pessimistic or
> not enough.
>
> CB
>
>> --
>> Surveillance is pervasive. Go Dark.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Feb 14 14:21:15 2014
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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           : Overview: Real Time Protocols for Brower-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-09.txt
	Pages           : 21
	Date            : 2014-02-14

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   The document will be publishd as an Applicability Statement - it does
   not itself specify any protocol, but specifies which other
   specifications RTCWEB compliant implementations are supposed to
   follow.

   This document is a work item of the RTCWEB working group.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview-09

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-09


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb 14 15:01:36 2014
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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 Security Architecture
        Author          : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-09.txt
	Pages           : 43
	Date            : 2014-02-14

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 WebRTC.


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-09

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http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-security-arch-09


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until the htmlized version and diff are available at tools.ietf.org.

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From nobody Fri Feb 14 15:29:22 2014
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--001a11336a2ea8356b04f2662cdd
Content-Type: text/plain; charset=ISO-8859-1

RTCWEB WG members may be interested in the following draft:

http://tools.ietf.org/html/draft-miers-tls-sas-00

This draft specifies a Short Authentication String feature for
TLS and DTLS which would allow a pair of communicating
to compare SAS values computed from the TLS/DTLS
handshake in order to detect active attack.

-Ekr

--001a11336a2ea8356b04f2662cdd
Content-Type: text/html; charset=ISO-8859-1

<div dir="ltr"><div>RTCWEB WG members may be interested in the following draft:</div><div><br></div><div><a href="http://tools.ietf.org/html/draft-miers-tls-sas-00">http://tools.ietf.org/html/draft-miers-tls-sas-00</a><br>

</div><div><br></div><div>This draft specifies a Short Authentication String feature for</div><div>TLS and DTLS which would allow a pair of communicating</div><div>to compare SAS values computed from the TLS/DTLS</div><div>

handshake in order to detect active attack.</div><div><br></div><div>-Ekr</div><div><br></div></div>

--001a11336a2ea8356b04f2662cdd--


From nobody Fri Feb 14 16:09:49 2014
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From: Adam Langley <agl@google.com>
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On Fri, Feb 14, 2014 at 6:28 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> RTCWEB WG members may be interested in the following draft:
>
> http://tools.ietf.org/html/draft-miers-tls-sas-00
>
> This draft specifies a Short Authentication String feature for
> TLS and DTLS which would allow a pair of communicating
> to compare SAS values computed from the TLS/DTLS
> handshake in order to detect active attack.

I think the draft should discuss why a channel binding doesn't work.
It's hinted at in the introduction but it's a subtle point because
they are otherwise very similar.

Given that I see M. Green as an author, should I assume that the
requirement for 64 bytes of randomness for a /short/ authentication
string is to ensure that users of Dual_EC DRBG have no problems
leaking the necessary entropy to a passive observer? :)


Cheers

AGL


From nobody Sat Feb 15 23:22:38 2014
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OK, I think the text may be confusing; I think we mean the same things. The
value that should be placed in the SDP is the protocol that identifies
"WebRTC Data Channel", not the protocol used in the specific data channel.


On Fri, Feb 14, 2014 at 6:26 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> Section 5.2.1 says:
>
>         "Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-
>         passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
>         "a=setup" lines MUST be included as mentioned above, along with an
>         "a=sctpmap" line referencing the SCTP port number and specifying
> the
>         application protocol indicated in [I-D.ietf-rtcweb-data-protocol]."
>
> The statement that the sctpmap attribute specifies the application
> protocol indicate in the DCP is not aligned with the sctp-sdp draft. The
> sctp-sdp draft specifies that the attribute specifies the application
> running on top of the SCTP association - which in this case is the WebRTC
> Data Channel. The protocol running on top of that data channel is then
> specified in the DCP.
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">OK, I think the text may be confusing; I think we mean the=
 same things. The value that should be placed in the SDP is the protocol th=
at identifies &quot;WebRTC Data Channel&quot;, not the protocol used in the=
 specific data channel.</div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Feb 1=
4, 2014 at 6:26 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mail=
to:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@eric=
sson.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">Hi,<br>
<br>
Section 5.2.1 says:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;Within the data m=3D section, the &quot;a=
=3Dmid&quot;, &quot;a=3Dice-ufrag&quot;, &quot;a=3Dice-<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 passwd&quot;, &quot;a=3Dice-options&quot;, &quo=
t;a=3Dcandidate&quot;, &quot;a=3Dfingerprint&quot;, and<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;a=3Dsetup&quot; lines MUST be included as=
 mentioned above, along with an<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;a=3Dsctpmap&quot; line referencing the SC=
TP port number and specifying the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 application protocol indicated in [I-D.ietf-rtc=
web-data-protocol].&quot;<br>
<br>
The statement that the sctpmap attribute specifies the application protocol=
 indicate in the DCP is not aligned with the sctp-sdp draft. The sctp-sdp d=
raft specifies that the attribute specifies the application running on top =
of the SCTP association - which in this case is the WebRTC Data Channel. Th=
e protocol running on top of that data channel is then specified in the DCP=
.<br>


<br>
Regards,<br>
<br>
Christer<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>

--001a1133e89000813a04f280e7a2--


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From: Tim Panton <tim@phonefromhere.com>
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On 14 Feb 2014, at 16:46, cowwoc <cowwoc@bbs.darktech.org> wrote:

> On 14/02/2014 9:42 AM, Cb B wrote:
>> Right. The source port will be a function of the amp server such as =
53, 123, 19... all very common. The destination port is of the spoofers =
choosing.
>> But, as the us-cert advisory notes, there are several impacted =
appilication ports. Well-known webrtc fixed ports would help for a white =
list.
>>=20
>> Creating a blacklist of just 53, 123, 19 makes us still behind the =
curve on whatever is next.
>>=20
>=20
> Which is why we will never be able to stop DDoS attacks:
> With dumb firewall rules that rely upon port numbers alone.
> Unless the destination address is able to communicate back to the =
source (or as close as possible to the source) that it does not wish to =
continue receiving (any) packets from it, and in so doing pushes the =
problem back as far as possible to the source.


I don't agree, when we set up an ISP in the '90s we had egress filter =
rules that dropped any packets with =46rom address that were not
routed as 'inside' our network.

At the time we were told to drop that (along with a pile of other =
security features which are slowly returning).=20
I see no reason why anyone would peer with an ISP that refused to do =
that.

(We now return to your #rtcweb program).

T.
=20
=20

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

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"><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 14 Feb 2014, at 16:46, cowwoc &lt;<a href="mailto:cowwoc@bbs.darktech.org">cowwoc@bbs.darktech.org</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
  
    <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
  
  <div text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 14/02/2014 9:42 AM, Cb B wrote:<br>
    </div>
    <blockquote cite="mid:CAD6AjGRSVHTK7apQ1x3j0pE=dkeFeXBKc0U3z4GkCTywVvckTA@mail.gmail.com" type="cite">Right. The source port will be a function of the amp
      server such as 53, 123, 19... all very common. The destination
      port is of the spoofers choosing.
      <p dir="ltr">But, as the us-cert advisory notes, there are several
        impacted appilication ports. Well-known webrtc fixed ports would
        help for a white list. </p><p dir="ltr">Creating a blacklist of just 53, 123, 19 makes us
        still behind the curve on whatever is next. </p>
    </blockquote>
    <br>
    Which is why we will never be able to stop DDoS attacks:<br>
    <ol>
      <li>With dumb firewall rules that rely upon port numbers alone.</li>
      <li>Unless the destination address is able to communicate back to
        the source (or as close as possible to the source) that it does
        not wish to continue receiving (any) packets from it, and in so
        doing pushes the problem back as far as possible to the source.</li>
    </ol></div></blockquote><div><br></div><div><br></div><div>I don't agree, when we set up an ISP in the '90s we had egress filter rules that dropped any packets with From address that were not</div><div>routed as 'inside' our network.</div><div><br></div><div>At the time we were told to drop that (along with a pile of other security features which are slowly returning).&nbsp;</div><div>I see no reason why anyone would peer with an ISP that refused to do that.</div><div><br></div><div>(We now return to your #rtcweb program).</div><div><br></div><div>T.</div><div>&nbsp;</div><div>&nbsp;</div><br><blockquote type="cite"><div text="#000000" bgcolor="#FFFFFF"><p>Gili<br>
    </p>
  </div>

_______________________________________________<br>rtcweb mailing list<br><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></blockquote></div><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div>Tim Panton - Web/VoIP consultant and implementor</div><div><a href="http://www.westhawk.co.uk">www.westhawk.co.uk</a></div><div><br></div></span><br class="Apple-interchange-newline">

</div>
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On 13 Feb 2014, at 23:50, cowwoc <cowwoc@bbs.darktech.org> wrote:

> On 13/02/2014 6:00 PM, Dave Taht wrote:
>> On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org> =
wrote:
>>> On 13/02/2014 5:46 PM, Dave Taht wrote:
>>>> The biggest downside, as I see it, is targeting advancements in the =
state
>>>> of
>>>> the art, at windows. 98% of the world or so run non-windows based =
cell
>>>> phones and tablets, and in terms of total users, probably outnumber
>>>> the windows contingent at this point.
>>>>=20
>>>> SCTP and MPTCP are quite feasible on android and IOS.
>>>=20
>>> It doesn't matter how many smartphones there are. What matters is =
how many
>>> of them will be used to do meaningful video chat. The screen real =
estate on
>>> these devices is way too small.
>> In my experience, everybody is using tablets and handheld devices for
>> video chat.
>> It is a natural extension of the usage of the device to extend it =
from
>> phone calls
>> to video calls.
>>=20
>> The lack of a working camera on most desktops is a hindrance, and the =
placement
>> of cameras on most laptops is not ideal.
>=20
> All laptops and tablets come with decent cameras. I will agree that =
WebRTC on tablets will be strong, but smartphones is really pushing it. =
Most of the time I've seen people engage in video chat it was between =
family members; far less for business use. And in those cases, I've seen =
people jump on tablets and laptops instead of having to deal with a =
tiny, underpowered smartphone for video. These are just personal =
observations, not science, so please take them with a grain of salt.
>=20
> Gili

Again, I disagree, I often use a smartphone for video, but only if I can =
airplay/chromecast the video to a suitable TV.


T.


Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] UDP transport problem
Thread-Index: AQHPKx/ReTPUYSIEFUimiJt3/lhYxJq3+JDm
Date: Sun, 16 Feb 2014 14:57:26 +0000
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References: <CAD6AjGRiQ1UF5n3JG9HPRQFM+TD54Xz-dpTn5u9bX+__BMfesQ@mail.gmail.com> <CABkgnnVbZp7yBvpY1ARuaBXS=TOipY=BhXzrd=h5DY-76oF9Pw@mail.gmail.com> <CAD6AjGSxS4jNRGotsE_no0XhewvDqcVZ+Kmx1aMW9qorqSKR+w@mail.gmail.com> <52FD2FA4.8040701@alvestrand.no> <CAD6AjGTbSJEV2cJj5QyLktyZPv8SJa7h-QHKVtdUXnF3K6xwHA@mail.gmail.com> <52FD46F4.7030804@bbs.darktech.org> <CAA93jw4_+xAVza-YDpPD80Fj749i=vgOSz7sAty_Zp4U2TuO6g@mail.gmail.com> <52FD4C82.8040300@bbs.darktech.org> <CAA93jw5gEUzQeF74o_tt5KgdqFiedXzT5G0WdARsdcRnVEe6EQ@mail.gmail.com> <52FD5A4E.8060604@bbs.darktech.org>, <CE142A3F-F105-4E9A-AF64-C42A052F448A@phonefromhere.com>
In-Reply-To: <CE142A3F-F105-4E9A-AF64-C42A052F448A@phonefromhere.com>
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Cc: "rtcweb@ietf.org >> rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] UDP transport problem
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Why would we assume anything about the devices this stuff will run on in th=
e future? Technology change tells me we'll all be wrong and underestimate w=
hat's next. Assume WebRTC on everything, including my smarty ring. :-)

Jeremy

> On Feb 16, 2014, at 9:03 AM, "Tim Panton" <tim@phonefromhere.com> wrote:
>=20
>=20
>> On 13 Feb 2014, at 23:50, cowwoc <cowwoc@bbs.darktech.org> wrote:
>>=20
>>> On 13/02/2014 6:00 PM, Dave Taht wrote:
>>>> On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org> wrot=
e:
>>>>> On 13/02/2014 5:46 PM, Dave Taht wrote:
>>>>> The biggest downside, as I see it, is targeting advancements in the s=
tate
>>>>> of
>>>>> the art, at windows. 98% of the world or so run non-windows based cel=
l
>>>>> phones and tablets, and in terms of total users, probably outnumber
>>>>> the windows contingent at this point.
>>>>>=20
>>>>> SCTP and MPTCP are quite feasible on android and IOS.
>>>>=20
>>>> It doesn't matter how many smartphones there are. What matters is how =
many
>>>> of them will be used to do meaningful video chat. The screen real esta=
te on
>>>> these devices is way too small.
>>> In my experience, everybody is using tablets and handheld devices for
>>> video chat.
>>> It is a natural extension of the usage of the device to extend it from
>>> phone calls
>>> to video calls.
>>>=20
>>> The lack of a working camera on most desktops is a hindrance, and the p=
lacement
>>> of cameras on most laptops is not ideal.
>>=20
>> All laptops and tablets come with decent cameras. I will agree that WebR=
TC on tablets will be strong, but smartphones is really pushing it. Most of=
 the time I've seen people engage in video chat it was between family membe=
rs; far less for business use. And in those cases, I've seen people jump on=
 tablets and laptops instead of having to deal with a tiny, underpowered sm=
artphone for video. These are just personal observations, not science, so p=
lease take them with a grain of salt.
>>=20
>> Gili
>=20
> Again, I disagree, I often use a smartphone for video, but only if I can =
airplay/chromecast the video to a suitable TV.
>=20
>=20
> T.
>=20
>=20
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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--047d7b3a8174c8733704f28be566
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Forgive me for adding in my own two cents, as I only recently joined the
list ( as digest even), and am still reading like mad and trying to come up
to speed on this huge, complex area....

But I do know something about UX and people behavior, so I must say I agree
with Jeremy, you can't really assume that people won't do something.  It is
more typical for people to use any communication device in any possible
way...even if it isn't ideal for the communication.  Other factors often
are more important, like being in a situation where a preferred device
isn't available (examples. A car.  A remote location. No wifi or
connections. Not wanting to carry a larger device.)

And beyond that, new tech emerges constantly, some fail, some fail but come
back later ( like tablets, wearables, etc) and some devices get new looks
and UX such that they suddenly catch on....ipod to iphone to ipad...and
such pervasive popularity moves them from gadget to personal only to
business use...which has happened with phones, the iphone inspired UX, and
tablets....Linux was another success that also accelerated into wide
acceptance when it became, not just more reliable, but also more *general*
user friendly....it made it easier for techies to convince non-tech finance
types to buy in....

So my 2cents, uninformed in many other ways, is to recommend assuming the
widest possible usage and scale, if possible.

I'll go back to lurking now.... ;)
CG

>
> ---------- Forwarded message ----------
> From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
> To: Tim Panton <tim@phonefromhere.com>
> Cc: "rtcweb@ietf.org >> rtcweb@ietf.org" <rtcweb@ietf.org>
> Date: Sun, 16 Feb 2014 14:57:26 +0000
> Subject: Re: [rtcweb] UDP transport problem
> Why would we assume anything about the devices this stuff will run on in
the future? Technology change tells me we'll all be wrong and underestimate
what's next. Assume WebRTC on everything, including my smarty ring. :-)
>
> Jeremy
>
> > On Feb 16, 2014, at 9:03 AM, "Tim Panton" <tim@phonefromhere.com> wrote:
> >
> >
> >> On 13 Feb 2014, at 23:50, cowwoc <cowwoc@bbs.darktech.org> wrote:
> >>
> >>> On 13/02/2014 6:00 PM, Dave Taht wrote:
> >>>> On Thu, Feb 13, 2014 at 5:51 PM, cowwoc <cowwoc@bbs.darktech.org>
wrote:
> >>>>> On 13/02/2014 5:46 PM, Dave Taht wrote:
> >>>>> The biggest downside, as I see it, is targeting advancements in the
state
> >>>>> of
> >>>>> the art, at windows. 98% of the world or so run non-windows based
cell
> >>>>> phones and tablets, and in terms of total users, probably outnumber
> >>>>> the windows contingent at this point.
> >>>>>
> >>>>> SCTP and MPTCP are quite feasible on android and IOS.
> >>>>
> >>>> It doesn't matter how many smartphones there are. What matters is
how many
> >>>> of them will be used to do meaningful video chat. The screen real
estate on
> >>>> these devices is way too small.
> >>> In my experience, everybody is using tablets and handheld devices for
> >>> video chat.
> >>> It is a natural extension of the usage of the device to extend it from
> >>> phone calls
> >>> to video calls.
> >>>
> >>> The lack of a working camera on most desktops is a hindrance, and the
placement
> >>> of cameras on most laptops is not ideal.
> >>
> >> All laptops and tablets come with decent cameras. I will agree that
WebRTC on tablets will be strong, but smartphones is really pushing it.
Most of the time I've seen people engage in video chat it was between
family members; far less for business use. And in those cases, I've seen
people jump on tablets and laptops instead of having to deal with a tiny,
underpowered smartphone for video. These are just personal observations,
not science, so please take them with a grain of salt.
> >>
> >> Gili
> >
> > Again, I disagree, I often use a smartphone for video, but only if I
can airplay/chromecast the video to a suitable TV.
> >
> >
> > T.
> >
> >
> > Tim Panton - Web/VoIP consultant and implementor
> > www.westhawk.co.uk
> >
> >
> >
> > _______________________________________________
> > 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
>

--047d7b3a8174c8733704f28be566
Content-Type: text/html; charset=ISO-8859-1
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<p><br>
Forgive me for adding in my own two cents, as I only recently joined the li=
st ( as digest even), and am still reading like mad and trying to come up t=
o speed on this huge, complex area....</p>
<p>But I do know something about UX and people behavior, so I must say I ag=
ree with Jeremy, you can&#39;t really assume that people won&#39;t do somet=
hing.=A0 It is more typical for people to use any communication device in a=
ny possible way...even if it isn&#39;t ideal for the communication.=A0 Othe=
r factors often are more important, like being in a situation where a prefe=
rred device isn&#39;t available (examples. A car.=A0 A remote location. No =
wifi or connections. Not wanting to carry a larger device.)</p>

<p>And beyond that, new tech emerges constantly, some fail, some fail but c=
ome back later ( like tablets, wearables, etc) and some devices get new loo=
ks and UX such that they suddenly catch on....ipod to iphone to ipad...and =
such pervasive popularity moves them from gadget to personal only to busine=
ss use...which has happened with phones, the iphone inspired UX, and tablet=
s....Linux was another success that also accelerated into wide acceptance w=
hen it became, not just more reliable, but also more *general* user friendl=
y....it made it easier for techies to convince non-tech finance types to bu=
y in....</p>

<p>So my 2cents, uninformed in many other ways, is to recommend assuming th=
e widest possible usage and scale, if possible.</p>
<p>I&#39;ll go back to lurking now.... ;)<br>
CG</p>
<p>&gt; <br>
&gt; ---------- Forwarded message ----------<br>
&gt; From:=A0&quot;Jeremy Laurenson (jlaurens)&quot; &lt;<a href=3D"mailto:=
jlaurens@cisco.com">jlaurens@cisco.com</a>&gt;<br>
&gt; To:=A0Tim Panton &lt;<a href=3D"mailto:tim@phonefromhere.com">tim@phon=
efromhere.com</a>&gt;<br>
&gt; Cc:=A0&quot;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> &gt=
;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a h=
ref=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
&gt; Date:=A0Sun, 16 Feb 2014 14:57:26 +0000<br>
&gt; Subject:=A0Re: [rtcweb] UDP transport problem<br>
&gt; Why would we assume anything about the devices this stuff will run on =
in the future? Technology change tells me we&#39;ll all be wrong and undere=
stimate what&#39;s next. Assume WebRTC on everything, including my smarty r=
ing. :-)<br>

&gt;<br>
&gt; Jeremy<br>
&gt;<br>
&gt; &gt; On Feb 16, 2014, at 9:03 AM, &quot;Tim Panton&quot; &lt;<a href=
=3D"mailto:tim@phonefromhere.com">tim@phonefromhere.com</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;&gt; On 13 Feb 2014, at 23:50, cowwoc &lt;<a href=3D"mailto:cowwoc=
@bbs.darktech.org">cowwoc@bbs.darktech.org</a>&gt; wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;&gt; On 13/02/2014 6:00 PM, Dave Taht wrote:<br>
&gt; &gt;&gt;&gt;&gt; On Thu, Feb 13, 2014 at 5:51 PM, cowwoc &lt;<a href=
=3D"mailto:cowwoc@bbs.darktech.org">cowwoc@bbs.darktech.org</a>&gt; wrote:<=
br>
&gt; &gt;&gt;&gt;&gt;&gt; On 13/02/2014 5:46 PM, Dave Taht wrote:<br>
&gt; &gt;&gt;&gt;&gt;&gt; The biggest downside, as I see it, is targeting a=
dvancements in the state<br>
&gt; &gt;&gt;&gt;&gt;&gt; of<br>
&gt; &gt;&gt;&gt;&gt;&gt; the art, at windows. 98% of the world or so run n=
on-windows based cell<br>
&gt; &gt;&gt;&gt;&gt;&gt; phones and tablets, and in terms of total users, =
probably outnumber<br>
&gt; &gt;&gt;&gt;&gt;&gt; the windows contingent at this point.<br>
&gt; &gt;&gt;&gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt;&gt;&gt; SCTP and MPTCP are quite feasible on android and =
IOS.<br>
&gt; &gt;&gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt;&gt; It doesn&#39;t matter how many smartphones there are.=
 What matters is how many<br>
&gt; &gt;&gt;&gt;&gt; of them will be used to do meaningful video chat. The=
 screen real estate on<br>
&gt; &gt;&gt;&gt;&gt; these devices is way too small.<br>
&gt; &gt;&gt;&gt; In my experience, everybody is using tablets and handheld=
 devices for<br>
&gt; &gt;&gt;&gt; video chat.<br>
&gt; &gt;&gt;&gt; It is a natural extension of the usage of the device to e=
xtend it from<br>
&gt; &gt;&gt;&gt; phone calls<br>
&gt; &gt;&gt;&gt; to video calls.<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; The lack of a working camera on most desktops is a hindra=
nce, and the placement<br>
&gt; &gt;&gt;&gt; of cameras on most laptops is not ideal.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; All laptops and tablets come with decent cameras. I will agre=
e that WebRTC on tablets will be strong, but smartphones is really pushing =
it. Most of the time I&#39;ve seen people engage in video chat it was betwe=
en family members; far less for business use. And in those cases, I&#39;ve =
seen people jump on tablets and laptops instead of having to deal with a ti=
ny, underpowered smartphone for video. These are just personal observations=
, not science, so please take them with a grain of salt.<br>

&gt; &gt;&gt;<br>
&gt; &gt;&gt; Gili<br>
&gt; &gt;<br>
&gt; &gt; Again, I disagree, I often use a smartphone for video, but only i=
f I can airplay/chromecast the video to a suitable TV.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; T.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Tim Panton - Web/VoIP consultant and implementor<br>
&gt; &gt; <a href=3D"http://www.westhawk.co.uk">www.westhawk.co.uk</a><br>
&gt; &gt;<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">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<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">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
</p>

--047d7b3a8174c8733704f28be566--


From nobody Sun Feb 16 23:45:41 2014
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From: "Lijing (Jessie)" <lijing80@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "randell-ietf@jesup.org" <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt
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Hi,

One question about the implementation of data channel.

Section 4 says:
   Req. 4:   The application SHOULD be able to provide guidance as to
             the relative priority of each data channel relative to each
             other, and relative to the media streams.  This will
             interact with the congestion control algorithms.

It seems that for now there is no W3C API parameters to set the priority of=
 data channel. I wonder if the priority of the data channel isn't open to t=
he application and is decided by the browser.=20
So I want to know how the application be able to provide guidance as to the=
 priority.

Best Regards,

Jessie


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WG,

In two weeks we will meet in London. I hope everyone prepares well by
reviewing the new drafts. As you might have seen on:

https://datatracker.ietf.org/wg/rtcweb/

Most of the WG documents where updated the last week. We are going to
discuss many of them in the WG session. See draft agenda:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11431.html

That discussion will be much more productive if YOU have read the latest
version. You can also greatly help the WG and the Authors by sending
your review comments to mailing list. That way we can ensure that we
discuss the most relevant open issues. It will also will greatly help
improving the documents. So PLEASE post your reviews!

We will not meet our goals of finishing all our core documents this year
without your help!

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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WG,

This version is the authors attempt to make a version ready for WG last
call. We list below the changes that we consider significant beyond
editorial improvements.

- Further clarified that draft-ietf-rtcweb-security-arch defines the
crypto suits and key-management solution for the SAVPF profile.

- Removed the discussion of the SHIM based approach (draft-westerlund-
  avtcore-transport-multiplexing)

- Changed SLI and RPSI to RECOMMENDED to implement level based on WG
  mailing list discussion.

- Clarified that the RTP packet stream association with
  MediaStramStracks is the one that MSID provides.

- Removed Simulcast as open issue. We have removed the discussion of
explicitly signaled simulcast and clarified the one that can be done
using multiple mediaStreamTracks based on the same media source. These
can be included in the same or different PeerConnections with different
parameters for basic simulcast support. I still expect that when
simulcast in RTP and SDP has been specified that adoption of that is
considered for WebRTC.

- Added an WebRTC API requirement regarding CSRC list handling.

I recommend that people review these changes and provide feedback.

The authors will continue to strive for a WG last call directly after WG
meeting in London.

Cheers

Magnus


On 2014-02-14 20:51, 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           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
>         Authors         : Colin Perkins
>                           Magnus Westerlund
>                           Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-12.txt
> 	Pages           : 41
> 	Date            : 2014-02-14
> 
> 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-12
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-12
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> 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
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Mon Feb 17 03:30:34 2014
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From: "Liuyan (Scarlett)" <scarlett.liuyan@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00.txt - the data channel's state and the SCTP association
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Hi,

Section 5.5.2 does say:

	"o  As described in Section 4.2.2, if the SCTP association is not yet
      established, then the newly created data channels are in the
      Connecting state, else if the SCTP association is already
      established, then the newly created data channels are in the Open
      state."
  .....
  It is not clear that if the SCTP association is failed to establish, how =
to we deal with the newly created data channels.
  Furthermore, how long the valid duration of the connecting state of the n=
ewly created data channels are before the SCTP association has not yet esta=
blished ?=20

Section 5.5.3 does say:
       "5.2.3.  Closing a data channel

      When the application requests the closing of a data channel that was
      externally negotiated, the browser always performs an in-band SSN
      reset for this channel.

      It is specific to the sub-protocol whether this closing must in
      addition be signaled to the peer via a new SDP offer/answer exchange.

      The application must also close any data channel that was externally
      negotiated, for which the stream identifiers are not listed in an
      incoming SDP offer."
   ...
   When there are multiple data channels with different sub-protocol per a =
SCTP association,=20
Maybe it needs to identify how to close a single data channel, how to close=
 all data channels with the same sub-protocol and how to close a SCTP assoc=
iation.
Of course, closing a SCTP association can just set SCTP port number zero.:-=
)
 =20
Regards,
Scarlett
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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Hi,
(as individual)

I just reviewed the -05 of the audio draft and realized that it removed
all discussion of what packetization times are expected to be supported
by an implementation. For opus this is not that difficult as the range
is limited to multiples of the audio frames it can produce.

The current edit I think comes from Jean-Marc and Cullen's private
discussion who's outcome was communicated to the list on the 2014-01-31.
The main part of the message reads:

> We keep the part about what happens with RTP in
> draft-ietf-rtcweb-audio but move the parts about SDP off to JSEP. I
> think that means all we need here is basically MUST implement G.711 &
> Opus along with their RTP payload formats.
> 
> The ranges of size of packets, frames  and other things seem to be
> adequately covered by the specs for the codecs and WebRTC is not
> chaining theses codecs so seems good enough. The JSEP draft that is
> pointing at all the parts of SDP that need to be supported can deal
> with the ptime and maxptime in SDP.

I didn't get to comment this immediately as I went on vacation. But here
is my follow up on this thread. If you don't want to read what the
existing specs says and background motivations, jump to the end and read
from "Trying to conclude:"

First of all lets investigate what the two specs says about
packetization time.

Opus:
http://tools.ietf.org/id/draft-ietf-payload-rtp-opus-01.txt

4.2.  Payload Structure

   The Opus encoder can be set to output encoded frames representing
   2.5, 5, 10, 20, 40, or 60 ms of speech or audio data.  Further, an
   arbitrary number of frames can be combined into a packet.  The
   maximum packet length is limited to the amount of encoded data
   representing 120 ms of speech or audio data.

Section 6.1

   maxptime:  the decoder's maximum length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet that can be encapsulated in a received packet
      according to Section 6 of [RFC4566].  Possible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4.  If no value is specified, 120 is
      assumed as default.  This value is a recommendation by the
      decoding side to ensure the best performance for the decoder.  The
      decoder MUST be capable of accepting any allowed packet sizes to
      ensure maximum compatibility.

   ptime:  the decoder's recommended length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet according to Section 6 of [RFC4566].  Possible values
      are 3, 5, 10, 20, 40, or 60 or an arbitrary multiple of Opus frame
      sizes rounded up to the next full integer value up to a maximum
      value of 120 as defined in Section 4.  If no value is specified,
      20 is assumed as default.  If ptime is greater than maxptime,
      ptime MUST be ignored.  This parameter MAY be changed during a
      session.  This value is a recommendation by the decoding side to
      ensure the best performance for the decoder.  The decoder MUST be
      capable of accepting any allowed packet sizes to ensure maximum
      compatibility.

   minptime:  the decoder's minimum length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet that SHOULD be encapsulated in a received packet
      according to Section 6 of [RFC4566].  Possible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4.  If no value is specified, 3 is
      assumed as default.  This value is a recommendation by the
      decoding side to ensure the best performance for the decoder.  The
      decoder MUST be capable to accept any allowed packet sizes to
      ensure maximum compatibility.


Thus, I agree for Opus this is well-defined. An receiver MUST support
any combination of frames that the encoder can produce up to a total of
120 ms. And it has well defined usage of ptime and maxptime and also
defines a min ptime.

Lets then look at G.711:

This is the whole PCMA and PCMU payload format definition in RFC3551:

4.5.14 PCMA and PCMU

   PCMA and PCMU are specified in ITU-T Recommendation G.711.  Audio
   data is encoded as eight bits per sample, after logarithmic scaling.
   PCMU denotes mu-law scaling, PCMA A-law scaling.  A detailed
   description is given by Jayant and Noll [15].  Each G.711 octet SHALL
   be octet-aligned in an RTP packet.  The sign bit of each G.711 octet
   SHALL correspond to the most significant bit of the octet in the RTP
   packet (i.e., assuming the G.711 samples are handled as octets on the
   host machine, the sign bit SHALL be the most significant bit of the
   octet as defined by the host machine format).  The 56 kb/s and 48
   kb/s modes of G.711 are not applicable to RTP, since PCMA and PCMU
   MUST always be transmitted as 8-bit samples.

   See Section 4.1 regarding silence suppression.

This doesn't say anything about the packetization. Fortunately Section
4.2 of RFC 3551 do talk about this, and as the RTP/SAVPF profile used by
WebRTC derives from RTP/AVP (RFC3551) this do apply.

4.2  Operating Recommendations

   The following recommendations are default operating parameters.
   Applications SHOULD be prepared to handle other values.  The ranges
   given are meant to give guidance to application writers, allowing a
   set of applications conforming to these guidelines to interoperate
   without additional negotiation.  These guidelines are not intended to
   restrict operating parameters for applications that can negotiate a
   set of interoperable parameters, e.g., through a conference control
   protocol.

   For packetized audio, the default packetization interval SHOULD have
   a duration of 20 ms or one frame, whichever is longer, unless
   otherwise noted in Table 1 (column "ms/packet").  The packetization
   interval determines the minimum end-to-end delay; longer packets
   introduce less header overhead but higher delay and make packet loss
   more noticeable.  For non-interactive applications such as lectures
   or for links with severe bandwidth constraints, a higher
   packetization delay MAY be used.  A receiver SHOULD accept packets
   representing between 0 and 200 ms of audio data.  (For framed audio
   encodings, a receiver SHOULD accept packets with a number of frames
   equal to 200 ms divided by the frame duration, rounded up.)  This
   restriction allows reasonable buffer sizing for the receiver.

As can see this recommends that one per default support 20 ms, and that
receivers are capable of handling up to 200 ms.

So, for PCMA and PCMU the picture are less clear, there are
recommendations, but no hard requirements. Also they are sample based
codecs and thus can produce payloads of any length (bytes and samples).

When it comes the signalling, we do have ptime and maxptime defined in
the base-spec of SDP [RFC4566]

      a=ptime:<packet time>

         This gives the length of time in milliseconds represented by
         the media in a packet.  This is probably only meaningful for
         audio data, but may be used with other media types if it makes
         sense.  It should not be necessary to know ptime to decode RTP
         or vat audio, and it is intended as a recommendation for the
         encoding/packetisation of audio.  It is a media-level
         attribute, and it is not dependent on charset.

      a=maxptime:<maximum packet time>

         This gives the maximum amount of media that can be encapsulated
         in each packet, expressed as time in milliseconds.  The time
         SHALL be calculated as the sum of the time the media present in
         the packet represents.  For frame-based codecs, the time SHOULD
         be an integer multiple of the frame size.  This attribute is
         probably only meaningful for audio data, but may be used with
         other media types if it makes sense.  It is a media-level
         attribute, and it is not dependent on charset.  Note that this
         attribute was introduced after RFC 2327, and non-updated
         implementations will ignore this attribute.

Thus, these can be used to provide a single recommended packetization
interval and an upper limit if supported.

The fact that ptime only can indicate a single rate becomes a potential
issue as you can't determine a remote peer preferences for other rates,
if an WebRTC endpoint likes to modify its rate due to congestion control
reasons. Changing the packetization rate is one of the tools that give a
most significant bit-rate change for audio, and it can even be applied
without changing the encoding rate, something crucial for doing any
bit-rate adaptation for G.711.

For your notes, JSEP does currently do not discuss packetization times
or the ptime or maxptime SDP parameter at all.

Trying to conclude:

I see an issue that we don't provide firmer requirements on what
packetization intervals that should be supported by a WebRTC receiver.

I would propose that we actually write into the audio draft in general
that a WebRTC endpoint SHALL support receiving audio RTP payloads that
contain up to 200 ms of audio if the RTP payload format supports it.

When it comes to sending I would also like to provide some minimal
requirements, these may need to be on codec basis, and I think it is
G.711 that is lacking here. Thus, I think an WebRTC endpoint SHALL be
capable of producing packetization times in the RTP payloads with the
following amount of time: 10, 20, 40, 60 ms.

I also think we should formalize the requirement to support the ptime
and maxptime signalling to maximize the possibility for interop with any
legacy systems.

I do see a need for the audio draft to discuss the potential issues here
that can affect interoperability.

Cheers

Magnus Westerlund
(As individual)

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From nobody Tue Feb 18 02:06:22 2014
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Changed milestone "Send Use Cases document
(draft-ietf-rtcweb-use-cases-and- requirements) to IESG for
publication as Informational", set due date to February 2014 from
October 2012.

Changed milestone "Complete Overview (and hold for dependency
resolution) (draft-ietf-rtcweb-overview)", set due date to February
2014 from October 2012.

Changed milestone "Send Security and Privacy Problem Statement
(draft-ietf- rtcweb-security) to IESG for publication as
Informational", set due date to March 2014 from October 2012.

Changed milestone "Send Media Transport (draft-ietf-rtcweb-rtp-usage)
to IESG for publication as Proposed Standard", set due date to April
2014 from January 2013.

Changed milestone "Audio Processing and Audio Codecs
(draft-ietf-rtcweb-audio) to IESG for publication as Proposed
Standard", set due date to April 2014 from January 2013.

Changed milestone "Send Data Stream Transport for non-media data
(draft-ietf- rtcweb-data-channel) to IESG for publication as Proposed
Standard", set due date to May 2014 from May 2103.

Changed milestone "Send Security Solution
(draft-ietf-rtcweb-security-arch) to IESG for publication as Proposed
Standard", set due date to June 2014 from January 2013.

Changed milestone "Send Signalling Negotiation and NAT Traversal
(draft-ietf- rtcweb-jsep) to IESG for publication as Proposed
Standard", set due date to September 2014 from January 2013.

Changed milestone "Send STUN Usage for Consent Freshness to IESG for
publication as proposed standard", set due date to September 2014 from
March 2014.

Changed milestone "Video Processing and Video Codecs
(draft-ietf-rtcweb-video) to IESG for publication as Proposed
StandardVideo Processing and Video Codecs (draft-ietf-rtcweb-video) to
IESG for publication as Proposed Standard", set due date to December
2014 from January 2013.

URL: http://datatracker.ietf.org/wg/rtcweb/charter/


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Deleted milestone "Send Quality of Service markings of RTCWeb packets
(draft- ietf-rtcweb-qos) to IESG for publication as Proposed
Standard".

URL: http://datatracker.ietf.org/wg/rtcweb/charter/


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WG,

As one of the consumers of what RMCAT WG will produce it is important
that you checks that they are considering valid requirements. Please
review and send notice that you have done so to the RMCAT WG mailing
list and provide any feedback you have on the document.

Cheers

Magnus Westerlund

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Hi,

we'd like to start the working group last call for =
draft-ietf-rmcat-cc-requirements-02. Please send your feedback to the =
list.

We're letting this LC run until March 12 (after the London meeting), so =
there will also be a chance to discuss any feedback there.

Lars

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From nobody Tue Feb 18 06:11:34 2014
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: "Liuyan (Scarlett)" <scarlett.liuyan@huawei.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00.txt - the data channel's state and the SCTP association
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Hi Scarlett,
Please note that this document has been replaced by
draft-ejzak-mmusic-data-channel-sdpneg-00 (this is noted in datatracker)
and that discussion of the topic should be directed to the mmusic list
(although there is some obvious overlap with the data channel discussion
regarding external negotiation).  I copied both groups for this email only
- please respond only to mmusic.

" It is not clear that if the SCTP association is failed to establish, how
to we deal with the newly created data channels.
  Furthermore, how long the valid duration of the connecting state of the
newly created data channels are before the SCTP association has not yet
established ?"

This issue is common to all forms of data channel negotiation and should be
addressed by the core documents rather than this one.  It appears that a
data channel can be in the connecting state indefinitely until the peer
connection is closed.

"When there are multiple data channels with different sub-protocol per a
SCTP association,
Maybe it needs to identify how to close a single data channel, how to close
all data channels with the same sub-protocol and how to close a SCTP
association.
Of course, closing a SCTP association can just set SCTP port number zero.:-)
"

The procedure describes how to close a single data channel.  We did not see
a need to have a special way to close all data channels with the same
sub-protocol since they can just be closed individually.  I don't see a
compelling use case for this, but perhaps you have one?  The document
should point out that data channels terminate with the SCTP association and
peer connection.  Sometimes we forget to state the most obvious things!

BR, Richard


On Mon, Feb 17, 2014 at 5:30 AM, Liuyan (Scarlett) <
scarlett.liuyan@huawei.com> wrote:

> Hi,
>
> Section 5.5.2 does say:
>
>         "o  As described in Section 4.2.2, if the SCTP association is not
> yet
>       established, then the newly created data channels are in the
>       Connecting state, else if the SCTP association is already
>       established, then the newly created data channels are in the Open
>       state."
>   .....
>   It is not clear that if the SCTP association is failed to establish, how
> to we deal with the newly created data channels.
>   Furthermore, how long the valid duration of the connecting state of the
> newly created data channels are before the SCTP association has not yet
> established ?
>
> Section 5.5.3 does say:
>        "5.2.3.  Closing a data channel
>
>       When the application requests the closing of a data channel that was
>       externally negotiated, the browser always performs an in-band SSN
>       reset for this channel.
>
>       It is specific to the sub-protocol whether this closing must in
>       addition be signaled to the peer via a new SDP offer/answer exchange.
>
>       The application must also close any data channel that was externally
>       negotiated, for which the stream identifiers are not listed in an
>       incoming SDP offer."
>    ...
>    When there are multiple data channels with different sub-protocol per a
> SCTP association,
> Maybe it needs to identify how to close a single data channel, how to
> close all data channels with the same sub-protocol and how to close a SCTP
> association.
> Of course, closing a SCTP association can just set SCTP port number
> zero.:-)
>
> Regards,
> Scarlett
> _______________________________________________
> 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">Hi Scarlett,<div>Please note that this document has been r=
eplaced by draft-ejzak-mmusic-data-channel-sdpneg-00 (this is noted in data=
tracker) and that discussion of the topic should be directed to the mmusic =
list (although there is some obvious overlap with the data channel discussi=
on regarding external negotiation). =A0I copied both groups for this email =
only - please respond only to mmusic.</div>
<div><br></div><div>&quot;<span style=3D"font-family:arial,sans-serif;font-=
size:13px">=A0</span><span style=3D"font-family:arial,sans-serif;font-size:=
13px">It is not clear that if the SCTP association is failed to establish, =
how to we deal with the newly created data channels.</span></div>
<div><span style=3D"font-family:arial,sans-serif;font-size:13px">=A0 Furthe=
rmore, how long the valid duration of the connecting state of the newly cre=
ated data channels are before the SCTP association has not yet established =
?</span>&quot;</div>
<div><br></div><div>This issue is common to all forms of data channel negot=
iation and should be addressed by the core documents rather than this one. =
=A0It appears that a data channel can be in the connecting state indefinite=
ly until the peer connection is closed.</div>
<div><br></div><div>&quot;<span style=3D"font-family:arial,sans-serif;font-=
size:13px">When there are multiple data channels with different sub-protoco=
l per a SCTP association,</span></div><span style=3D"font-family:arial,sans=
-serif;font-size:13px">Maybe it needs to identify how to close a single dat=
a channel, how to close all data channels with the same sub-protocol and ho=
w to close a SCTP association.</span><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<div><span style=3D"font-family:arial,sans-serif;font-size:13px">Of course,=
 closing a SCTP association can just set SCTP port number zero.:-)</span>&q=
uot;</div><div><br></div><div>The procedure describes how to close a single=
 data channel. =A0We did not see a need to have a special way to close all =
data channels with the same sub-protocol since they can just be closed indi=
vidually. =A0I don&#39;t see a compelling use case for this, but perhaps yo=
u have one? =A0The document should point out that data channels terminate w=
ith the SCTP association and peer connection. =A0Sometimes we forget to sta=
te the most obvious things!</div>
<div><br></div><div>BR, Richard=A0</div></div><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Mon, Feb 17, 2014 at 5:30 AM, Liuyan (S=
carlett) <span dir=3D"ltr">&lt;<a href=3D"mailto:scarlett.liuyan@huawei.com=
" target=3D"_blank">scarlett.liuyan@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">Hi,<br>
<br>
Section 5.5.2 does say:<br>
<br>
=A0 =A0 =A0 =A0 &quot;o =A0As described in Section 4.2.2, if the SCTP assoc=
iation is not yet<br>
=A0 =A0 =A0 established, then the newly created data channels are in the<br=
>
=A0 =A0 =A0 Connecting state, else if the SCTP association is already<br>
=A0 =A0 =A0 established, then the newly created data channels are in the Op=
en<br>
=A0 =A0 =A0 state.&quot;<br>
=A0 .....<br>
=A0 It is not clear that if the SCTP association is failed to establish, ho=
w to we deal with the newly created data channels.<br>
=A0 Furthermore, how long the valid duration of the connecting state of the=
 newly created data channels are before the SCTP association has not yet es=
tablished ?<br>
<br>
Section 5.5.3 does say:<br>
=A0 =A0 =A0 =A0&quot;5.2.3. =A0Closing a data channel<br>
<br>
=A0 =A0 =A0 When the application requests the closing of a data channel tha=
t was<br>
=A0 =A0 =A0 externally negotiated, the browser always performs an in-band S=
SN<br>
=A0 =A0 =A0 reset for this channel.<br>
<br>
=A0 =A0 =A0 It is specific to the sub-protocol whether this closing must in=
<br>
=A0 =A0 =A0 addition be signaled to the peer via a new SDP offer/answer exc=
hange.<br>
<br>
=A0 =A0 =A0 The application must also close any data channel that was exter=
nally<br>
=A0 =A0 =A0 negotiated, for which the stream identifiers are not listed in =
an<br>
=A0 =A0 =A0 incoming SDP offer.&quot;<br>
=A0 =A0...<br>
=A0 =A0When there are multiple data channels with different sub-protocol pe=
r a SCTP association,<br>
Maybe it needs to identify how to close a single data channel, how to close=
 all data channels with the same sub-protocol and how to close a SCTP assoc=
iation.<br>
Of course, closing a SCTP association can just set SCTP port number zero.:-=
)<br>
<br>
Regards,<br>
Scarlett<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>
_______________________________________________<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>

--089e013c6ae494471104f2aed8b7--


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Subject: [rtcweb] Milestones changed for rtcweb WG
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Changed milestone "Send Specification of Transport Protocols and their
NAT Traversal to IESG for publication as Proposed Standard", set state
to active from review, accepting new milestone.

Changed milestone "Send Overview (after dependencies are ready) to
IESG for publication as Applicability Statement", set state to active
from review, accepting new milestone.

URL: http://datatracker.ietf.org/wg/rtcweb/charter/


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--089e01294412f110ba04f2af576e
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Are there plans to discuss any of the documents that are dependencies in
the MMUSIC WG at this meeting?  As I'm recalling, the initial agenda for
the last interim had a number of MMUSIC WG documents, thus there was an
agenda change to also organize part of the meeting as an MMUSIC WG interim.


Thanks,
Mary.


On Thu, Feb 13, 2014 at 10:46 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> WG,
>
> We have received some feedback on the dates. After having done some
> checking we propose to shift the interim meeting one day earlier to
> reduce the overlap with the 3GPP meeting. Thus, the proposed dates for
> the interim is 19-21 of May (Mon-Wed).
>
> If you are willing to host, please contact the WG chairs.
>
> Cheers
>
> Magnus, Ted and Cullen
>
>
> On 2014-01-29 08:59, Magnus Westerlund wrote:
> > WG,
> >
> > The chairs of this WG and the W3C are considering an joint interim
> > meeting. The dates we chairs have arrived as feasible are May 20-22
> > (Tue-Thu). We are considering to split up the time between the two WGs
> > across all three days. So please take note of these dates in your
> > calendar now.
> >
> > The target region for this interim would be East Coast of North America=
,
> > but we have at this stage no host or set location. If you are willing t=
o
> > host, please contact the chairs.
> >
> > We haven't decided on this interim yet, and appreciate your input into
> > holding one. The goals of the interim would be to advanced the state of
> > the core documents we need to finish up. We expect that we will have
> > some document that may have WG last call issues needing face to face
> > time to speed up resolving them. We expect to have documents that are
> > under development, like JSEP that will have thorny issues to deal with.
> > Especially issues that interact with the API where we can benefit from
> > having the W3C WG also present to come to joint conclusions.
> >
> > Cheers
> >
> > Magnus Westerlund
> > (For the RTCWEB Chairs)
> >
> > ----------------------------------------------------------------------
> > Services, Media and Network features, Ericsson Research EAB/TXM
> > ----------------------------------------------------------------------
> > Ericsson AB                 | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e01294412f110ba04f2af576e
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<div dir=3D"ltr">Are there plans to discuss any of the documents that are d=
ependencies in the MMUSIC WG at this meeting? =A0As I&#39;m recalling, the =
initial agenda for the last interim had a number of MMUSIC WG documents, th=
us there was an agenda change to also organize part of the meeting as an MM=
USIC WG interim. =A0<div>
<br></div><div>Thanks,</div><div>Mary.=A0</div></div><div class=3D"gmail_ex=
tra"><br><br><div class=3D"gmail_quote">On Thu, Feb 13, 2014 at 10:46 AM, M=
agnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@=
ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</spa=
n> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">WG,<br>
<br>
We have received some feedback on the dates. After having done some<br>
checking we propose to shift the interim meeting one day earlier to<br>
reduce the overlap with the 3GPP meeting. Thus, the proposed dates for<br>
the interim is 19-21 of May (Mon-Wed).<br>
<br>
If you are willing to host, please contact the WG chairs.<br>
<br>
Cheers<br>
<br>
Magnus, Ted and Cullen<br>
<br>
<br>
On 2014-01-29 08:59, Magnus Westerlund wrote:<br>
&gt; WG,<br>
&gt;<br>
&gt; The chairs of this WG and the W3C are considering an joint interim<br>
&gt; meeting. The dates we chairs have arrived as feasible are May 20-22<br=
>
&gt; (Tue-Thu). We are considering to split up the time between the two WGs=
<br>
&gt; across all three days. So please take note of these dates in your<br>
&gt; calendar now.<br>
&gt;<br>
&gt; The target region for this interim would be East Coast of North Americ=
a,<br>
&gt; but we have at this stage no host or set location. If you are willing =
to<br>
&gt; host, please contact the chairs.<br>
&gt;<br>
&gt; We haven&#39;t decided on this interim yet, and appreciate your input =
into<br>
&gt; holding one. The goals of the interim would be to advanced the state o=
f<br>
&gt; the core documents we need to finish up. We expect that we will have<b=
r>
&gt; some document that may have WG last call issues needing face to face<b=
r>
&gt; time to speed up resolving them. We expect to have documents that are<=
br>
&gt; under development, like JSEP that will have thorny issues to deal with=
.<br>
&gt; Especially issues that interact with the API where we can benefit from=
<br>
&gt; having the W3C WG also present to come to joint conclusions.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt; (For the RTCWEB Chairs)<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Services, Media and Network features, Ericsson Research EAB/TXM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Phone =A0<a href=3D"tel:=
%2B46%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Mobile <a href=3D"te=
l:%2B46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
&gt; SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerl=
und@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<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>
&gt;<br>
&gt;<br>
<br>
<br>
--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Phone =A0<a href=3D"tel:%2B46=
%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Mobile <a href=3D"tel:%2B=
46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</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><br></div>

--089e01294412f110ba04f2af576e--


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On 2014-02-18 15:46, Mary Barnes wrote:
> Are there plans to discuss any of the documents that are dependencies in
> the MMUSIC WG at this meeting?  As I'm recalling, the initial agenda for
> the last interim had a number of MMUSIC WG documents, thus there was an
> agenda change to also organize part of the meeting as an MMUSIC WG
> interim.  

At this stage this is an RTCWEB/WEBRTC interim only. This WG part of the
time we intended to focus on resolving any open issues with the WG's
documents.

We have asked the MMUSIC chairs about the interim meeting, which at the
time we asked declined interest in a co-located interim meeting.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Tue Feb 18 07:34:42 2014
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Thanks!


On Tue, Feb 18, 2014 at 9:19 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-02-18 15:46, Mary Barnes wrote:
> > Are there plans to discuss any of the documents that are dependencies i=
n
> > the MMUSIC WG at this meeting?  As I'm recalling, the initial agenda fo=
r
> > the last interim had a number of MMUSIC WG documents, thus there was an
> > agenda change to also organize part of the meeting as an MMUSIC WG
> > interim.
>
> At this stage this is an RTCWEB/WEBRTC interim only. This WG part of the
> time we intended to focus on resolving any open issues with the WG's
> documents.
>
> We have asked the MMUSIC chairs about the interim meeting, which at the
> time we asked declined interest in a co-located interim meeting.


> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

--e89a8f64782d3b5b7504f2b002cf
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<div dir=3D"ltr">Thanks!<br><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">On Tue, Feb 18, 2014 at 9:19 AM, Magnus Westerlund <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"=
_blank">magnus.westerlund@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"">On 2014-02-18 15:46, Mary Ba=
rnes wrote:<br>
&gt; Are there plans to discuss any of the documents that are dependencies =
in<br>
&gt; the MMUSIC WG at this meeting? =A0As I&#39;m recalling, the initial ag=
enda for<br>
&gt; the last interim had a number of MMUSIC WG documents, thus there was a=
n<br>
&gt; agenda change to also organize part of the meeting as an MMUSIC WG<br>
&gt; interim.<br>
<br>
</div>At this stage this is an RTCWEB/WEBRTC interim only. This WG part of =
the<br>
time we intended to focus on resolving any open issues with the WG&#39;s<br=
>
documents.<br>
<br>
We have asked the MMUSIC chairs about the interim meeting, which at the<br>
time we asked declined interest in a co-located interim meeting.</blockquot=
e><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
<br>
Cheers<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Phone =A0<a href=3D"tel:%2B46=
%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Mobile <a href=3D"tel:%2B=
46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

--e89a8f64782d3b5b7504f2b002cf--


From nobody Tue Feb 18 14:58:13 2014
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Date: Tue, 18 Feb 2014 23:58:00 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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I may be a little simple-minded, but if we have a recommendation that
receivers MUST be able to receive all packetizations of G.711 and OPUS
up to 200 ms per packet, and that receivers should signal this by
sending "a=maxptime:200" in their SDP, what situations exist where
interoperability is not going to be achieved?

I mean - if a G.711 receiver is able to handle any packet size thrown at
it, why do we need to put requirements on the sender (apart from "don't
send bigger packets than the maxptime")? For interoperability with
non-WebRTC endpoints?

Of course, I do have a question, now that all the ptime and maxtime
definitions have been put on the table:

Is it true that ptime and maxptime only specify the SDP sender's
declaration of his desires for incoming packet streams? That is, that
there is nowhere specified what the sender intends to send? I think that
is true for ptime in RFC 3264 offer/answer, and would assume that the
same is true for maxptime.



From nobody Tue Feb 18 15:36:04 2014
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Subject: Re: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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One thing I did not add in my previous reply:

If an offerer sets the direction of an m-line to sendrecv, and the
answerer sets the direction to sendonly, the offerer will have no
indication of whether the answerer intended the suspension of traffic to
be temporary (msid-control: disable) or permanent (msid-control: reject
or msid-control: stop).

That was actually the main concern that drove the desire for a new
mechanism.
Once the new mechanism was in place, it seemed logical to be explicit
about the other possible actions too, rather than relying on the
sendrecv/sendonly/recvonly parameter.

          Harald


From nobody Tue Feb 18 23:27:38 2014
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From: Justin Uberti <juberti@google.com>
Date: Tue, 18 Feb 2014 23:27:09 -0800
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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On Fri, Feb 14, 2014 at 6:22 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> Section 4.1.1. defines, for the PeerConnection constructor, the
> possibility to specify BUNDLE policy.
>
> However, I am missing the possibility to indicate whether, within a BUNDLE
> group, the same port number shall be used or not.
>
> Section 5.2.1 does say:
>
>         "o  The port value is set to the port of the default ICE candidate
> for
>         this m= section; if this m= section is not being bundled into
>         another m= section, the port value MUST be unique."
>
> ...but, it doesn't talk about the case when the port IS being bundled into
> another m= section. Normally, in the initial Offer, the port value will be
> unique also in that case, but if the PeerConnection is created due to
> forking, and it is known that the remote peer supports BUNDLE, then
> identical port values can be used already in the initial offer.
>

If a m= section is being bundled into another m= section, then the port
number must be the same, as indicated in the BUNDLE WG draft.

I did not include a policy option to force use of the same port; the
closest thing would be to use a policy of "all", which will mark all lines
except the first as bundle-only and achieve basically the same result,
although it would require a second offer to update the bundle-only ports.
If this is a problem we could consider adding another policy value.

--089e01183eb4419ce804f2bd52a2
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Feb 14, 2014 at 6:22 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@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">Hi,<br>
<br>
Section 4.1.1. defines, for the PeerConnection constructor, the possibility=
 to specify BUNDLE policy.<br>
<br>
However, I am missing the possibility to indicate whether, within a BUNDLE =
group, the same port number shall be used or not.<br>
<br>
Section 5.2.1 does say:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;o =C2=A0The port value is set to the port=
 of the default ICE candidate for<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 this m=3D section; if this m=3D section is not =
being bundled into<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 another m=3D section, the port value MUST be un=
ique.&quot;<br>
<br>
...but, it doesn&#39;t talk about the case when the port IS being bundled i=
nto another m=3D section. Normally, in the initial Offer, the port value wi=
ll be unique also in that case, but if the PeerConnection is created due to=
 forking, and it is known that the remote peer supports BUNDLE, then identi=
cal port values can be used already in the initial offer.<br>

</blockquote><div><br></div><div>If a m=3D section is being bundled into an=
other m=3D section, then the port number must be the same, as indicated in =
the BUNDLE WG draft.=C2=A0<br></div><div><br></div><div>I did not include a=
 policy option to force use of the same port; the closest thing would be to=
 use a policy of &quot;all&quot;, which will mark all lines except the firs=
t as bundle-only and achieve basically the same result, although it would r=
equire a second offer to update the bundle-only ports. If this is a problem=
 we could consider adding another policy value.</div>

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

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To: Harald Alvestrand <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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On 2014-02-18 23:58, Harald Alvestrand wrote:
> I may be a little simple-minded, but if we have a recommendation that
> receivers MUST be able to receive all packetizations of G.711 and OPUS
> up to 200 ms per packet, and that receivers should signal this by
> sending "a=maxptime:200" in their SDP, what situations exist where
> interoperability is not going to be achieved?

Interoperability is going to be achieve in the direction towards this
endpoint. The question is if we achieve interoperability in the
direction from that endpoint.

> 
> I mean - if a G.711 receiver is able to handle any packet size thrown at
> it, why do we need to put requirements on the sender (apart from "don't
> send bigger packets than the maxptime")? For interoperability with
> non-WebRTC endpoints?

If the above endpoint would be to send 109 samples in each RTP payload
then I think more than one of the current PCMA receivers would fall on
its back. In an WebRTC to WebRTC case, interoperability would be
achieved. However, if you talk about legacy interoperability (across a
gateway) you want to avoid having to basically insert a jitter buffer
and reconstitute the RTP payloads into something that that system
supports. That is why I suggest that certain common sizes an WebRTC
implementation MUST be capable of producing.

> 
> Of course, I do have a question, now that all the ptime and maxtime
> definitions have been put on the table:
> 
> Is it true that ptime and maxptime only specify the SDP sender's
> declaration of his desires for incoming packet streams? That is, that
> there is nowhere specified what the sender intends to send? I think that
> is true for ptime in RFC 3264 offer/answer, and would assume that the
> same is true for maxptime.

Yes, there are no attribute for its intended send direction unless the
m= block is a sendonly/recvonly O/A exchange where the sendonly
direction can kind of indicate it is preference for what it want/will send.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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Subject: Re: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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Hi,

If the indication is =93permanent=94, rather than using a new msid-control =
attribute, could it be indicated by NOT including some information in the A=
nswer?

Regards,

Christer

Sent from Windows Mail

From: 'Harald Alvestrand'<mailto:harald@alvestrand.no>
Sent: =FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD:=FD35=FD =
=FDAM
To: Hans-Christer Holmberg<mailto:christer.holmberg@ericsson.com>, rtcweb@i=
etf.org<mailto:rtcweb@ietf.org>

One thing I did not add in my previous reply:

If an offerer sets the direction of an m-line to sendrecv, and the
answerer sets the direction to sendonly, the offerer will have no
indication of whether the answerer intended the suspension of traffic to
be temporary (msid-control: disable) or permanent (msid-control: reject
or msid-control: stop).

That was actually the main concern that drove the desire for a new
mechanism.
Once the new mechanism was in place, it seemed logical to be explicit
about the other possible actions too, rather than relying on the
sendrecv/sendonly/recvonly parameter.

          Harald


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i', 'Segoe UI', 'Meiryo', 'Microsoft YaHei UI', 'Microsoft JhengHei UI', 'M=
algun Gothic', 'sans-serif';font-size:12pt;">
<div>Hi,</div>
<div><br>
</div>
<div>If the indication is&nbsp;=93permanent=94, rather than using a new msi=
d-control attribute, could it be indicated by NOT including some informatio=
n in the Answer?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer<br>
</div>
<div data-signatureblock=3D"true">
<div><br>
</div>
<div>Sent from Windows Mail</div>
<div><br>
</div>
</div>
<div style=3D"padding-top: 5px; border-top-color: rgb(229, 229, 229); borde=
r-top-width: 1px; border-top-style: solid;">
<div><font face=3D" 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei UI', =
'Microsoft JhengHei UI', 'Malgun Gothic', 'sans-serif'" style=3D"line-heigh=
t: 15pt; letter-spacing: 0.02em; font-family: &quot;Calibri&quot;, &quot;Se=
goe UI&quot;, &quot;Meiryo&quot;, &quot;Microsoft YaHei UI&quot;, &quot;Mic=
rosoft JhengHei UI&quot;, &quot;Malgun Gothic&quot;, &quot;sans-serif&quot;=
; font-size: 12pt;"><b>From:</b>&nbsp;<a href=3D"mailto:harald@alvestrand.n=
o" target=3D"_parent">'Harald
 Alvestrand'</a><br>
<b>Sent:</b>&nbsp;=FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD=
:=FD35=FD =FDAM<br>
<b>To:</b>&nbsp;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D=
"_parent">Hans-Christer Holmberg</a>,
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_parent">rtcweb@ietf.org</a></=
font></div>
</div>
<div><br>
</div>
<div dir=3D"">
<div id=3D"readingPaneBodyContent">One thing I did not add in my previous r=
eply:<br>
<br>
If an offerer sets the direction of an m-line to sendrecv, and the<br>
answerer sets the direction to sendonly, the offerer will have no<br>
indication of whether the answerer intended the suspension of traffic to<br=
>
be temporary (msid-control: disable) or permanent (msid-control: reject<br>
or msid-control: stop).<br>
<br>
That was actually the main concern that drove the desire for a new<br>
mechanism.<br>
Once the new mechanism was in place, it seemed logical to be explicit<br>
about the other possible actions too, rather than relying on the<br>
sendrecv/sendonly/recvonly parameter.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
</div>
</div>
</div>
</body>
</html>

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Dear all,

We have submit a draft about authentication for WebRTC telephony =
terminal use-case.=20
Your attention and comments are welcome.



A new version of I-D, draft-cheng-rtcweb-teleauth-00.txt
has been successfully submitted by Minpeng Qi and posted to the
IETF repository.

Name:		draft-cheng-rtcweb-teleauth
Revision:	00
Title:		Security Authentication of WebRTC Communication Service =
for Telephony Terminal
Document date:	2014-02-08
Group:		Individual Submission
Pages:		12
URL:            =
http://www.ietf.org/id/draft-cheng-rtcweb-teleauth-00.txt
Status:         =
https://datatracker.ietf.org/doc/draft-cheng-rtcweb-teleauth
Htmlized:       =
http://tools.ietf.org/html/draft-cheng-rtcweb-teleauth-00


Abstract
   The WebRTC use-cases and related requirements are defined in
   [draft-ietf-rtcweb-use-cases-and-requirements] that contains browser =
to
   browser use-cases and browser-GW/server use-cases (e.g., telephony
   terminal). In the use-case of telephony terminal, it is necessary for
   telephony terminal to be able to attest his identity to the telephony
   operator. Unlike the current authentication specified in
   [draft-ietf-rtcweb-security] such as PKI based authentication and web
   based peer authentication WebRTC communication is directly controlled
   by the telephony operator, which poses new authentication methods,
   including re-using existence authentication mechanism of telephony
   operator and authentication by using web credentials. This document
   presents the security authentication of WebRTC communication for
   telephony terminal.

BRs,
Minpeng=



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Hi,

I have reviewed the Transports for RTCWEB
(draft-ietf-rtcweb-transports-02) and have the following comments.

1. Section 1.

 This document focuses on the data
   transport protocos that are used by conforming implementations.

Missing "l" in protocos.

2. Section 3.1:
   For both protocols, IPv4 and IPv6 support is assumed; applications
   MUST be able to utilize both IPv4 and IPv6 where available.

What "applications" are intended in this sentence. I get a feeling that
"applications" here could be replaced by "WebRTC implementations"


3. Section 3.1:
   For UDP, this specification assumes the ability to set the DSCP code
   point of the sockets opened on a per-packet basis, in order to
   achieve the prioritizations described in
   [I-D.dhesikan-tsvwg-rtcweb-qos] when multiple media types are
   multiplexed.  It does not assume that the DSCP codepoints will be
   honored, and does assume that they may be zeroed or changed, since
   this is a local configuration issue.

I wonder if this should be moved into 3.2 section instead. At least the
part in 3.1 should focus on the DSCP setting part on the transport flow.
The implications of it working or not should be moved to 3.2. A forward
pointer to that second can be used instead.

4. Order of Section 3.2 and 3.3.

I think the order should be swapped between 3.2. and 3.3. That way the
Multiplexing section can provide definitions of all the stream concepts
that exists in WebRTC and how they can be combined. More about this later.

5. Content of Section 3.2

This section needs to be expanded to start with a general discussion of
how QoS can be applied to get a benefit. But, I think one need to be
clear that it can't be assumed to be available in implementations or
WebRTC based applications.

It needs to also discuss flow-based QoS mechanism. What are the
implications if that is what is available and what choices does one have
to address this by configuring the multiplexing different. I think this
is the logical place to take the main discussion of that, as it can
address both RTP streams and the data channel together. Please consider
pointers to appropriate discussion in Section 12.1.3 in
draft-ietf-rtcweb-rtp-usage.

I do note that writing this section appropriately is difficult until the
content of draft-dhesikan-tsvwg-rtcweb-qos has firmed up.

6. Content of Section 3.3:

I think this section should focus on making clear what different types
of streams that exist in the WebRTC context, and what choices of
multiplexing that are possible. Basically ensure that the full set of
combinations are made clear.

A. All RTP packet streams and data channel over a single UDP flow.

B. All RTP packet streams over one UDP flow and the data channel over
another UDP flow.

C. The RTP packet streams over different UDP flows based on content type
and separate data channel.

D. Each RTP packet stream over its own UDP flow (RTP session) and the
data channel over its own UDP flow.

The above with the additional dimension, that each RTP sessions, RTCP
can be in its own UDP flow.

There is also a question if the data channel can be aggregated with only
one RTP session, where there are multiple ones configured.

The QoS related discussion of this can be moved to the QoS section to
keep that focused.

However, the discussion of the pro and cons with having one or more flow
is good and could even be expanded to list reasons such as legacy
interop over a gateway.

7. Section 3.3:
   RTCWEB implementations MUST support the ability to send and receive
   multiple SSRCs on the same transport, and MUST support the ability to
   send and receive multiple SSRCs on multiple simultaneous transports,
   including the ability to send and receive audio and video on the same
   transport.

I think this sentence is restating what is already required by Section
4.1 in draft-ietf-rtcweb-rtp-usage: If you want to enforce this, I
suggest to do that by reference to that text rather than using RFC 2119
normative statements.


8. Section 3.4:
   ICE [RFC5245] MUST be supported.  The implementation MUST be a full
   ICE implementation, not ICE-Lite.

   In order to deal with situations where both parties are behind NATs
   which perform endpoint-dependent mapping (as defined in [RFC5128]
   section 2.4), TURN [RFC5766] MUST be supported.

These paragraph implies STUN and TURN server configuration. Should that
configuration question be discussed with the appropriate pointers to the
API?

9. Section 3.4:
   In order to deal with firewalls that block all UDP traffic, TURN
   using TCP between the client and the server MUST be supported, and
   TURN using TLS between the client and the server MUST be supported.
   See [RFC5766] section 2.1 for details.

I think the following part: "TURN using TLS between" should be changed
over "TURN using TLS over TCP between" to make it clear that this is
using TLS/TCP rather than TLS/FOO.

10. Section 3.4:
   TURN TCP candidates [RFC6062] SHOULD be supported; this allows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)

My understanding of RFC6062 is that it allows one to establish a vanilla
TCP connection on the far side of the TURN server as well as connecting
that one to an almost vanilla TCP connection (just a initial TURN
ConnectionBind request) that A establish to the TURN server, i.e.
WebRTC-A -(Turn/TCP)-> TURN_Server -(TCP)-> WebRTC-B.

The end-result is that a relayed vanilla TCP connection is established
between the WebRTC endpoints (A and B). As the WebRTC transport so far
defined only works with datagrams, i.e. DTLS or RTP/RTCP packets they
can't be sent natively over a TCP byte stream. Thus, if this mode of
operation is to be supported a framing is required on this leg.

A possibility here could be RFC 4571 if one want to go this way.

11. Section 3.4:

   ICE-TCP candidates [RFC6544] MAY be supported; this may allow
   applications to communicate to peers with public IP addresses across
   UDP-blocking firewalls without using a TURN server.

Also this option would need to use a framing of the datagrams.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Hi,

Several comments on data channel related drafts.

Firstly, There are several "protocol" concepts appeared in different data c=
hannel related drafts, and it seems there is no clear statement about the r=
elationship between these concepts.

draft-ietf-rtcweb-data-protocol has two protocol concept:

1.  The protocol byte in data channel open message defined as the protocol =
for the channel.

2.  SCTP payload protocol identifier (PPID)


WebRTC W3C API has one protocol concept,

3.  Protocol defined as "Subprotocol name used for this channel"


Q1: is the W3C API parameter "protocol" be the same as the protocol byte in=
 the data channel open message?


In previous email discussion, Peter Thatcher said :

"The PPID isn't exposed to the application.  It's just used to

distinguish text/binary/control.    If an application uses different

data channels with different protocols, it should use the SID to distinguis=
h between them, not the PPID.  In fact, even if it wanted to use the PPID, =
it wouldn't be able to."



Q2: if the answer of Q1 is yes. Then an application can distinguish differe=
nt data channels with different protocols by the protocol byte in the data =
channel open message. Is my understanding right?

draft-ejzak-mmusic-data-channel-sdpneg provides a way to negotiate sub-prot=
ocol parameters out of band.

4.  Sub-protocol: The 'sub-protocol' parameter indicates which protocol the=
 client expects to exchange via the channel.

example: a=3Ddcmap:5000 stream=3D2;label=3D"channel 2"; subprotocol=3D"BFCP=
";max_retr=3D3

Q3: is this "subprotocol" be the same concept as the protocol byte in the d=
ata channel open message? And the W3C API "protocol" should be set as BFCP?

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Secondly, in draft-ietf-rtcweb-data-channel-07, section 8 defines the follo=
wing PPIDs
      +------------------------------------+-----------+-----------+
      | Value                              | SCTP PPID | Reference |
      +------------------------------------+-----------+-----------+
      | WebRTC String                      | 51        | [RFCXXXX] |
      | WebRTC Binary Partial (Deprecated) | 52        | [RFCXXXX] |
      | WebRTC Binary                      | 53        | [RFCXXXX] |
      | WebRTC String Partial (Deprecated) | 54        | [RFCXXXX] |
      +------------------------------------+-----------+-----------+

Q4: It is confusing what is the intention to define "WebRTC Binary Partial"=
 and "WebRTC String Partial". I guess these two PPIDs are provided to the a=
pplication in the case when application supports fragmentation of large mes=
sages. But, at the meantime, in this draft, section 6.6 also says: The usag=
e of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial" is depre=
cated. Why the usage is deprecated? Does this implies that SCTP layer is re=
sponsible for fragmentation of large messages, and it is unnecessary for th=
e upper application to implement the fragmentation of large messages?

Q5: Since PPID 52 and 54 are defined, then how can the browser know whether=
 the data sent from the upper application layer is a part of a whole messag=
e and the SCTP PPID should be signaled as Partial 52/54, or this is a whole=
 message and should be signaled as 51/54?

Q6: if one side wants to send two messages, and the application split the f=
irst big message to two packages, then the browser will send three SCTP pac=
kages whose PPID are 54(part of first message), 51(part of first message), =
51(the whole second message) respectively. When the receiver side receives =
these three SCTP packages, how can the receiver side know which two SCTP pa=
ckages can be assembled to get the first big message?

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

The last question, which I have asked in previous email and have not receiv=
ed any response until now, so I mention this question again.

in draft-ietf-rtcweb-data-channel-07, Section 4 says:

   Req. 4:   The application SHOULD be able to provide guidance as to

             the relative priority of each data channel relative to each

             other, and relative to the media streams.  This will

             interact with the congestion control algorithms.



And in draft-ietf-rtcweb-data-protocol-03, DATA_CHANNEL_OPEN Message is def=
ined as below
0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  Message Type |  Channel Type |            Priority           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Reliability Parameter                      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |         Label Length          |       Protocol Length         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   \                                                               /
   |                             Label                             |
   /                                                               \
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   \                                                               /
   |                            Protocol                           |
   /                                                               \

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Priority: 2 bytes (integer)
      The priority of the channel as described in
      [I-D.ietf-rtcweb-data-channel].  The higher the number, the lower
      the priority.



Q7: It seems that for now there is no W3C API parameters to set the priorit=
y of data channel. I wonder whether the priority of the data channel is ope=
n to the application or is only decided by the browser. In other words, I w=
ant to know how the application be able to provide guidance as to the prior=
ity.



Best Regards,



Jessie


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Several comments on data channe=
l related drafts.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Firstly, There are several &#82=
20;protocol&#8221; concepts appeared in different data channel related draf=
ts, and it seems there is no clear statement about the relationship between=
 these concepts.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US">draft-ietf-rtcweb-data-protocol has two protocol concept:<o:p></=
o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:18.0pt;text-indent:-18.0=
pt;mso-list:l0 level1 lfo2">
<![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-fa=
mily:&quot;Courier New&quot;;color:black"><span style=3D"mso-list:Ignore">1=
.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US">The protocol byte in da=
ta channel open message defined as the protocol for the channel.<o:p></o:p>=
</span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:18.0pt;text-indent:-18.0=
pt;mso-list:l0 level1 lfo2">
<![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-fa=
mily:&quot;Courier New&quot;;color:black"><span style=3D"mso-list:Ignore">2=
.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US">SCTP payload protocol i=
dentifier (PPID)<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:18.0pt;text-indent:0cm">=
<span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">WebRTC W3C API has one protocol=
 concept, <o:p>
</o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:18.0pt;text-indent:-18.0=
pt;mso-list:l0 level1 lfo2">
<![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-fa=
mily:&quot;Courier New&quot;;color:black"><span style=3D"mso-list:Ignore">3=
.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US">Protocol defined as &#8=
220;Subprotocol name used for this channel&#8221;<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:18.0pt;text-indent:0cm">=
<span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US">Q1: is the W3C API parameter &#8220;protocol&#8221; be the same =
as the protocol byte in the data channel open message?<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">In previous email discussion, P=
eter Thatcher said :<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:#1F497D">&#82=
20;</span><span lang=3D"EN-US">The PPID isn't exposed to the application.&n=
bsp; It's just used to<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">distinguish text/binary/cont=
rol.&nbsp;&nbsp;&nbsp; If an application uses different<o:p></o:p></span></=
p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">data channels with different=
 protocols, it should use the SID to distinguish between them, not the PPID=
.&nbsp; In fact, even if it wanted to use the PPID, it wouldn't be able to.=
</span><span lang=3D"EN-US" style=3D"color:#1F497D">&#8221;<o:p></o:p></spa=
n></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p=
>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Q2: if the answer of Q1 is y=
es. Then an application can distinguish different data channels with differ=
ent protocols by the protocol byte in the data channel open message. Is my =
understanding right?</span><span lang=3D"EN-US"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">draft-ejzak-mmusic-data-channel=
-sdpneg provides a way to negotiate sub-protocol parameters out of band.<o:=
p></o:p></span></p>
<p class=3D"MsoListParagraph" align=3D"left" style=3D"margin-left:18.0pt;te=
xt-align:left;text-indent:-18.0pt;mso-list:l0 level1 lfo2;text-autospace:no=
ne">
<![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-fa=
mily:&quot;Courier New&quot;;color:black"><span style=3D"mso-list:Ignore">4=
.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US">Sub-protocol: The 'sub-=
protocol' parameter indicates which protocol the client expects to exchange=
 via the channel.&nbsp;
</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Cou=
rier New&quot;;color:black"><o:p></o:p></span></p>
<p class=3D"MsoListParagraph" align=3D"left" style=3D"margin-left:18.0pt;te=
xt-align:left;text-autospace:none">
<span lang=3D"EN-US">example: a=3Ddcmap:5000 stream=3D2;label=3D&quot;chann=
el 2&quot;; subprotocol=3D&quot;BFCP&quot;;max_retr=3D3<o:p></o:p></span></=
p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US">Q3: is this &#8220;subprotocol&#8221; be the =
same concept as the protocol byte in the data channel open message? And the=
 W3C API &#8220;protocol&#8221; should be set as BFCP?<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Secondly, in draft-ietf-rtcweb-=
data-channel-07, section 8 defines the following PPIDs<o:p></o:p></span></p=
>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#43;---------=
---------------------------&#43;-----------&#43;-----------&#43;<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | Value&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; | SCTP PPID | Reference |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#43;---------=
---------------------------&#43;-----------&#43;-----------&#43;<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WebRTC Strin=
g&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | 51&nbsp; &nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;| [RFCXXXX] |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WebRTC Binar=
y Partial (Deprecated) | 52&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | [RF=
CXXXX] |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WebRTC Binar=
y&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | 53&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; | [RFCXXXX] |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WebRTC Strin=
g Partial (Deprecated) | 54&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | [RF=
CXXXX] |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;=
color:black">&#43;------------------------------------&#43;-----------&#43;=
-----------&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Q4: It is confusing what is the=
 intention to define &#8220;WebRTC Binary Partial&#8221; and &#8220;WebRTC =
String Partial&#8221;. I guess these two PPIDs are provided to the applicat=
ion in the case when application supports fragmentation of large
 messages. But, at the meantime, in this draft, section 6.6 also says: The =
usage of the PPIDs &quot;WebRTC String Partial&quot; and &quot;WebRTC Binar=
y Partial&quot; is deprecated. Why the usage is deprecated? Does this impli=
es that SCTP layer is responsible for fragmentation
 of large messages, and it is unnecessary for the upper application to impl=
ement the fragmentation of large messages?<span style=3D"color:#1F497D"><o:=
p></o:p></span></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Q5: Since PPID 52 and 54 are de=
fined, then how can the browser know whether the data sent from the upper a=
pplication layer is a part of a whole message and the SCTP PPID should be s=
ignaled as Partial 52/54, or this is
 a whole message and should be signaled as 51/54?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Q6: if one side wants to send t=
wo messages, and the application split the first big message to two package=
s, then the browser will send three SCTP packages whose PPID are 54(part of=
 first message), 51(part of first message),
 51(the whole second message) respectively. When the receiver side receives=
 these three SCTP packages, how can the receiver side know which two SCTP p=
ackages can be assembled to get the first big message?
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The last question, which I have=
 asked in previous email and have not received any response until now, so I=
 mention this question again.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">in draft-ietf-rtcweb-data-ch=
annel-07, Section 4 says:<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; Req. 4:&nbsp;&n=
bsp; The application SHOULD be able to provide guidance as to<o:p></o:p></s=
pan></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the relative priority of each =
data channel relative to each<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; other, and relative to the med=
ia streams.&nbsp; This will<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; interact with the congestion c=
ontrol algorithms.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p=
>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">And in draft-ietf-rtcweb-dat=
a-protocol-03, DATA_CHANNEL_OPEN Message is defined as below<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; 2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 =
3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;=
-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; |&nbsp; Message Type |&nbsp; Cha=
nnel Type |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; Priority&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o=
:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;=
-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; Reliability Parameter&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;=
-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; Label Length&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Protocol Length&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;=
-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; \&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;/<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Label&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; |<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; /&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; \<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;=
-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#=
43;-&#43;-&#43;<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; \&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Protocol&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o=
:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp; /&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; \<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Courier New&quot;;color:black">&nbsp;&nbsp; &#43;-&#43;-&#4=
3;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-=
&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#43;-&#4=
3;-&#43;-&#43;-&#43;-&#43;-&#43;</span><span lang=3D"EN-US"><o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">Priority: 2 bytes (integer)<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;;color:black">&nbsp;&nbsp;&nbsp;&nbsp;
</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Cou=
rier New&quot;">&nbsp;The priority of the channel as described in<o:p></o:p=
></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<u>[I-D.ietf-rtcweb-data-channel]</u>.&nbsp; The higher the number, the low=
er<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left;text-autospa=
ce:none"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;C=
ourier New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;"=
>the priority.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Q7: It seems that for now th=
ere is no W3C API parameters to set the priority of data channel. I wonder =
whether the priority of the data channel is open to the application or is o=
nly decided by the browser. In other
 words, I want to know how the application be able to provide guidance as t=
o the priority.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Best Regards,<o:p></o:p></sp=
an></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Jessie<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_A3045C90BB645147BC99159AA47ABAC7495EAD61SZXEMA510MBXchi_--


From nobody Wed Feb 19 10:12:01 2014
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Date: Wed, 19 Feb 2014 19:11:44 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, rtcweb@ietf.org
References: <530320F7.4090300@ericsson.com> <5303E578.3000207@alvestrand.no> <53046842.2010108@ericsson.com>
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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
> On 2014-02-18 23:58, Harald Alvestrand wrote:
>> I may be a little simple-minded, but if we have a recommendation that
>> receivers MUST be able to receive all packetizations of G.711 and OPUS
>> up to 200 ms per packet, and that receivers should signal this by
>> sending "a=maxptime:200" in their SDP, what situations exist where
>> interoperability is not going to be achieved?
> Interoperability is going to be achieve in the direction towards this
> endpoint. The question is if we achieve interoperability in the
> direction from that endpoint.

So, given that there is nothing in the G.711 specification about which
sample sizes a G.711 receiver has to support - the right approach seems
to be to amend the G.711 MIME registration with this information.

This is not a WebRTC issue; it is an absence of specification for the
non-WebRTC devices that use G.711. The RTCWEB specifications can
reasonably be expected to point to existing information about this
issue; it is not reasonable (in my mind) that the RTCWEB WG decide what
these values should be.


-- 
Surveillance is pervasive. Go Dark.


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Date: Wed, 19 Feb 2014 19:12:35 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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On 02/19/2014 09:19 AM, Christer Holmberg wrote:
> Hi,
>
> If the indication is “permanent”, rather than using a new msid-control
> attribute, could it be indicated by NOT including some information in
> the Answer?

Certainly, but then you have to include some information in the answer
when the indication is not permanent.

Which information were you thinking of utilizing as a signal?


>
> Regards,
>
> Christer
>
> Sent from Windows Mail
>
> *From:* 'Harald Alvestrand' <mailto:harald@alvestrand.no>
> *Sent:* ýWednesdayý, ýFebruaryý ý19ý, ý2014 ý1ý:ý35ý ýAM
> *To:* Hans-Christer Holmberg <mailto:christer.holmberg@ericsson.com>,
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>
> One thing I did not add in my previous reply:
>
> If an offerer sets the direction of an m-line to sendrecv, and the
> answerer sets the direction to sendonly, the offerer will have no
> indication of whether the answerer intended the suspension of traffic to
> be temporary (msid-control: disable) or permanent (msid-control: reject
> or msid-control: stop).
>
> That was actually the main concern that drove the desire for a new
> mechanism.
> Once the new mechanism was in place, it seemed logical to be explicit
> about the other possible actions too, rather than relying on the
> sendrecv/sendonly/recvonly parameter.
>
>           Harald
>


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 02/19/2014 09:19 AM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1D1A9B80@ESESSMB209.ericsson.se"
      type="cite">
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        <div>Hi,</div>
        <div><br>
        </div>
        <div>If the indication is “permanent”, rather than using a new
          msid-control attribute, could it be indicated by NOT including
          some information in the Answer?</div>
      </div>
    </blockquote>
    <br>
    Certainly, but then you have to include some information in the
    answer when the indication is not permanent.<br>
    <br>
    Which information were you thinking of utilizing as a signal?<br>
    <br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1D1A9B80@ESESSMB209.ericsson.se"
      type="cite">
      <div data-externalstyle="false" dir="ltr" style="font-family:
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        'Microsoft JhengHei UI', 'Malgun Gothic',
        'sans-serif';font-size:12pt;">
        <div><br>
        </div>
        <div>Regards,</div>
        <div><br>
        </div>
        <div>Christer<br>
        </div>
        <div data-signatureblock="true">
          <div><br>
          </div>
          <div>Sent from Windows Mail</div>
          <div><br>
          </div>
        </div>
        <div style="padding-top: 5px; border-top-color: rgb(229, 229,
          229); border-top-width: 1px; border-top-style: solid;">
          <div><font style="line-height: 15pt; letter-spacing: 0.02em;
              font-family: &quot;Calibri&quot;, &quot;Segoe UI&quot;,
              &quot;Meiryo&quot;, &quot;Microsoft YaHei UI&quot;,
              &quot;Microsoft JhengHei UI&quot;, &quot;Malgun
              Gothic&quot;, &quot;sans-serif&quot;; font-size: 12pt;"
              face=" 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei
              UI', 'Microsoft JhengHei UI', 'Malgun Gothic',
              'sans-serif'"><b>From:</b> <a moz-do-not-send="true"
                href="mailto:harald@alvestrand.no" target="_parent">'Harald

                Alvestrand'</a><br>
              <b>Sent:</b> ýWednesdayý, ýFebruaryý ý19ý, ý2014 ý1ý:ý35ý
              ýAM<br>
              <b>To:</b> <a moz-do-not-send="true"
                href="mailto:christer.holmberg@ericsson.com"
                target="_parent">Hans-Christer Holmberg</a>,
              <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org"
                target="_parent">rtcweb@ietf.org</a></font></div>
        </div>
        <div><br>
        </div>
        <div dir="">
          <div id="readingPaneBodyContent">One thing I did not add in my
            previous reply:<br>
            <br>
            If an offerer sets the direction of an m-line to sendrecv,
            and the<br>
            answerer sets the direction to sendonly, the offerer will
            have no<br>
            indication of whether the answerer intended the suspension
            of traffic to<br>
            be temporary (msid-control: disable) or permanent
            (msid-control: reject<br>
            or msid-control: stop).<br>
            <br>
            That was actually the main concern that drove the desire for
            a new<br>
            mechanism.<br>
            Once the new mechanism was in place, it seemed logical to be
            explicit<br>
            about the other possible actions too, rather than relying on
            the<br>
            sendrecv/sendonly/recvonly parameter.<br>
            <br>
                      Harald<br>
            <br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

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On 02/19/2014 11:08 AM, Magnus Westerlund wrote:
> Hi,
>
> I have reviewed the Transports for RTCWEB
> (draft-ietf-rtcweb-transports-02)
Thanks!
>  and have the following comments.
>
> 1. Section 1.
>
>  This document focuses on the data
>    transport protocos that are used by conforming implementations.
>
> Missing "l" in protocos.
Fixed.
>
> 2. Section 3.1:
>    For both protocols, IPv4 and IPv6 support is assumed; applications
>    MUST be able to utilize both IPv4 and IPv6 where available.
>
> What "applications" are intended in this sentence. I get a feeling that
> "applications" here could be replaced by "WebRTC implementations"

Javascript applications. I think I should add the word "application" to
-overview's vocabulary; it is actually used in that meaning in that
document (also in the forms "Web application" and "browser-based
application".

>
>
> 3. Section 3.1:
>    For UDP, this specification assumes the ability to set the DSCP code
>    point of the sockets opened on a per-packet basis, in order to
>    achieve the prioritizations described in
>    [I-D.dhesikan-tsvwg-rtcweb-qos] when multiple media types are
>    multiplexed.  It does not assume that the DSCP codepoints will be
>    honored, and does assume that they may be zeroed or changed, since
>    this is a local configuration issue.
>
> I wonder if this should be moved into 3.2 section instead. At least the
> part in 3.1 should focus on the DSCP setting part on the transport flow.
> The implications of it working or not should be moved to 3.2. A forward
> pointer to that second can be used instead.

I think it fits better in 3.1 - this is about the assumptions that the
specifcation makes about the parts of the system that it is not a
specification for.

>
> 4. Order of Section 3.2 and 3.3.
>
> I think the order should be swapped between 3.2. and 3.3. That way the
> Multiplexing section can provide definitions of all the stream concepts
> that exists in WebRTC and how they can be combined. More about this later.
>
> 5. Content of Section 3.2
>
> This section needs to be expanded to start with a general discussion of
> how QoS can be applied to get a benefit. But, I think one need to be
> clear that it can't be assumed to be available in implementations or
> WebRTC based applications.

I want to push back on this. It seems inappropriate that the RTCWEB QoS
specification start off with a tutorial on what QoS is and how one can
use it to gain a benefit. That material should be available elsewhere.
(Recommendations?)

>
> It needs to also discuss flow-based QoS mechanism. What are the
> implications if that is what is available and what choices does one have
> to address this by configuring the multiplexing different.

I do not want to discuss material for which there are no references.
What particular flow-based mechanisms do you have in mind, and what are
the relevant references?

>  I think this
> is the logical place to take the main discussion of that, as it can
> address both RTP streams and the data channel together. Please consider
> pointers to appropriate discussion in Section 12.1.3 in
> draft-ietf-rtcweb-rtp-usage.

I do not think this document should repeat any material from rtp-usage,
but rtp-usage does not consider any aspect of QoS marking for multiple
streams using the same 5-tuple, nor does it say anything about
flow-based or DPI-based mechanisms apart from noting their existence.

Agree that the multiplexing aspect should be discussed somewhere, but
I'm not sure where.
>
> I do note that writing this section appropriately is difficult until the
> content of draft-dhesikan-tsvwg-rtcweb-qos has firmed up.

True.
>
> 6. Content of Section 3.3:
>
> I think this section should focus on making clear what different types
> of streams that exist in the WebRTC context, and what choices of
> multiplexing that are possible. Basically ensure that the full set of
> combinations are made clear.

Isn't this material that really should be in
draft-ietf-avtcore-rtp-multi-stream or
draft-ietf-avtcore-multiplex-guidelines?

Again, this seems the wrong draft for having a discussion of those issues.
>
> A. All RTP packet streams and data channel over a single UDP flow.
>
> B. All RTP packet streams over one UDP flow and the data channel over
> another UDP flow.
>
> C. The RTP packet streams over different UDP flows based on content type
> and separate data channel.
>
> D. Each RTP packet stream over its own UDP flow (RTP session) and the
> data channel over its own UDP flow.
>
> The above with the additional dimension, that each RTP sessions, RTCP
> can be in its own UDP flow.
>
> There is also a question if the data channel can be aggregated with only
> one RTP session, where there are multiple ones configured.
>
> The QoS related discussion of this can be moved to the QoS section to
> keep that focused.
>
> However, the discussion of the pro and cons with having one or more flow
> is good and could even be expanded to list reasons such as legacy
> interop over a gateway.
>
> 7. Section 3.3:
>    RTCWEB implementations MUST support the ability to send and receive
>    multiple SSRCs on the same transport, and MUST support the ability to
>    send and receive multiple SSRCs on multiple simultaneous transports,
>    including the ability to send and receive audio and video on the same
>    transport.
>
> I think this sentence is restating what is already required by Section
> 4.1 in draft-ietf-rtcweb-rtp-usage: If you want to enforce this, I
> suggest to do that by reference to that text rather than using RFC 2119
> normative statements.

Yes, rtp-usage contains these requirements. I can delete it from here.
Thanks!

However, this removes the handle on which I hung the discussion of why
one would want to multiplex or not, which I inserted because you
requested it. This discussion also seems to be present in rtp-usage, so
it seems appropriate to delete that too.

>
>
> 8. Section 3.4:
>    ICE [RFC5245] MUST be supported.  The implementation MUST be a full
>    ICE implementation, not ICE-Lite.
>
>    In order to deal with situations where both parties are behind NATs
>    which perform endpoint-dependent mapping (as defined in [RFC5128]
>    section 2.4), TURN [RFC5766] MUST be supported.
>
> These paragraph implies STUN and TURN server configuration. Should that
> configuration question be discussed with the appropriate pointers to the
> API?
This document has no pointers to the API at the moment, and I do not
want to add them - it's a layering violation of sorts. We can add
pointers to JSEP, which is kind of close to the API.

But the main point seems to be that STUN and TURN servers MUST be
configurable, both from the browser setup and from the (JS) application.
I'm adding a sentence saying that.

>
> 9. Section 3.4:
>    In order to deal with firewalls that block all UDP traffic, TURN
>    using TCP between the client and the server MUST be supported, and
>    TURN using TLS between the client and the server MUST be supported.
>    See [RFC5766] section 2.1 for details.
>
> I think the following part: "TURN using TLS between" should be changed
> over "TURN using TLS over TCP between" to make it clear that this is
> using TLS/TCP rather than TLS/FOO.
Done.
>
> 10. Section 3.4:
>    TURN TCP candidates [RFC6062] SHOULD be supported; this allows
>    applications to achieve peer-to-peer communication when both parties
>    are behind UDP-blocking firewalls using a single TURN server.  (In
>    this case, one can also achieve communication using two TURN servers
>    that use TCP between the server and the client, and UDP between the
>    TURN servers.)
>
> My understanding of RFC6062 is that it allows one to establish a vanilla
> TCP connection on the far side of the TURN server as well as connecting
> that one to an almost vanilla TCP connection (just a initial TURN
> ConnectionBind request) that A establish to the TURN server, i.e.
> WebRTC-A -(Turn/TCP)-> TURN_Server -(TCP)-> WebRTC-B.
>
> The end-result is that a relayed vanilla TCP connection is established
> between the WebRTC endpoints (A and B). As the WebRTC transport so far
> defined only works with datagrams, i.e. DTLS or RTP/RTCP packets they
> can't be sent natively over a TCP byte stream. Thus, if this mode of
> operation is to be supported a framing is required on this leg.
>
> A possibility here could be RFC 4571 if one want to go this way.

I added a 4571 reference, and a note saying this is up for discussion.
If we frame RTP over TCP, we also have to decide what we do about the SCTP.

>
> 11. Section 3.4:
>
>    ICE-TCP candidates [RFC6544] MAY be supported; this may allow
>    applications to communicate to peers with public IP addresses across
>    UDP-blocking firewalls without using a TURN server.
>
> Also this option would need to use a framing of the datagrams.

Yep.

Updated draft to be submitted once submission reopens.

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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Hi,

If the indication is NOT permanent, can't we use the existing direction att=
ributes (sendrecv, sendonly etc)?

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Harald Alvestrand <harald@alvestrand.no> wrote:



On 02/19/2014 09:19 AM, Christer Holmberg wrote:
Hi,

If the indication is =93permanent=94, rather than using a new msid-control =
attribute, could it be indicated by NOT including some information in the A=
nswer?

Certainly, but then you have to include some information in the answer when=
 the indication is not permanent.

Which information were you thinking of utilizing as a signal?



Regards,

Christer

Sent from Windows Mail

From: 'Harald Alvestrand'<mailto:harald@alvestrand.no>
Sent: =FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD:=FD35=FD =
=FDAM
To: Hans-Christer Holmberg<mailto:christer.holmberg@ericsson.com>, rtcweb@i=
etf.org<mailto:rtcweb@ietf.org>

One thing I did not add in my previous reply:

If an offerer sets the direction of an m-line to sendrecv, and the
answerer sets the direction to sendonly, the offerer will have no
indication of whether the answerer intended the suspension of traffic to
be temporary (msid-control: disable) or permanent (msid-control: reject
or msid-control: stop).

That was actually the main concern that drove the desire for a new
mechanism.
Once the new mechanism was in place, it seemed logical to be explicit
about the other possible actions too, rather than relying on the
sendrecv/sendonly/recvonly parameter.

          Harald




--
Surveillance is pervasive. Go Dark.


--_000_pi22myx9puvcw80lbn1s1ktu1392843033581emailandroidcom_
Content-Type: text/html; charset="windows-1256"
Content-Transfer-Encoding: quoted-printable

<html>
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256">
</head>
<body bgcolor=3D"#FFFFFF">
<pre style=3D"word-wrap:break-word; font-size:10.0pt; font-family:Tahoma; c=
olor:black">Hi,=0A=
=0A=
If the indication is NOT permanent, can't we use the existing direction att=
ributes (sendrecv, sendonly etc)?=0A=
=0A=
Regards,=0A=
=0A=
Christer=0A=
=0A=
Sent from my Sony Ericsson Xperia arc S=0A=
=0A=
Harald Alvestrand &lt;harald@alvestrand.no&gt; wrote:=0A=
=0A=
</pre>
<div>
<div class=3D"moz-cite-prefix">On 02/19/2014 09:19 AM, Christer Holmberg wr=
ote:<br>
</div>
<blockquote type=3D"cite"><style type=3D"text/css">
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<div dir=3D"ltr" style=3D"font-family:'Calibri','Segoe UI','Meiryo','Micros=
oft YaHei UI','Microsoft JhengHei UI','Malgun Gothic','sans-serif'; font-si=
ze:12pt">
<div>Hi,</div>
<div><br>
</div>
<div>If the indication is&nbsp;=93permanent=94, rather than using a new msi=
d-control attribute, could it be indicated by NOT including some informatio=
n in the Answer?</div>
</div>
</blockquote>
<br>
Certainly, but then you have to include some information in the answer when=
 the indication is not permanent.<br>
<br>
Which information were you thinking of utilizing as a signal?<br>
<br>
<br>
<blockquote type=3D"cite">
<div dir=3D"ltr" style=3D"font-family:'Calibri','Segoe UI','Meiryo','Micros=
oft YaHei UI','Microsoft JhengHei UI','Malgun Gothic','sans-serif'; font-si=
ze:12pt">
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer<br>
</div>
<div>
<div><br>
</div>
<div>Sent from Windows Mail</div>
<div><br>
</div>
</div>
<div style=3D"padding-top:5px; border-top-color:rgb(229,229,229); border-to=
p-width:1px; border-top-style:solid">
<div><font face=3D" 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei
              UI', 'Microsoft JhengHei UI', 'Malgun Gothic',
              'sans-serif'" style=3D""><b>From:</b>&nbsp;<a href=3D"mailto:=
harald@alvestrand.no" target=3D"_parent">'Harald Alvestrand'</a><br>
<b>Sent:</b>&nbsp;=FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD=
:=FD35=FD =FDAM<br>
<b>To:</b>&nbsp;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D=
"_parent">Hans-Christer Holmberg</a>,
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_parent">rtcweb@ietf.org</a></=
font></div>
</div>
<div><br>
</div>
<div dir=3D"">
<div id=3D"readingPaneBodyContent">One thing I did not add in my previous r=
eply:<br>
<br>
If an offerer sets the direction of an m-line to sendrecv, and the<br>
answerer sets the direction to sendonly, the offerer will have no<br>
indication of whether the answerer intended the suspension of traffic to<br=
>
be temporary (msid-control: disable) or permanent (msid-control: reject<br>
or msid-control: stop).<br>
<br>
That was actually the main concern that drove the desire for a new<br>
mechanism.<br>
Once the new mechanism was in place, it seemed logical to be explicit<br>
about the other possible actions too, rather than relying on the<br>
sendrecv/sendonly/recvonly parameter.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
</div>
</div>
</div>
</blockquote>
<br>
<br>
<pre class=3D"moz-signature" cols=3D"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
</div>
</body>
</html>

--_000_pi22myx9puvcw80lbn1s1ktu1392843033581emailandroidcom_--


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On 02/19/2014 09:50 PM, Christer Holmberg wrote:
> Hi,
>
> If the indication is NOT permanent, can't we use the existing direction attributes (sendrecv, sendonly etc)?

We could - but how would we then indicate "permanent"?

Omitting the direction attribute means the same as "sendrecv", if I'm
not mistaken. So you can't have meant omitting that.

So what were you suggesting?


> Regards,
>
> Christer
>
> Sent from my Sony Ericsson Xperia arc S
>
> Harald Alvestrand <harald@alvestrand.no> wrote:
>
> On 02/19/2014 09:19 AM, Christer Holmberg wrote:
>> Hi,
>>
>> If the indication is “permanent”, rather than using a new
>> msid-control attribute, could it be indicated by NOT including some
>> information in the Answer?
>
> Certainly, but then you have to include some information in the answer
> when the indication is not permanent.
>
> Which information were you thinking of utilizing as a signal?
>
>
>>
>> Regards,
>>
>> Christer
>>
>> Sent from Windows Mail
>>
>> *From:* 'Harald Alvestrand' <mailto:harald@alvestrand.no>
>> *Sent:* ýWednesdayý, ýFebruaryý ý19ý, ý2014 ý1ý:ý35ý ýAM
>> *To:* Hans-Christer Holmberg <mailto:christer.holmberg@ericsson.com>,
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>
>> One thing I did not add in my previous reply:
>>
>> If an offerer sets the direction of an m-line to sendrecv, and the
>> answerer sets the direction to sendonly, the offerer will have no
>> indication of whether the answerer intended the suspension of traffic to
>> be temporary (msid-control: disable) or permanent (msid-control: reject
>> or msid-control: stop).
>>
>> That was actually the main concern that drove the desire for a new
>> mechanism.
>> Once the new mechanism was in place, it seemed logical to be explicit
>> about the other possible actions too, rather than relying on the
>> sendrecv/sendonly/recvonly parameter.
>>
>>           Harald
>>
>
>
> -- 
> Surveillance is pervasive. Go Dark.


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 02/19/2014 09:50 PM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
      cite="mid:pi22myx9puvcw80lbn1s1ktu.1392843033581@email.android.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1256">
      <pre style="word-wrap:break-word; font-size:10.0pt; font-family:Tahoma; color:black">Hi,

If the indication is NOT permanent, can't we use the existing direction attributes (sendrecv, sendonly etc)?
</pre>
    </blockquote>
    <br>
    We could - but how would we then indicate "permanent"?<br>
    <br>
    Omitting the direction attribute means the same as "sendrecv", if
    I'm not mistaken. So you can't have meant omitting that.<br>
    <br>
    So what were you suggesting?<br>
    <br>
    <br>
    <blockquote
      cite="mid:pi22myx9puvcw80lbn1s1ktu.1392843033581@email.android.com"
      type="cite">
      <pre style="word-wrap:break-word; font-size:10.0pt; font-family:Tahoma; color:black">
Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a> wrote:

</pre>
      <div>
        <div class="moz-cite-prefix">On 02/19/2014 09:19 AM, Christer
          Holmberg wrote:<br>
        </div>
        <blockquote type="cite">
          <style type="text/css">
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          <div dir="ltr" style="font-family:'Calibri','Segoe
            UI','Meiryo','Microsoft YaHei UI','Microsoft JhengHei
            UI','Malgun Gothic','sans-serif'; font-size:12pt">
            <div>Hi,</div>
            <div><br>
            </div>
            <div>If the indication is “permanent”, rather than using a
              new msid-control attribute, could it be indicated by NOT
              including some information in the Answer?</div>
          </div>
        </blockquote>
        <br>
        Certainly, but then you have to include some information in the
        answer when the indication is not permanent.<br>
        <br>
        Which information were you thinking of utilizing as a signal?<br>
        <br>
        <br>
        <blockquote type="cite">
          <div dir="ltr" style="font-family:'Calibri','Segoe
            UI','Meiryo','Microsoft YaHei UI','Microsoft JhengHei
            UI','Malgun Gothic','sans-serif'; font-size:12pt">
            <div><br>
            </div>
            <div>Regards,</div>
            <div><br>
            </div>
            <div>Christer<br>
            </div>
            <div>
              <div><br>
              </div>
              <div>Sent from Windows Mail</div>
              <div><br>
              </div>
            </div>
            <div style="padding-top:5px;
              border-top-color:rgb(229,229,229); border-top-width:1px;
              border-top-style:solid">
              <div><font style="" face=" 'Calibri', 'Segoe UI',
                  'Meiryo', 'Microsoft YaHei UI', 'Microsoft JhengHei
                  UI', 'Malgun Gothic', 'sans-serif'"><b>From:</b> <a
                    moz-do-not-send="true"
                    href="mailto:harald@alvestrand.no" target="_parent">'Harald
                    Alvestrand'</a><br>
                  <b>Sent:</b> ýWednesdayý, ýFebruaryý ý19ý, ý2014
                  ý1ý:ý35ý ýAM<br>
                  <b>To:</b> <a moz-do-not-send="true"
                    href="mailto:christer.holmberg@ericsson.com"
                    target="_parent">Hans-Christer Holmberg</a>,
                  <a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org" target="_parent">rtcweb@ietf.org</a></font></div>
            </div>
            <div><br>
            </div>
            <div dir="">
              <div id="readingPaneBodyContent">One thing I did not add
                in my previous reply:<br>
                <br>
                If an offerer sets the direction of an m-line to
                sendrecv, and the<br>
                answerer sets the direction to sendonly, the offerer
                will have no<br>
                indication of whether the answerer intended the
                suspension of traffic to<br>
                be temporary (msid-control: disable) or permanent
                (msid-control: reject<br>
                or msid-control: stop).<br>
                <br>
                That was actually the main concern that drove the desire
                for a new<br>
                mechanism.<br>
                Once the new mechanism was in place, it seemed logical
                to be explicit<br>
                about the other possible actions too, rather than
                relying on the<br>
                sendrecv/sendonly/recvonly parameter.<br>
                <br>
                          Harald<br>
                <br>
              </div>
            </div>
          </div>
        </blockquote>
        <br>
        <br>
        <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------030909010605010002060504--


From nobody Thu Feb 20 08:05:39 2014
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-security-06
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Hi,

I have reviewed Security Considerations for WebRTC
(draft-ietf-rtcweb-security-06) and have some comments.

1. Section 4.2.1:

I noted this:
   There also needs to be some mechanism for the browser to verify that
   the target of the traffic continues to wish to receive it.  Because
   ICE keepalives are indications, they will not work here, so some
   other mechanism is needed as described in
   [I-D.muthu-behave-consent-freshness].

Not that the current information is wrong, except that the reference
should be to the WG version of the draft. However, this I think is the
only part of the document I found which prevents another WG last call on
this document, due to the open issue regarding method of communication
consent refresh.

2. Section 4.2.2.:

First sentence duplicated.

3. Section 4.2.2:

When I read this section, my immediate question was, doesn't RTP need
corresponding protection. To my understanding, the answer would be yes,
but it is more difficult to inject data and get a certain plaintext even
without the SRTP encryption scrambling the payload part. Still, that
assumes that an JS application, can write the payload data itself.
Secondly, it does assume that one MUST use encryption in SRTP.

Thus, I wonder if it is worth spending a paragraph confirming the need,
although reduced for masking also of RTP content, and that the same
requirement exist for RTP.

4. Section 4.3.2.2:
   Once uses have checked the SAS once, key continuity is required to
   avoid them needing to check it on every call.

I believe "uses" in above sentence should be "the user"?

5. Section 6.

Please correct my last name.

6. General:

I think the difficulty with this document is to determine if is missing
something? It is providing the high level system security issues. Have
we caught all relevant ones? I think we need a lot of eyes to check for
that. And that requires thought and not being restricted by the document.

I think a potential issue that isn't discussed in this document is the
security threat of driving data volumes into a network beyond ones
"fair" share. This would simply be to illustrate that communication
consent is not sufficient, to protect other users of a shared network
the WebRTC endpoint MUST prevent transmission of data volumes far
outside of the fair share.

7. General:

I noticed that some of the other WG documents may think they have
outsourced their security consideration to this document. I want to
bring this up as a general question and ensure that we have WG agreement
on this.

This document discusses general system security considerations to ensure
that the whole thing we put together achieve security. Each component of
the solution will have to discuss security considerations details around
their components.

As we plan to WG last call this document as soon as possible, we are
running out of time to lift any security issues to this level.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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On Thu, Feb 20, 2014 at 8:05 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I think a potential issue that isn't discussed in this document is the
> security threat of driving data volumes into a network beyond ones
> "fair" share. This would simply be to illustrate that communication
> consent is not sufficient, to protect other users of a shared network
> the WebRTC endpoint MUST prevent transmission of data volumes far
> outside of the fair share.
>
>
>
Magnus, I'm not sure why you think that issue should be dealt with in this
document.
This document should deal with cases where a malicious script could send
transmissions
that are not intended or not welcome.  Bbut managing network capacity for
*intended*
and *welcome* transmissions is not a security concern.  It's a congestion
management
issue, at least as I see it.

Am I missing something about your concern?

Ted

--089e0158aa12b2df1304f2d9a0af
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Feb 20, 2014 at 8:05 AM=
, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlu=
nd@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</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">Hi,<br>
<br>
I think a potential issue that isn&#39;t discussed in this document is the<=
br>
security threat of driving data volumes into a network beyond ones<br>
&quot;fair&quot; share. This would simply be to illustrate that communicati=
on<br>
consent is not sufficient, to protect other users of a shared network<br>
the WebRTC endpoint MUST prevent transmission of data volumes far<br>
outside of the fair share.<br>
<br><br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-fa=
mily:georgia,serif;display:inline">Magnus, I&#39;m not sure why you think t=
hat issue should be dealt with in this document.<br></div><div class=3D"gma=
il_default" style=3D"font-family:georgia,serif;display:inline">
This document should deal with cases where a malicious script could send tr=
ansmissions<br></div><div class=3D"gmail_default" style=3D"font-family:geor=
gia,serif;display:inline">that are not intended or not welcome.=A0 Bbut man=
aging network capacity for *intended*<br>
and *welcome* transmissions is not a security concern.=A0 It&#39;s a conges=
tion management<br>issue, at least as I see it.<br><br></div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;display:inline">Am I miss=
ing something about your concern?<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;d=
isplay:inline">Ted</div>=A0</div></div></div></div>

--089e0158aa12b2df1304f2d9a0af--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Stream control: [MMUSIC] msid-04 submitted
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Hi,

My suggestion was to omit something else. But, at the moment I don=92t have=
 the drafts in front of me, so I can=92t suggest anything specific.

Regards,

Christer

Sent from Windows Mail

From: 'Harald Alvestrand'<mailto:harald@alvestrand.no>
Sent: =FDThursday=FD, =FDFebruary=FD =FD20=FD, =FD2014 =FD2=FD:=FD22=FD =FD=
AM
To: Hans-Christer Holmberg<mailto:christer.holmberg@ericsson.com>, rtcweb@i=
etf.org<mailto:rtcweb@ietf.org>

On 02/19/2014 09:50 PM, Christer Holmberg wrote:

Hi,

If the indication is NOT permanent, can't we use the existing direction att=
ributes (sendrecv, sendonly etc)?


We could - but how would we then indicate "permanent"?

Omitting the direction attribute means the same as "sendrecv", if I'm not m=
istaken. So you can't have meant omitting that.

So what were you suggesting?



Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Harald Alvestrand <harald@alvestrand.no><mailto:harald@alvestrand.no> wrote=
:



On 02/19/2014 09:19 AM, Christer Holmberg wrote:
Hi,

If the indication is =93permanent=94, rather than using a new msid-control =
attribute, could it be indicated by NOT including some information in the A=
nswer?

Certainly, but then you have to include some information in the answer when=
 the indication is not permanent.

Which information were you thinking of utilizing as a signal?



Regards,

Christer

Sent from Windows Mail

From: 'Harald Alvestrand'<mailto:harald@alvestrand.no>
Sent: =FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD:=FD35=FD =
=FDAM
To: Hans-Christer Holmberg<mailto:christer.holmberg@ericsson.com>, rtcweb@i=
etf.org<mailto:rtcweb@ietf.org>

One thing I did not add in my previous reply:

If an offerer sets the direction of an m-line to sendrecv, and the
answerer sets the direction to sendonly, the offerer will have no
indication of whether the answerer intended the suspension of traffic to
be temporary (msid-control: disable) or permanent (msid-control: reject
or msid-control: stop).

That was actually the main concern that drove the desire for a new
mechanism.
Once the new mechanism was in place, it seemed logical to be explicit
about the other possible actions too, rather than relying on the
sendrecv/sendonly/recvonly parameter.

          Harald




--
Surveillance is pervasive. Go Dark.




--
Surveillance is pervasive. Go Dark.


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algun Gothic', 'sans-serif';font-size:12pt;">
<div>Hi,</div>
<div><br>
</div>
<div>My suggestion was to omit something else. But, at the moment I don=92t=
 have the drafts in front of me, so I can=92t suggest anything specific.</d=
iv>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer<br>
</div>
<div data-signatureblock=3D"true">
<div><br>
</div>
<div>Sent from Windows Mail</div>
<div><br>
</div>
</div>
<div style=3D"padding-top: 5px; border-top-color: rgb(229, 229, 229); borde=
r-top-width: 1px; border-top-style: solid;">
<div><font face=3D" 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei UI', =
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t: 15pt; letter-spacing: 0.02em; font-family: &quot;Calibri&quot;, &quot;Se=
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rosoft JhengHei UI&quot;, &quot;Malgun Gothic&quot;, &quot;sans-serif&quot;=
; font-size: 12pt;"><b>From:</b>&nbsp;<a href=3D"mailto:harald@alvestrand.n=
o" target=3D"_parent">'Harald
 Alvestrand'</a><br>
<b>Sent:</b>&nbsp;=FDThursday=FD, =FDFebruary=FD =FD20=FD, =FD2014 =FD2=FD:=
=FD22=FD =FDAM<br>
<b>To:</b>&nbsp;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D=
"_parent">Hans-Christer Holmberg</a>,
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_parent">rtcweb@ietf.org</a></=
font></div>
</div>
<div><br>
</div>
<div dir=3D"">
<div class=3D"moz-cite-prefix">On 02/19/2014 09:50 PM, Christer Holmberg wr=
ote:<br>
</div>
<blockquote style=3D"margin-top: 0px; margin-bottom: 0px;">
<pre style=3D"color: black; font-family: Tahoma; font-size: 10pt; -ms-word-=
wrap: break-word;">Hi,=0A=
=0A=
If the indication is NOT permanent, can't we use the existing direction att=
ributes (sendrecv, sendonly etc)?=0A=
</pre>
</blockquote>
<br>
We could - but how would we then indicate &quot;permanent&quot;?<br>
<br>
Omitting the direction attribute means the same as &quot;sendrecv&quot;, if=
 I'm not mistaken. So you can't have meant omitting that.<br>
<br>
So what were you suggesting?<br>
<br>
<br>
<blockquote style=3D"margin-top: 0px; margin-bottom: 0px;">
<pre style=3D"color: black; font-family: Tahoma; font-size: 10pt; -ms-word-=
wrap: break-word;">Regards,=0A=
=0A=
Christer=0A=
=0A=
Sent from my Sony Ericsson Xperia arc S=0A=
=0A=
Harald Alvestrand <a class=3D"moz-txt-link-rfc2396E" href=3D"mailto:harald@=
alvestrand.no" target=3D"_parent">&lt;harald@alvestrand.no&gt;</a> wrote:=
=0A=
=0A=
</pre>
<div>
<div class=3D"moz-cite-prefix">On 02/19/2014 09:19 AM, Christer Holmberg wr=
ote:<br>
</div>
<blockquote style=3D"margin-top: 0px; margin-bottom: 0px;">
<div dir=3D"ltr">
<div>Hi,</div>
<div><br>
</div>
<div>If the indication is&nbsp;=93permanent=94, rather than using a new msi=
d-control attribute, could it be indicated by NOT including some informatio=
n in the Answer?</div>
</div>
</blockquote>
<br>
Certainly, but then you have to include some information in the answer when=
 the indication is not permanent.<br>
<br>
Which information were you thinking of utilizing as a signal?<br>
<br>
<br>
<blockquote style=3D"margin-top: 0px; margin-bottom: 0px;">
<div dir=3D"ltr">
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer<br>
</div>
<div>
<div><br>
</div>
<div>Sent from Windows Mail</div>
<div><br>
</div>
</div>
<div style=3D"padding-top: 5px; border-top-color: rgb(229, 229, 229); borde=
r-top-width: 1px; border-top-style: solid;">
<div><font face=3D" 'Calibri', 'Segoe UI',                    'Meiryo', 'Mi=
crosoft YaHei UI', 'Microsoft JhengHei                    UI', 'Malgun Goth=
ic', 'sans-serif'"><b>From:</b>&nbsp;<a href=3D"mailto:harald@alvestrand.no=
" target=3D"_parent">'Harald Alvestrand'</a><br>
<b>Sent:</b>&nbsp;=FDWednesday=FD, =FDFebruary=FD =FD19=FD, =FD2014 =FD1=FD=
:=FD35=FD =FDAM<br>
<b>To:</b>&nbsp;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D=
"_parent">Hans-Christer Holmberg</a>,
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_parent">rtcweb@ietf.org</a></=
font></div>
</div>
<div><br>
</div>
<div dir=3D"">
<div id=3D"readingPaneBodyContent">One thing I did not add in my previous r=
eply:<br>
<br>
If an offerer sets the direction of an m-line to sendrecv, and the<br>
answerer sets the direction to sendonly, the offerer will have no<br>
indication of whether the answerer intended the suspension of traffic to<br=
>
be temporary (msid-control: disable) or permanent (msid-control: reject<br>
or msid-control: stop).<br>
<br>
That was actually the main concern that drove the desire for a new<br>
mechanism.<br>
Once the new mechanism was in place, it seemed logical to be explicit<br>
about the other possible actions too, rather than relying on the<br>
sendrecv/sendonly/recvonly parameter.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
</div>
</div>
</div>
</blockquote>
<br>
<br>
<pre class=3D"moz-signature">-- =0A=
Surveillance is pervasive. Go Dark.=0A=
</pre>
</div>
</blockquote>
<br>
<br>
<pre class=3D"moz-signature">-- =0A=
Surveillance is pervasive. Go Dark.=0A=
</pre>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D1ACAA5ESESSMB209erics_--


From nobody Thu Feb 20 14:13:33 2014
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On Feb 19, 2014, at 12:07 PM, Lijing (Jessie) <lijing80@huawei.com> =
wrote:

> Hi,
> =20
> Several comments on data channel related drafts.
> =20
> Firstly, There are several =93protocol=94 concepts appeared in =
different data channel related drafts, and it seems there is no clear =
statement about the relationship between these concepts.
> =20
> draft-ietf-rtcweb-data-protocol has two protocol concept:
> 1.  The protocol byte in data channel open message defined as the =
protocol for the channel.
> 2.  SCTP payload protocol identifier (PPID)
> =20
> WebRTC W3C API has one protocol concept,
> 3.  Protocol defined as =93Subprotocol name used for this channel=94
> =20
> Q1: is the W3C API parameter =93protocol=94 be the same as the =
protocol byte in the data channel open message?
Correct. The DOMString protocol provided in the W3C API in negotiated in =
the DATA_CHANNEL_OPEN message as
the protocol field. Please note that the Protocol field in the =
DATA_CHANNEL_OPEN message is not a byte,
but a string.
> =20
> =20
> In previous email discussion, Peter Thatcher said :
> =93The PPID isn't exposed to the application.  It's just used to
> distinguish text/binary/control.    If an application uses different
> data channels with different protocols, it should use the SID to =
distinguish between them, not the PPID.  In fact, even if it wanted to =
use the PPID, it wouldn't be able to.=94
Data channels are identified by the SID on the wire, which corresponds =
to the id field in the W3C API.
The JS API has no access to the PPID.
> =20
> Q2: if the answer of Q1 is yes. Then an application can distinguish =
different data channels with different protocols by the protocol byte in =
the data channel open message. Is my understanding right?
Not sure what the application is. The W3C API lets you distinguish the =
data channels, since
you have different objects for it.
> =20
> draft-ejzak-mmusic-data-channel-sdpneg provides a way to negotiate =
sub-protocol parameters out of band.
> 4.  Sub-protocol: The 'sub-protocol' parameter indicates which =
protocol the client expects to exchange via the channel.=20
> example: a=3Ddcmap:5000 stream=3D2;label=3D"channel 2"; =
subprotocol=3D"BFCP";max_retr=3D3
> =20
> Q3: is this =93subprotocol=94 be the same concept as the protocol byte =
in the data channel open message? And the W3C API =93protocol=94 should =
be set as BFCP?
Again: there is no protocol byte... It is a string and is is the =
subprotocol...
> =20
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> =20
> Secondly, in draft-ietf-rtcweb-data-channel-07, section 8 defines the =
following PPIDs
>       +------------------------------------+-----------+-----------+
>       | Value                              | SCTP PPID | Reference |
>       +------------------------------------+-----------+-----------+
>       | WebRTC String                      | 51        | [RFCXXXX] |
>       | WebRTC Binary Partial (Deprecated) | 52        | [RFCXXXX] |
>       | WebRTC Binary                      | 53        | [RFCXXXX] |
>       | WebRTC String Partial (Deprecated) | 54        | [RFCXXXX] |
>       +------------------------------------+-----------+-----------+
> =20
> Q4: It is confusing what is the intention to define =93WebRTC Binary =
Partial=94 and =93WebRTC String Partial=94. I guess these two PPIDs are =
provided to the application in the case when application supports =
fragmentation of large messages. But, at the meantime, in this draft, =
section 6.6 also says: The usage of the PPIDs "WebRTC String Partial" =
and "WebRTC Binary Partial" is deprecated. Why the usage is deprecated? =
Does this implies that SCTP layer is responsible for fragmentation of =
large messages, and it is unnecessary for the upper application to =
implement the fragmentation of large messages?
Section 6.6 explains it:

The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary =
Partial" is deprecated. They were used for a PPID based fragmentation =
and reassembly of user messages belonging to reliable and ordered data =
channels.
> =20
> Q5: Since PPID 52 and 54 are defined, then how can the browser know =
whether the data sent from the upper application layer is a part of a =
whole message and the SCTP PPID should be signaled as Partial 52/54, or =
this is a whole message and should be signaled as 51/54?
The PPIDs 52 and 54 were intended to implement fragmentation/reassembly =
inside the data channel implementation.
This is how it is implemented in Firefox. There is now way to access it =
from the JS API.
However, the usage is deprecated and therefore you don't need to =
implement it.
> =20
> Q6: if one side wants to send two messages, and the application split =
the first big message to two packages, then the browser will send three =
SCTP packages whose PPID are 54(part of first message), 51(part of first =
message), 51(the whole second message) respectively. When the receiver =
side receives these three SCTP packages, how can the receiver side know =
which two SCTP packages can be assembled to get the first big message?
It worked only for ordered reliable data channels. So the sequence was =
used. Once you got PPID 51, the message is complete.
> =20
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> =20
> The last question, which I have asked in previous email and have not =
received any response until now, so I mention this question again.
> in draft-ietf-rtcweb-data-channel-07, Section 4 says:
>    Req. 4:   The application SHOULD be able to provide guidance as to
>              the relative priority of each data channel relative to =
each
>              other, and relative to the media streams.  This will
>              interact with the congestion control algorithms.
> =20
> And in draft-ietf-rtcweb-data-protocol-03, DATA_CHANNEL_OPEN Message =
is defined as below
> 0                   1                   2                   3
>     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |  Message Type |  Channel Type |            Priority           |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |                    Reliability Parameter                      |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |         Label Length          |       Protocol Length         |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    \                                                               /
>    |                             Label                             |
>    /                                                               \
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    \                                                               /
>    |                            Protocol                           |
>    /                                                               \
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> Priority: 2 bytes (integer)
>       The priority of the channel as described in
>       [I-D.ietf-rtcweb-data-channel].  The higher the number, the =
lower
>       the priority.
> =20
> Q7: It seems that for now there is no W3C API parameters to set the =
priority of data channel. I wonder whether the priority of the data =
channel is open to the application or is only decided by the browser. In =
other words, I want to know how the application be able to provide =
guidance as to the priority.
You are right. The W3C API does not cover the priority yet. My =
understanding is
that this is missing and needs to be added. I expect that the priority =
is settable
via the JS API.

Best regards
Michael
> =20
> Best Regards,
> =20
> Jessie
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 2014-02-20 18:13, Ted Hardie wrote:
> On Thu, Feb 20, 2014 at 8:05 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     Hi,
> 
>     I think a potential issue that isn't discussed in this document is the
>     security threat of driving data volumes into a network beyond ones
>     "fair" share. This would simply be to illustrate that communication
>     consent is not sufficient, to protect other users of a shared network
>     the WebRTC endpoint MUST prevent transmission of data volumes far
>     outside of the fair share.
> 
> 
> 
> Magnus, I'm not sure why you think that issue should be dealt with in
> this document.
> This document should deal with cases where a malicious script could send
> transmissions
> that are not intended or not welcome.  Bbut managing network capacity
> for *intended*
> and *welcome* transmissions is not a security concern.  It's a
> congestion management
> issue, at least as I see it.
> 
> Am I missing something about your concern?
> 

My concern is that a malicious individual wants to DDoS a particular
network. I buy AD time from on of the network or in any other way ensure
that the mailicious JS identifies when each client is topologically
right for me to connect it to another client to drive traffic through my
targets network. Then I ensure these clients establish a peer connection
and I try to blast as much traffic through the network as possible. From
the point of the attack this doesn't matter if is real-time media or
data traffic. The point is to drive as much traffic into the network as
possible. From that perspective any traffic that is sent in the context
of a peerconnection needs to be congestion controlled. That at least
keeps the traffic down to a "fair" share.

We are mitigating this attack, however what I was really wondering over
is if the security considerations document should make note that lack of
appropriate congestion control is a security vulnerability and enables
certain type of load attacks.

Does this make my concern clearer?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Lijing (Jessie)" <lijing80@huawei.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Comments on data channel related drafts
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Date: Fri, 21 Feb 2014 09:21:10 +0000
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Subject: Re: [rtcweb] Comments on data channel related drafts
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Hi,

Thanks a lot to Michael's interpretation, and it is really helpful.

Two more issues to be discussed:
1. I am still not quite sure what "PPID based fragmentation and reassembly =
" is. Does it mean SCTP performs segmentation and reassembly based on the p=
ath MTU? Or it refers to other method or protocols?

2. The priority of data channel involves two aspects, one aspect is priorit=
y of one data channel compared with priority of other data channels, and th=
e other aspect is priority of one data channel compared with audio and vide=
o media stream tracks. If the data channel priority is settable via JS API,=
 then the audio and video priority should also be set via JS API in a way. =
I think the priority should be set considering all the media types includin=
g audio, video and data channel.

BR,
Jessie
-----Original Message-----
From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]=20
Sent: Friday, February 21, 2014 6:13 AM
To: Lijing (Jessie)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Comments on data channel related drafts

On Feb 19, 2014, at 12:07 PM, Lijing (Jessie) <lijing80@huawei.com> wrote:

> Hi,
> =20
> Several comments on data channel related drafts.
> =20
> Firstly, There are several "protocol" concepts appeared in different data=
 channel related drafts, and it seems there is no clear statement about the=
 relationship between these concepts.
> =20
> draft-ietf-rtcweb-data-protocol has two protocol concept:
> 1.  The protocol byte in data channel open message defined as the protoco=
l for the channel.
> 2.  SCTP payload protocol identifier (PPID)
> =20
> WebRTC W3C API has one protocol concept, 3.  Protocol defined as=20
> "Subprotocol name used for this channel"
> =20
> Q1: is the W3C API parameter "protocol" be the same as the protocol byte =
in the data channel open message?
Correct. The DOMString protocol provided in the W3C API in negotiated in th=
e DATA_CHANNEL_OPEN message as the protocol field. Please note that the Pro=
tocol field in the DATA_CHANNEL_OPEN message is not a byte, but a string.
> =20
> =20
> In previous email discussion, Peter Thatcher said :
> "The PPID isn't exposed to the application.  It's just used to
> distinguish text/binary/control.    If an application uses different
> data channels with different protocols, it should use the SID to distingu=
ish between them, not the PPID.  In fact, even if it wanted to use the PPID=
, it wouldn't be able to."
Data channels are identified by the SID on the wire, which corresponds to t=
he id field in the W3C API.
The JS API has no access to the PPID.
> =20
> Q2: if the answer of Q1 is yes. Then an application can distinguish diffe=
rent data channels with different protocols by the protocol byte in the dat=
a channel open message. Is my understanding right?
Not sure what the application is. The W3C API lets you distinguish the data=
 channels, since you have different objects for it.
> =20
> draft-ejzak-mmusic-data-channel-sdpneg provides a way to negotiate sub-pr=
otocol parameters out of band.
> 4.  Sub-protocol: The 'sub-protocol' parameter indicates which protocol t=
he client expects to exchange via the channel.=20
> example: a=3Ddcmap:5000 stream=3D2;label=3D"channel 2";=20
> subprotocol=3D"BFCP";max_retr=3D3
> =20
> Q3: is this "subprotocol" be the same concept as the protocol byte in the=
 data channel open message? And the W3C API "protocol" should be set as BFC=
P?
Again: there is no protocol byte... It is a string and is is the subprotoco=
l...
> =20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> =3D=3D
> =20
> Secondly, in draft-ietf-rtcweb-data-channel-07, section 8 defines the fol=
lowing PPIDs
>       +------------------------------------+-----------+-----------+
>       | Value                              | SCTP PPID | Reference |
>       +------------------------------------+-----------+-----------+
>       | WebRTC String                      | 51        | [RFCXXXX] |
>       | WebRTC Binary Partial (Deprecated) | 52        | [RFCXXXX] |
>       | WebRTC Binary                      | 53        | [RFCXXXX] |
>       | WebRTC String Partial (Deprecated) | 54        | [RFCXXXX] |
>       +------------------------------------+-----------+-----------+
> =20
> Q4: It is confusing what is the intention to define "WebRTC Binary Partia=
l" and "WebRTC String Partial". I guess these two PPIDs are provided to the=
 application in the case when application supports fragmentation of large m=
essages. But, at the meantime, in this draft, section 6.6 also says: The us=
age of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial" is dep=
recated. Why the usage is deprecated? Does this implies that SCTP layer is =
responsible for fragmentation of large messages, and it is unnecessary for =
the upper application to implement the fragmentation of large messages?
Section 6.6 explains it:

The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial" =
is deprecated. They were used for a PPID based fragmentation and reassembly=
 of user messages belonging to reliable and ordered data channels.
> =20
> Q5: Since PPID 52 and 54 are defined, then how can the browser know wheth=
er the data sent from the upper application layer is a part of a whole mess=
age and the SCTP PPID should be signaled as Partial 52/54, or this is a who=
le message and should be signaled as 51/54?
The PPIDs 52 and 54 were intended to implement fragmentation/reassembly ins=
ide the data channel implementation.
This is how it is implemented in Firefox. There is now way to access it fro=
m the JS API.
However, the usage is deprecated and therefore you don't need to implement =
it.
> =20
> Q6: if one side wants to send two messages, and the application split the=
 first big message to two packages, then the browser will send three SCTP p=
ackages whose PPID are 54(part of first message), 51(part of first message)=
, 51(the whole second message) respectively. When the receiver side receive=
s these three SCTP packages, how can the receiver side know which two SCTP =
packages can be assembled to get the first big message?
It worked only for ordered reliable data channels. So the sequence was used=
. Once you got PPID 51, the message is complete.
> =20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> =3D=3D=3D
> =20
> The last question, which I have asked in previous email and have not rece=
ived any response until now, so I mention this question again.
> in draft-ietf-rtcweb-data-channel-07, Section 4 says:
>    Req. 4:   The application SHOULD be able to provide guidance as to
>              the relative priority of each data channel relative to each
>              other, and relative to the media streams.  This will
>              interact with the congestion control algorithms.
> =20
> And in draft-ietf-rtcweb-data-protocol-03, DATA_CHANNEL_OPEN Message is d=
efined as below
> 0                   1                   2                   3
>     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |  Message Type |  Channel Type |            Priority           |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |                    Reliability Parameter                      |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    |         Label Length          |       Protocol Length         |
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    \                                                               /
>    |                             Label                             |
>    /                                                               \
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>    \                                                               /
>    |                            Protocol                           |
>    /                                                               \
>    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> Priority: 2 bytes (integer)
>       The priority of the channel as described in
>       [I-D.ietf-rtcweb-data-channel].  The higher the number, the lower
>       the priority.
> =20
> Q7: It seems that for now there is no W3C API parameters to set the prior=
ity of data channel. I wonder whether the priority of the data channel is o=
pen to the application or is only decided by the browser. In other words, I=
 want to know how the application be able to provide guidance as to the pri=
ority.
You are right. The W3C API does not cover the priority yet. My understandin=
g is that this is missing and needs to be added. I expect that the priority=
 is settable via the JS API.

Best regards
Michael
> =20
> Best Regards,
> =20
> Jessie
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On Feb 21, 2014, at 10:21 AM, Lijing (Jessie) <lijing80@huawei.com> =
wrote:

> Hi,
>=20
> Thanks a lot to Michael's interpretation, and it is really helpful.
>=20
> Two more issues to be discussed:
> 1. I am still not quite sure what "PPID based fragmentation and =
reassembly " is. Does it mean SCTP performs segmentation and reassembly =
based on the path MTU? Or it refers to other method or protocols?
SCTP performs fragmentation and reassembly as an internal mechanism. =
This is not related to the use of the PPID.
The "PPID based fragmentation and reassembly" is implemented on top of =
SCTP, but below the JS API.
So it is NOT done by the Javascript application.
When using NDATA, the PPID stuff is not necessary, since fragmentation =
and reassembly will be
done by SCTP without monopolizing any stream.
>=20
> 2. The priority of data channel involves two aspects, one aspect is =
priority of one data channel compared with priority of other data =
channels, and the other aspect is priority of one data channel compared =
with audio and video media stream tracks. If the data channel priority =
is settable via JS API, then the audio and video priority should also be =
set via JS API in a way. I think the priority should be set considering =
all the media types including audio, video and data channel.
I agree completely. And there is currently also no way of specifying a =
priority for the
media streams. However, this is an issue for W3C...

Best regards
Michael
>=20
> BR,
> Jessie
> -----Original Message-----
> From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]=20
> Sent: Friday, February 21, 2014 6:13 AM
> To: Lijing (Jessie)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Comments on data channel related drafts
>=20
> On Feb 19, 2014, at 12:07 PM, Lijing (Jessie) <lijing80@huawei.com> =
wrote:
>=20
>> Hi,
>>=20
>> Several comments on data channel related drafts.
>>=20
>> Firstly, There are several "protocol" concepts appeared in different =
data channel related drafts, and it seems there is no clear statement =
about the relationship between these concepts.
>>=20
>> draft-ietf-rtcweb-data-protocol has two protocol concept:
>> 1.  The protocol byte in data channel open message defined as the =
protocol for the channel.
>> 2.  SCTP payload protocol identifier (PPID)
>>=20
>> WebRTC W3C API has one protocol concept, 3.  Protocol defined as=20
>> "Subprotocol name used for this channel"
>>=20
>> Q1: is the W3C API parameter "protocol" be the same as the protocol =
byte in the data channel open message?
> Correct. The DOMString protocol provided in the W3C API in negotiated =
in the DATA_CHANNEL_OPEN message as the protocol field. Please note that =
the Protocol field in the DATA_CHANNEL_OPEN message is not a byte, but a =
string.
>>=20
>>=20
>> In previous email discussion, Peter Thatcher said :
>> "The PPID isn't exposed to the application.  It's just used to
>> distinguish text/binary/control.    If an application uses different
>> data channels with different protocols, it should use the SID to =
distinguish between them, not the PPID.  In fact, even if it wanted to =
use the PPID, it wouldn't be able to."
> Data channels are identified by the SID on the wire, which corresponds =
to the id field in the W3C API.
> The JS API has no access to the PPID.
>>=20
>> Q2: if the answer of Q1 is yes. Then an application can distinguish =
different data channels with different protocols by the protocol byte in =
the data channel open message. Is my understanding right?
> Not sure what the application is. The W3C API lets you distinguish the =
data channels, since you have different objects for it.
>>=20
>> draft-ejzak-mmusic-data-channel-sdpneg provides a way to negotiate =
sub-protocol parameters out of band.
>> 4.  Sub-protocol: The 'sub-protocol' parameter indicates which =
protocol the client expects to exchange via the channel.=20
>> example: a=3Ddcmap:5000 stream=3D2;label=3D"channel 2";=20
>> subprotocol=3D"BFCP";max_retr=3D3
>>=20
>> Q3: is this "subprotocol" be the same concept as the protocol byte in =
the data channel open message? And the W3C API "protocol" should be set =
as BFCP?
> Again: there is no protocol byte... It is a string and is is the =
subprotocol...
>>=20
>> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> =3D=3D
>>=20
>> Secondly, in draft-ietf-rtcweb-data-channel-07, section 8 defines the =
following PPIDs
>>      +------------------------------------+-----------+-----------+
>>      | Value                              | SCTP PPID | Reference |
>>      +------------------------------------+-----------+-----------+
>>      | WebRTC String                      | 51        | [RFCXXXX] |
>>      | WebRTC Binary Partial (Deprecated) | 52        | [RFCXXXX] |
>>      | WebRTC Binary                      | 53        | [RFCXXXX] |
>>      | WebRTC String Partial (Deprecated) | 54        | [RFCXXXX] |
>>      +------------------------------------+-----------+-----------+
>>=20
>> Q4: It is confusing what is the intention to define "WebRTC Binary =
Partial" and "WebRTC String Partial". I guess these two PPIDs are =
provided to the application in the case when application supports =
fragmentation of large messages. But, at the meantime, in this draft, =
section 6.6 also says: The usage of the PPIDs "WebRTC String Partial" =
and "WebRTC Binary Partial" is deprecated. Why the usage is deprecated? =
Does this implies that SCTP layer is responsible for fragmentation of =
large messages, and it is unnecessary for the upper application to =
implement the fragmentation of large messages?
> Section 6.6 explains it:
>=20
> The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary =
Partial" is deprecated. They were used for a PPID based fragmentation =
and reassembly of user messages belonging to reliable and ordered data =
channels.
>>=20
>> Q5: Since PPID 52 and 54 are defined, then how can the browser know =
whether the data sent from the upper application layer is a part of a =
whole message and the SCTP PPID should be signaled as Partial 52/54, or =
this is a whole message and should be signaled as 51/54?
> The PPIDs 52 and 54 were intended to implement =
fragmentation/reassembly inside the data channel implementation.
> This is how it is implemented in Firefox. There is now way to access =
it from the JS API.
> However, the usage is deprecated and therefore you don't need to =
implement it.
>>=20
>> Q6: if one side wants to send two messages, and the application split =
the first big message to two packages, then the browser will send three =
SCTP packages whose PPID are 54(part of first message), 51(part of first =
message), 51(the whole second message) respectively. When the receiver =
side receives these three SCTP packages, how can the receiver side know =
which two SCTP packages can be assembled to get the first big message?
> It worked only for ordered reliable data channels. So the sequence was =
used. Once you got PPID 51, the message is complete.
>>=20
>> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> =3D=3D=3D
>>=20
>> The last question, which I have asked in previous email and have not =
received any response until now, so I mention this question again.
>> in draft-ietf-rtcweb-data-channel-07, Section 4 says:
>>   Req. 4:   The application SHOULD be able to provide guidance as to
>>             the relative priority of each data channel relative to =
each
>>             other, and relative to the media streams.  This will
>>             interact with the congestion control algorithms.
>>=20
>> And in draft-ietf-rtcweb-data-protocol-03, DATA_CHANNEL_OPEN Message =
is defined as below
>> 0                   1                   2                   3
>>    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>   |  Message Type |  Channel Type |            Priority           |
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>   |                    Reliability Parameter                      |
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>   |         Label Length          |       Protocol Length         |
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>   \                                                               /
>>   |                             Label                             |
>>   /                                                               \
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>   \                                                               /
>>   |                            Protocol                           |
>>   /                                                               \
>>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>> Priority: 2 bytes (integer)
>>      The priority of the channel as described in
>>      [I-D.ietf-rtcweb-data-channel].  The higher the number, the =
lower
>>      the priority.
>>=20
>> Q7: It seems that for now there is no W3C API parameters to set the =
priority of data channel. I wonder whether the priority of the data =
channel is open to the application or is only decided by the browser. In =
other words, I want to know how the application be able to provide =
guidance as to the priority.
> You are right. The W3C API does not cover the priority yet. My =
understanding is that this is missing and needs to be added. I expect =
that the priority is settable via the JS API.
>=20
> Best regards
> Michael
>>=20
>> Best Regards,
>>=20
>> Jessie
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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Hi,
(as individual)

I have reviewed the WebRTC Data Channels draft
(draft-ietf-rtcweb-data-channel-07) and have some comments.

1. Section 1:
Non-media data types in the context of WebRTC are handled by using
   SCTP [RFC4960] encapsulated in DTLS [RFC6347].

There are many places in this draft where "media data" is a reference to
the RTP transport media. I would suggest to review this and consider to
change this to "RTP Transport media", or "RTP media". The reason is that
I expect also real-time media to flow over the data channel.

2. Section 1:
As the other SCTP extensions are discussed in the introduction, I wonder
why NADA and PR-Policy isn't.

3. Section 1:
Section 5 arguments SCTP over DTLS over UDP;

"arguments" looks strange in the above sentence fragment.

4. Section 3.1:
           Note that
           at any time there may be no media channels, or all media
           channels may be inactive, and that there may also be reliable
           data channels in use.

"no media channels" is a bit strange here as we not really call anything
regarding RTP channels. I would suggest to change this to "RTP Packet
Streams" or "RTP Stream".

5. Section 3.2:
   U-C 7:  Proxy browsing, where a browser uses data channels of a
           PeerConnection to send and receive HTTP/HTTPS requests and
           data, for example to avoid local Internet filtering or
           monitoring.

>From a data channel perspective this doesn't look strange at all,
however I get the shivers from the security implications of this. To my
understanding Peer A using Peer B to proxy browse must have B fetch all
elements that A needs into its sandbox and then transfer it to A over
the PeerConnections Data Channel. Thus, all you do are revealed to B,
massive privacy consideration. Or am I missing someway that A can tunnel
its HTTP request through B without having B see the content of those
requests, I am especially concerned with HTTPS.

If this use case is going to stay, I do want some security consideration
discussion around it.

6. Section 5:
"   o  The congestion control is modifiable for integration with media
      stream congestion control."

As this statement is only partial correct. There is no current mechanism
to negotiate congestion control, nor does it exist any alternative
defined SCTP congestion control. Thus, I would like to have this
sentence modified. I am however not certain to what the authors consider
important. It is the fact that it is possible in the future to define a
new CC algorithm and use that?

7. Section 5:
   o  provides privacy for the SCTP control information.

Is "privacy" really the right word here? Either being explicit, "source
authentication, integrity and confidentiality" or something like "security"?

8. section 5:

   DTLS implementations used for this stack SHOULD support controlling
   fields of the IP layer like the Don't Fragment (DF)-bit in case of
   IPv4 and the Differentiated Services Code Point (DSCP) field required
   for supporting [I-D.ietf-rtcweb-qos].

First of all this reference must be changed. What I am currently
uncertain of is what it should point on, most likely
draft-dhesikan-tsvwg-rtcweb-qos-04. But, I am not yet certain how the
scope is going to be divided between that document and
draft-ietf-rtcweb-transports.

9. Section 5:

The initial Path MTU
   at the IP layer MUST NOT exceed 1200 bytes for IPv4 and 1280 for
   IPv6.

I have to say that I think MUST NOT in the above is to strict. I think
SHOULD NOT is appropriate. The reason is that a deployment may actually
know that a larger initial MTU is better. This value may also change due
to how the networks develop in the future.

10. Section 5:

Since SCTP does not support the
   negotiation of a congestion control algorithm yet, alternate
   congestion controls SHOULD either only require a different sender
   side behavior using existing information carried in the association
   or need also specify a negotiation of of a congestion control
   algorithm.

First of all I don't like the SHOULD in this sentence.

Secondly, I think we need to get a good agreement and statement what to
do with SCTP's congestion control algorithm in the future. I don't like
this statement saying basically agree on anything and use that. I think
the IETF specification should specify something to use that we know is
safe to deploy. Thus, I would strongly prefer this to simply to be
changed to say that: We know we want a new congestion control that
better can handle interaction the the RTP packet streams and its
congestion control, but that will not be initially available. Thus use
the specified congestion control and consider these issues. And if a new
CC algo becomes available that can be used and should be negotiated
between the endpoints somehow.

11. Section 6.1:
   Once support for message interleaving as currently being discussed in
   [I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.

I don't know why you have formulated this, so awkward? NDATA is
currently listed as a normative reference. This document is going to
hand in the RFC-editor queue until it becomes available or an AD orders
some rewriting. Thus, why not simply  write a simpler statement making
clear that one SHOULD support it.

I would also note that there are disagreement between this and the text
in Section 6.6.:

To overcome this
   limitation, [I-D.ietf-tsvwg-sctp-ndata]  defines an extension to
   support message interleaving.  Once this extension is available, it
   MUST be used.

Thus pick the level of requirement and then simply state that with the
motivation why. Personally I think SHOULD would be acceptable. There are
clear motivation why (in 6.6) why one SHOULD implelement it, and the
basic functionality is still there if it is not supported and its usage
will be negotiated as SCTP association creation.

12. Section 6.2:
 Additionally,
   the negotiation SHOULD include some type of congestion control
   selection.

Another congestion control statement that I am not happy with. See issue
10. There is no default and it is also not clear on what to really
negotiate.

13. Section 6.3:

   SCTP defines a stream as a unidirectional logical channel existing
   within an SCTP association one to another SCTP endpoint.

The "one" looks spurious.

14. Section 6.4:
These priorities MUST NOT be strict priorities.

Ok, this is one part of the agreement from the earlier meeting. There
was also agreement on using weighted fair queuing to solve the
priorities. The inter stream priority handling needs much firmer
specification.

15. Section 6.5:
"This can be based on the DTLS role (the client picks
   even stream identifiers, the server odd stream identifiers) or done
   in a different way."

This is very unclear and you can't know what will happen. I think it
would be better to say: Unless otherwise defined or negotiated, the
streams are picked based on th DTLS role.  ...

16. Section 6.5:
"In addition to choosing a stream, the application SHOULD also
   inform the protocol of the options to use for sending messages."

So what is the application, and what is the protocol in this sentence?

17. Section 7.

I find this insufficient. I would this section to answer the following
questions:

1. Any security issues that arise from the combination of mechanism.
2. Any significant issue that one needs to know about the behavior of
the mechanisms, any of the normative specs that should be highlighted?
3. Clarification on what security properties one achieve with the
defined mechanism.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 20 February 2014 08:05, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> When I read this section, my immediate question was, doesn't RTP need
> corresponding protection. To my understanding, the answer would be yes,
> but it is more difficult to inject data and get a certain plaintext even
> without the SRTP encryption scrambling the payload part. Still, that
> assumes that an JS application, can write the payload data itself.
> Secondly, it does assume that one MUST use encryption in SRTP.


Yes, with WebAudio, I can construct an SRTP packet that contains
ciphertext that I can control, assuming that I know the session keys
(yes), and I'm prepared to do a lot of work.  SRTP doesn't have the IV
per-packet that DTLS does.  What you can't do is make it look like a
non-SRTP packet, the first byte, length and MAC would be extremely
difficult to control (for the MAC, that's more or less the point...),
if not impossible.

So the question is, is there an intermediary that might be tricked
into interpreting an SRTP packet in a way that might have negative
consequences.  When masking was added to websockets, it was because
the confusion was demonstrated as being possible: intermediaries were
searching for HTTP requests and responses in ways that were easily
exploited and then caching the results.  To a large extent, that
attack relied on the fact that TCP is a stream-based protocol - it
didn't matter where the request appeared.  In this case, the mapping
of SRTP onto UDP doesn't really lead to intermediaries looking for
goodies inside packets.

I'm not going to say that there is a risk, but I think that I'd be
willing to take the risk given the plethora of mitigating
circumstances here.


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On Fri, Feb 21, 2014 at 12:08 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-02-20 18:13, Ted Hardie wrote:
> > On Thu, Feb 20, 2014 at 8:05 AM, Magnus Westerlund
> > <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>=
>
> > wrote:
> >
> >     Hi,
> >
> >     I think a potential issue that isn't discussed in this document is
> the
> >     security threat of driving data volumes into a network beyond ones
> >     "fair" share. This would simply be to illustrate that communication
> >     consent is not sufficient, to protect other users of a shared netwo=
rk
> >     the WebRTC endpoint MUST prevent transmission of data volumes far
> >     outside of the fair share.
> >
> >
> >
> > Magnus, I'm not sure why you think that issue should be dealt with in
> > this document.
> > This document should deal with cases where a malicious script could sen=
d
> > transmissions
> > that are not intended or not welcome.  Bbut managing network capacity
> > for *intended*
> > and *welcome* transmissions is not a security concern.  It's a
> > congestion management
> > issue, at least as I see it.
> >
> > Am I missing something about your concern?
> >
>
> My concern is that a malicious individual wants to DDoS a particular
> network. I buy AD time from on of the network or in any other way ensure
> that the mailicious JS identifies when each client is topologically
> right for me to connect it to another client to drive traffic through my
> targets network. Then I ensure these clients establish a peer connection
> and I try to blast as much traffic through the network as possible.


Sorry for my apparent failure to understand here, but we're still dealing
with
traffic to which the parties consent, right?  That is, you're thinking of
malicious
JS that sends channels worth of nonsense to blast the network while
something
the user cares about happens? (Two-player flappy bird but with a terrabit
of nonsense
screaming in the background?)



> From
> the point of the attack this doesn't matter if is real-time media or
> data traffic. The point is to drive as much traffic into the network as
> possible. From that perspective any traffic that is sent in the context
> of a peerconnection needs to be congestion controlled. That at least
> keeps the traffic down to a "fair" share.
>
> We are mitigating this attack, however what I was really wondering over
> is if the security considerations document should make note that lack of
> appropriate congestion control is a security vulnerability and enables
> certain type of load attacks.
>
>
So my take as an individual is that congestion control is a known desirable
property, and that we're addressing its lack as we can.  Adding that note t=
o
the security document doesn't seem to me to add much to this story.

Ted




> Does this make my concern clearer?
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Fri, Feb 21, 2014 at 12:08 A=
M, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerl=
und@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</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 class=3D"">O=
n 2014-02-20 18:13, Ted Hardie wrote:<br>
&gt; On Thu, Feb 20, 2014 at 8:05 AM, Magnus Westerlund<br>
</div>&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.wes=
terlund@ericsson.com</a> &lt;mailto:<a href=3D"mailto:magnus.westerlund@eri=
csson.com">magnus.westerlund@ericsson.com</a>&gt;&gt;<br>
<div><div class=3D"h5">&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 Hi,<br>
&gt;<br>
&gt; =A0 =A0 I think a potential issue that isn&#39;t discussed in this doc=
ument is the<br>
&gt; =A0 =A0 security threat of driving data volumes into a network beyond =
ones<br>
&gt; =A0 =A0 &quot;fair&quot; share. This would simply be to illustrate tha=
t communication<br>
&gt; =A0 =A0 consent is not sufficient, to protect other users of a shared =
network<br>
&gt; =A0 =A0 the WebRTC endpoint MUST prevent transmission of data volumes =
far<br>
&gt; =A0 =A0 outside of the fair share.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Magnus, I&#39;m not sure why you think that issue should be dealt with=
 in<br>
&gt; this document.<br>
&gt; This document should deal with cases where a malicious script could se=
nd<br>
&gt; transmissions<br>
&gt; that are not intended or not welcome. =A0Bbut managing network capacit=
y<br>
&gt; for *intended*<br>
&gt; and *welcome* transmissions is not a security concern. =A0It&#39;s a<b=
r>
&gt; congestion management<br>
&gt; issue, at least as I see it.<br>
&gt;<br>
&gt; Am I missing something about your concern?<br>
&gt;<br>
<br>
</div></div>My concern is that a malicious individual wants to DDoS a parti=
cular<br>
network. I buy AD time from on of the network or in any other way ensure<br=
>
that the mailicious JS identifies when each client is topologically<br>
right for me to connect it to another client to drive traffic through my<br=
>
targets network. Then I ensure these clients establish a peer connection<br=
>
and I try to blast as much traffic through the network as possible. </block=
quote><div><br><div class=3D"gmail_default" style=3D"font-family:georgia,se=
rif">Sorry for my apparent failure to understand here, but we&#39;re still =
dealing with<br>
traffic to which the parties consent, right?=A0 That is, you&#39;re thinkin=
g of malicious<br></div><div class=3D"gmail_default" style=3D"font-family:g=
eorgia,serif">JS that sends channels worth of nonsense to blast the network=
 while something<br>
the user cares about happens? (Two-player flappy bird but with a terrabit o=
f nonsense<br>screaming in the background?)</div><br>=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
From<br>
the point of the attack this doesn&#39;t matter if is real-time media or<br=
>
data traffic. The point is to drive as much traffic into the network as<br>
possible. From that perspective any traffic that is sent in the context<br>
of a peerconnection needs to be congestion controlled. That at least<br>
keeps the traffic down to a &quot;fair&quot; share.<br>
<br>
We are mitigating this attack, however what I was really wondering over<br>
is if the security considerations document should make note that lack of<br=
>
appropriate congestion control is a security vulnerability and enables<br>
certain type of load attacks.<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:georgia,serif">So my take as an individual is that congestion control is a=
 known desirable<br>property, and that we&#39;re addressing its lack as we =
can.=A0 Adding that note to<br>
the security document doesn&#39;t seem to me to add much to this story.<br>=
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
Ted<br></div><br><br>=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

Does this make my concern clearer?<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Phone =A0<a href=3D"tel:%2B46=
%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Mobile <a href=3D"tel:%2B=
46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

--001a1132e4ccb7f6a004f2f44c4b--


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From: Martin Thomson <martin.thomson@gmail.com>
To: Ted Hardie <ted.ietf@gmail.com>
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On 21 February 2014 17:02, Ted Hardie <ted.ietf@gmail.com> wrote:
> Sorry for my apparent failure to understand here, but we're still dealing
> with
> traffic to which the parties consent, right?  That is, you're thinking of
> malicious
> JS that sends channels worth of nonsense to blast the network while
> something
> the user cares about happens? (Two-player flappy bird but with a terrabit of
> nonsense
> screaming in the background?)


Maybe Magnus is looking for the mechanism described in
http://tools.ietf.org/html/draft-thomson-mmusic-rtcweb-bw-consent-00,
which might allow for control below the limit that the congestion
control permits.  I basically abandoned this, because it's small
potatoes. It's trivial to revoke consent if you notice a misbehaving
peer and then you only have to wait until the consent timer runs out
on the sender.  30s.


From nobody Sat Feb 22 16:20:20 2014
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(Catching up before the meeting)
Here are some comments on the latest version. There don't necessarily 
relate to recent changes. Rather they come from my perspective trying to 
use this for CLUE, whether using a webrtc endpoint or not.

* Section 5:

This section says:

    Each SCTP user message contains a so called Payload Protocol
    Identifier (PPID) that is passed to SCTP by its upper layer and sent
    to its peer.  This value can be used to multiplex multiple protocols
    over a single SCTP association.  The sender provides for each
    protocol a specific PPID and the receiver can demultiplex the
    messages based on the received PPID.  The PPID is used to distinguish
    UTF-8 encoded user data and binary encoded userdata.  The Data
    Channel Establishment Protocol defined in
    [I-D.ietf-rtcweb-data-protocol] uses also a specific PPID to be
    distinguished from user data.

The language has proved extremely confusing to several people who 
haven't followed the development and discussion of this from the 
beginning. It is because of the extreme ambiguity in the use of 
"protocol". While I guess we can't rename PPID, the other language could 
be clarified.

The problem is that we have SCTP protocol, "WebRTC Data Channel 
Establishment Protocol", and within the DATA_CHANNEL_OPEN message of the 
WebRTC Data Channel Establishment Protocol we have a "Protocol" field.

We also have layering of protocols and implementations. So it isn't 
clear above who "sender" and "receiver" above apply to. (Could be code 
that is calling the SCTP stack, or code that is calling the "data 
channel stack", or the browser, or a javascript application on top of 
browser.

IIUC, javascript users don't have any visibility to the PPID. But others 
implementing data channels may.

How about:

    Each SCTP user message contains a so called Payload Protocol
    Identifier (PPID) that is passed to SCTP by the data channel layer
    and sent to its peer.  This value is used to multiplex WebRTC Data
    Channel Establishment Protocol messages (defined in [I-D.ietf-rtcweb-
    data-protocol]) with messages containing user data on a data
    channel. The PPID is also used to distinguish UTF-8 encoded user
    data and binary encoded user data.

Also in this section is Figure 2 that shows layering. I'm unclear if 
"WEBRTC DATA" in intended to include rtcweb-data-protocol or not. That 
is defined in a separate draft, is mandatory to support but optional to 
use. So it could go either way. Might be good to clarify:


                  +---------+
                  |CHANNEL  |
                  |PROTOCOL |
                  +---------+
                  |DATA     |
                  |CHANNEL  |
                  +---------+
                  | SCTP    |
    +-----------------------+
    | STUN | SRTP | DTLS    |
    +-----------------------+
    |         ICE           |
    +-----------------------+
    | UDP1 | UDP2 | ...     |
    +-----------------------+

* Section 6.5

This section contains:

    Data channels can be opened by using internal or external
    negotiation.  The details are out of scope of this document.

    A simple protocol for internal negotiation is specified in
    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.

But internal and external negotiation are not defined in this document.

I *thought* that internal negotiation was by definition negotiation by 
use of rtcweb-data-protocol. (draft-ejzak-mmusic-data-channel-sdpneg-00 
thinks so too, but calls it "in-band negotiation".)

There should be a good definition of these terms, or reference to one. 
And more discussion if there can be other kinds of internal negotiation. 
(If so, how would one be chosen?)

* Section 6.6:

Say:

    All data sent on a Channel in both directions MUST be sent over the
    underlying stream using the reliability defined when the Channel was
    opened unless the options are changed, or per-message options are
    specified by a higher level.

Is the recipient to consider it an error if messages are received with 
options different from those defined for the channel?

Also, is it an error if messages are received with a PPID that isn't 
specified in Section 8? (And what about PPID 50?)

How are such channel errors to be treated?

	Thanks,
	Paul


From nobody Sat Feb 22 19:07:11 2014
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A few comments on this draft:

* Section 5.1:

       DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
          a partial reliable in-order bi-directional Communication
          channel.  User messages might not be transmitted or
          retransmitted after a specified life-time given in milli-
          seconds in the Reliability Parameter.  This life-time starts
          when providing the user message to the Javascript engine.

The last sentence above is troubling. This protocol won't always be used 
via Javascript. Can we please have a definition that doesn't depend on 
that? Is the timing being done by SCTP, or are you assuming that a data 
channel layer on top of SCTP is doing this? Is it specific to SCEP, or 
is it still applicable when the channel has been established via 
external negotiation?

Does use of this option imply that retransmission continues until this 
time limit is reached? Or might it stop after some implementation 
defined number of retransmissions?

The description of the reliability parameter says:

       This field is ignored if a reliable channel is used.

IMO folk wisdom says that some specific value (e.g., 0) should be 
prescribed to be used in such cases. That makes things more 
deterministic and provides more flexibility in extending the protocol in 
the future.

The name "Protocol" (and "Protocol Length") here is troublesome, because 
it is ambiguous with other sorts of uses. (See my comment about this 
point earlier today wrt draft-ietf-rtcweb-data-channel-07.) Others 
(including draft-ejzak-mmusic-data-channel-sdpneg-00 and websockets) 
call this "subprotocol". Using that would make it a little less confusing.

Also, ISTM that all of these things are applicable to externally 
negotiated channels, and so ought to be defined by 
draft-ietf-rtcweb-data-channel. (But of course their use in the 
DATA_CHANNEL_OPEN message belongs here.)

* Section 8.4:

ISTM there ought to be a template of the minimum information that the 
specification for a registered (sub)protocol MUST (SHOULD?) include. E.g.,

- what Channel Type(s) are valid for this (sub)protocol
- A contact for more information about this protocol

	Thanks,
	Paul


From nobody Sun Feb 23 01:55:18 2014
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From: Barry Dingle <btdingle@gmail.com>
Date: Sun, 23 Feb 2014 20:54:47 +1100
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In reference to your first *Section 1 comments*, I agree that we need to
clearly differentiate the SRTP media and the SCTP media. (Note - SRTP - not
RTP !)

I agree that there could be real-time media that can be sent over SCTP
Channels are still meet the criteria for 'real-time' as being 'received
within a few hundred milliseconds of being generated' - see WebRTC Overview
section 2.4.

Maybe we need to differentiate the Channel types - and then refer to the
data going over those Channels. For instance,if we define "SRTP Channels"
and "SCTP Channels", we could then refer to 'SRTP Channel media' which
would have a specific meaning.

The best place for the definition of SRTP Channel and SCTP Channel is
probably in the WebRTC Overview document.

Cheers,
/Barry Dingle



On Sat, Feb 22, 2014 at 3:27 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
> (as individual)
>
> I have reviewed the WebRTC Data Channels draft
> (draft-ietf-rtcweb-data-channel-07) and have some comments.
>
> 1. Section 1:
> Non-media data types in the context of WebRTC are handled by using
>    SCTP [RFC4960] encapsulated in DTLS [RFC6347].
>
> There are many places in this draft where "media data" is a reference to
> the RTP transport media. I would suggest to review this and consider to
> change this to "RTP Transport media", or "RTP media". The reason is that
> I expect also real-time media to flow over the data channel.
>
> 2. Section 1:
> As the other SCTP extensions are discussed in the introduction, I wonder
> why NADA and PR-Policy isn't.
>
> 3. Section 1:
> Section 5 arguments SCTP over DTLS over UDP;
>
> "arguments" looks strange in the above sentence fragment.
>
> 4. Section 3.1:
>            Note that
>            at any time there may be no media channels, or all media
>            channels may be inactive, and that there may also be reliable
>            data channels in use.
>
> "no media channels" is a bit strange here as we not really call anything
> regarding RTP channels. I would suggest to change this to "RTP Packet
> Streams" or "RTP Stream".
>
> 5. Section 3.2:
>    U-C 7:  Proxy browsing, where a browser uses data channels of a
>            PeerConnection to send and receive HTTP/HTTPS requests and
>            data, for example to avoid local Internet filtering or
>            monitoring.
>
> >From a data channel perspective this doesn't look strange at all,
> however I get the shivers from the security implications of this. To my
> understanding Peer A using Peer B to proxy browse must have B fetch all
> elements that A needs into its sandbox and then transfer it to A over
> the PeerConnections Data Channel. Thus, all you do are revealed to B,
> massive privacy consideration. Or am I missing someway that A can tunnel
> its HTTP request through B without having B see the content of those
> requests, I am especially concerned with HTTPS.
>
> If this use case is going to stay, I do want some security consideration
> discussion around it.
>
> 6. Section 5:
> "   o  The congestion control is modifiable for integration with media
>       stream congestion control."
>
> As this statement is only partial correct. There is no current mechanism
> to negotiate congestion control, nor does it exist any alternative
> defined SCTP congestion control. Thus, I would like to have this
> sentence modified. I am however not certain to what the authors consider
> important. It is the fact that it is possible in the future to define a
> new CC algorithm and use that?
>
> 7. Section 5:
>    o  provides privacy for the SCTP control information.
>
> Is "privacy" really the right word here? Either being explicit, "source
> authentication, integrity and confidentiality" or something like
> "security"?
>
> 8. section 5:
>
>    DTLS implementations used for this stack SHOULD support controlling
>    fields of the IP layer like the Don't Fragment (DF)-bit in case of
>    IPv4 and the Differentiated Services Code Point (DSCP) field required
>    for supporting [I-D.ietf-rtcweb-qos].
>
> First of all this reference must be changed. What I am currently
> uncertain of is what it should point on, most likely
> draft-dhesikan-tsvwg-rtcweb-qos-04. But, I am not yet certain how the
> scope is going to be divided between that document and
> draft-ietf-rtcweb-transports.
>
> 9. Section 5:
>
> The initial Path MTU
>    at the IP layer MUST NOT exceed 1200 bytes for IPv4 and 1280 for
>    IPv6.
>
> I have to say that I think MUST NOT in the above is to strict. I think
> SHOULD NOT is appropriate. The reason is that a deployment may actually
> know that a larger initial MTU is better. This value may also change due
> to how the networks develop in the future.
>
> 10. Section 5:
>
> Since SCTP does not support the
>    negotiation of a congestion control algorithm yet, alternate
>    congestion controls SHOULD either only require a different sender
>    side behavior using existing information carried in the association
>    or need also specify a negotiation of of a congestion control
>    algorithm.
>
> First of all I don't like the SHOULD in this sentence.
>
> Secondly, I think we need to get a good agreement and statement what to
> do with SCTP's congestion control algorithm in the future. I don't like
> this statement saying basically agree on anything and use that. I think
> the IETF specification should specify something to use that we know is
> safe to deploy. Thus, I would strongly prefer this to simply to be
> changed to say that: We know we want a new congestion control that
> better can handle interaction the the RTP packet streams and its
> congestion control, but that will not be initially available. Thus use
> the specified congestion control and consider these issues. And if a new
> CC algo becomes available that can be used and should be negotiated
> between the endpoints somehow.
>
> 11. Section 6.1:
>    Once support for message interleaving as currently being discussed in
>    [I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.
>
> I don't know why you have formulated this, so awkward? NDATA is
> currently listed as a normative reference. This document is going to
> hand in the RFC-editor queue until it becomes available or an AD orders
> some rewriting. Thus, why not simply  write a simpler statement making
> clear that one SHOULD support it.
>
> I would also note that there are disagreement between this and the text
> in Section 6.6.:
>
> To overcome this
>    limitation, [I-D.ietf-tsvwg-sctp-ndata]  defines an extension to
>    support message interleaving.  Once this extension is available, it
>    MUST be used.
>
> Thus pick the level of requirement and then simply state that with the
> motivation why. Personally I think SHOULD would be acceptable. There are
> clear motivation why (in 6.6) why one SHOULD implelement it, and the
> basic functionality is still there if it is not supported and its usage
> will be negotiated as SCTP association creation.
>
> 12. Section 6.2:
>  Additionally,
>    the negotiation SHOULD include some type of congestion control
>    selection.
>
> Another congestion control statement that I am not happy with. See issue
> 10. There is no default and it is also not clear on what to really
> negotiate.
>
> 13. Section 6.3:
>
>    SCTP defines a stream as a unidirectional logical channel existing
>    within an SCTP association one to another SCTP endpoint.
>
> The "one" looks spurious.
>
> 14. Section 6.4:
> These priorities MUST NOT be strict priorities.
>
> Ok, this is one part of the agreement from the earlier meeting. There
> was also agreement on using weighted fair queuing to solve the
> priorities. The inter stream priority handling needs much firmer
> specification.
>
> 15. Section 6.5:
> "This can be based on the DTLS role (the client picks
>    even stream identifiers, the server odd stream identifiers) or done
>    in a different way."
>
> This is very unclear and you can't know what will happen. I think it
> would be better to say: Unless otherwise defined or negotiated, the
> streams are picked based on th DTLS role.  ...
>
> 16. Section 6.5:
> "In addition to choosing a stream, the application SHOULD also
>    inform the protocol of the options to use for sending messages."
>
> So what is the application, and what is the protocol in this sentence?
>
> 17. Section 7.
>
> I find this insufficient. I would this section to answer the following
> questions:
>
> 1. Any security issues that arise from the combination of mechanism.
> 2. Any significant issue that one needs to know about the behavior of
> the mechanisms, any of the normative specs that should be highlighted?
> 3. Clarification on what security properties one achieve with the
> defined mechanism.
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d044519a9989f7a04f30fd90f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div>In reference to your first <u>Section 1 comments=
</u>, I agree that we need to clearly differentiate the SRTP media and the =
SCTP media. (Note - SRTP - not RTP !)<br><br></div><div>I agree that there =
could be real-time media that can be sent over SCTP Channels are still meet=
 the criteria for &#39;real-time&#39; as being &#39;received within a few h=
undred milliseconds of being generated&#39; - see WebRTC Overview section 2=
.4. <br>


<br></div>Maybe we need to differentiate the Channel types - and then refer=
 to the data going over those Channels. For instance,if we define &quot;SRT=
P Channels&quot; and &quot;SCTP Channels&quot;, we could then refer to &#39=
;SRTP Channel media&#39; which would have a specific meaning. <br>


<br></div>The best place for the definition of SRTP Channel and SCTP Channe=
l is probably in the WebRTC Overview document. <br><div><div><div class=3D"=
gmail_extra"><br clear=3D"all"><div>Cheers,<br>/Barry Dingle<br><br></div>


<br><br><div class=3D"gmail_quote">On Sat, Feb 22, 2014 at 3:27 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wro=
te:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
(as individual)<br>
<br>
I have reviewed the WebRTC Data Channels draft<br>
(draft-ietf-rtcweb-data-channel-07) and have some comments.<br>
<br>
1. Section 1:<br>
Non-media data types in the context of WebRTC are handled by using<br>
=C2=A0 =C2=A0SCTP [RFC4960] encapsulated in DTLS [RFC6347].<br>
<br>
There are many places in this draft where &quot;media data&quot; is a refer=
ence to<br>
the RTP transport media. I would suggest to review this and consider to<br>
change this to &quot;RTP Transport media&quot;, or &quot;RTP media&quot;. T=
he reason is that<br>
I expect also real-time media to flow over the data channel.<br>
<br>
2. Section 1:<br>
As the other SCTP extensions are discussed in the introduction, I wonder<br=
>
why NADA and PR-Policy isn&#39;t.<br>
<br>
3. Section 1:<br>
Section 5 arguments SCTP over DTLS over UDP;<br>
<br>
&quot;arguments&quot; looks strange in the above sentence fragment.<br>
<br>
4. Section 3.1:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Note that<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0at any time there may be no media =
channels, or all media<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0channels may be inactive, and that=
 there may also be reliable<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0data channels in use.<br>
<br>
&quot;no media channels&quot; is a bit strange here as we not really call a=
nything<br>
regarding RTP channels. I would suggest to change this to &quot;RTP Packet<=
br>
Streams&quot; or &quot;RTP Stream&quot;.<br>
<br>
5. Section 3.2:<br>
=C2=A0 =C2=A0U-C 7: =C2=A0Proxy browsing, where a browser uses data channel=
s of a<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0PeerConnection to send and receive=
 HTTP/HTTPS requests and<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0data, for example to avoid local I=
nternet filtering or<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0monitoring.<br>
<br>
&gt;From a data channel perspective this doesn&#39;t look strange at all,<b=
r>
however I get the shivers from the security implications of this. To my<br>
understanding Peer A using Peer B to proxy browse must have B fetch all<br>
elements that A needs into its sandbox and then transfer it to A over<br>
the PeerConnections Data Channel. Thus, all you do are revealed to B,<br>
massive privacy consideration. Or am I missing someway that A can tunnel<br=
>
its HTTP request through B without having B see the content of those<br>
requests, I am especially concerned with HTTPS.<br>
<br>
If this use case is going to stay, I do want some security consideration<br=
>
discussion around it.<br>
<br>
6. Section 5:<br>
&quot; =C2=A0 o =C2=A0The congestion control is modifiable for integration =
with media<br>
=C2=A0 =C2=A0 =C2=A0 stream congestion control.&quot;<br>
<br>
As this statement is only partial correct. There is no current mechanism<br=
>
to negotiate congestion control, nor does it exist any alternative<br>
defined SCTP congestion control. Thus, I would like to have this<br>
sentence modified. I am however not certain to what the authors consider<br=
>
important. It is the fact that it is possible in the future to define a<br>
new CC algorithm and use that?<br>
<br>
7. Section 5:<br>
=C2=A0 =C2=A0o =C2=A0provides privacy for the SCTP control information.<br>
<br>
Is &quot;privacy&quot; really the right word here? Either being explicit, &=
quot;source<br>
authentication, integrity and confidentiality&quot; or something like &quot=
;security&quot;?<br>
<br>
8. section 5:<br>
<br>
=C2=A0 =C2=A0DTLS implementations used for this stack SHOULD support contro=
lling<br>
=C2=A0 =C2=A0fields of the IP layer like the Don&#39;t Fragment (DF)-bit in=
 case of<br>
=C2=A0 =C2=A0IPv4 and the Differentiated Services Code Point (DSCP) field r=
equired<br>
=C2=A0 =C2=A0for supporting [I-D.ietf-rtcweb-qos].<br>
<br>
First of all this reference must be changed. What I am currently<br>
uncertain of is what it should point on, most likely<br>
draft-dhesikan-tsvwg-rtcweb-qos-04. But, I am not yet certain how the<br>
scope is going to be divided between that document and<br>
draft-ietf-rtcweb-transports.<br>
<br>
9. Section 5:<br>
<br>
The initial Path MTU<br>
=C2=A0 =C2=A0at the IP layer MUST NOT exceed 1200 bytes for IPv4 and 1280 f=
or<br>
=C2=A0 =C2=A0IPv6.<br>
<br>
I have to say that I think MUST NOT in the above is to strict. I think<br>
SHOULD NOT is appropriate. The reason is that a deployment may actually<br>
know that a larger initial MTU is better. This value may also change due<br=
>
to how the networks develop in the future.<br>
<br>
10. Section 5:<br>
<br>
Since SCTP does not support the<br>
=C2=A0 =C2=A0negotiation of a congestion control algorithm yet, alternate<b=
r>
=C2=A0 =C2=A0congestion controls SHOULD either only require a different sen=
der<br>
=C2=A0 =C2=A0side behavior using existing information carried in the associ=
ation<br>
=C2=A0 =C2=A0or need also specify a negotiation of of a congestion control<=
br>
=C2=A0 =C2=A0algorithm.<br>
<br>
First of all I don&#39;t like the SHOULD in this sentence.<br>
<br>
Secondly, I think we need to get a good agreement and statement what to<br>
do with SCTP&#39;s congestion control algorithm in the future. I don&#39;t =
like<br>
this statement saying basically agree on anything and use that. I think<br>
the IETF specification should specify something to use that we know is<br>
safe to deploy. Thus, I would strongly prefer this to simply to be<br>
changed to say that: We know we want a new congestion control that<br>
better can handle interaction the the RTP packet streams and its<br>
congestion control, but that will not be initially available. Thus use<br>
the specified congestion control and consider these issues. And if a new<br=
>
CC algo becomes available that can be used and should be negotiated<br>
between the endpoints somehow.<br>
<br>
11. Section 6.1:<br>
=C2=A0 =C2=A0Once support for message interleaving as currently being discu=
ssed in<br>
=C2=A0 =C2=A0[I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be support=
ed.<br>
<br>
I don&#39;t know why you have formulated this, so awkward? NDATA is<br>
currently listed as a normative reference. This document is going to<br>
hand in the RFC-editor queue until it becomes available or an AD orders<br>
some rewriting. Thus, why not simply =C2=A0write a simpler statement making=
<br>
clear that one SHOULD support it.<br>
<br>
I would also note that there are disagreement between this and the text<br>
in Section 6.6.:<br>
<br>
To overcome this<br>
=C2=A0 =C2=A0limitation, [I-D.ietf-tsvwg-sctp-ndata] =C2=A0defines an exten=
sion to<br>
=C2=A0 =C2=A0support message interleaving. =C2=A0Once this extension is ava=
ilable, it<br>
=C2=A0 =C2=A0MUST be used.<br>
<br>
Thus pick the level of requirement and then simply state that with the<br>
motivation why. Personally I think SHOULD would be acceptable. There are<br=
>
clear motivation why (in 6.6) why one SHOULD implelement it, and the<br>
basic functionality is still there if it is not supported and its usage<br>
will be negotiated as SCTP association creation.<br>
<br>
12. Section 6.2:<br>
=C2=A0Additionally,<br>
=C2=A0 =C2=A0the negotiation SHOULD include some type of congestion control=
<br>
=C2=A0 =C2=A0selection.<br>
<br>
Another congestion control statement that I am not happy with. See issue<br=
>
10. There is no default and it is also not clear on what to really<br>
negotiate.<br>
<br>
13. Section 6.3:<br>
<br>
=C2=A0 =C2=A0SCTP defines a stream as a unidirectional logical channel exis=
ting<br>
=C2=A0 =C2=A0within an SCTP association one to another SCTP endpoint.<br>
<br>
The &quot;one&quot; looks spurious.<br>
<br>
14. Section 6.4:<br>
These priorities MUST NOT be strict priorities.<br>
<br>
Ok, this is one part of the agreement from the earlier meeting. There<br>
was also agreement on using weighted fair queuing to solve the<br>
priorities. The inter stream priority handling needs much firmer<br>
specification.<br>
<br>
15. Section 6.5:<br>
&quot;This can be based on the DTLS role (the client picks<br>
=C2=A0 =C2=A0even stream identifiers, the server odd stream identifiers) or=
 done<br>
=C2=A0 =C2=A0in a different way.&quot;<br>
<br>
This is very unclear and you can&#39;t know what will happen. I think it<br=
>
would be better to say: Unless otherwise defined or negotiated, the<br>
streams are picked based on th DTLS role. =C2=A0...<br>
<br>
16. Section 6.5:<br>
&quot;In addition to choosing a stream, the application SHOULD also<br>
=C2=A0 =C2=A0inform the protocol of the options to use for sending messages=
.&quot;<br>
<br>
So what is the application, and what is the protocol in this sentence?<br>
<br>
17. Section 7.<br>
<br>
I find this insufficient. I would this section to answer the following<br>
questions:<br>
<br>
1. Any security issues that arise from the combination of mechanism.<br>
2. Any significant issue that one needs to know about the behavior of<br>
the mechanisms, any of the normative specs that should be highlighted?<br>
3. Clarification on what security properties one achieve with the<br>
defined mechanism.<br>
<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287" target=
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F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
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rget=3D"_blank">+46 73 0949079</a><br>
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<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>

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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-security-06
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On 2014-02-22 02:14, Martin Thomson wrote:
> On 21 February 2014 17:02, Ted Hardie <ted.ietf@gmail.com> wrote:
>> Sorry for my apparent failure to understand here, but we're still dealing
>> with
>> traffic to which the parties consent, right?  That is, you're thinking of
>> malicious
>> JS that sends channels worth of nonsense to blast the network while
>> something
>> the user cares about happens? (Two-player flappy bird but with a terrabit of
>> nonsense
>> screaming in the background?)

Yes, that would be the attack. And the target of the attack is the
networks in the middle, not the endpoints, they are simply tools for this.

> 
> 
> Maybe Magnus is looking for the mechanism described in
> http://tools.ietf.org/html/draft-thomson-mmusic-rtcweb-bw-consent-00,
> which might allow for control below the limit that the congestion
> control permits.  I basically abandoned this, because it's small
> potatoes. It's trivial to revoke consent if you notice a misbehaving
> peer and then you only have to wait until the consent timer runs out
> on the sender.  30s.
> 
> 

My position is that the above mitigation is unlikely to have significant
impact on this attack. This as it is difficult to know what values to
set. They vary over time and depending on network attachment, what
services the user tries to use etc.

There are two reasons I wanted this attack to be considered. First, it
provides a clear requirement on having congestion control as first level
mitigation.

Secondly, I think this could become a significant issue as data channel
PeerConnections can be opened without user consent. A malicious JS that
is sufficient well spread with a well working find and connect could
establish a large mesh of peer connections that could come close to
saturate each endpoints local access link, resulting in very heavy loads
on the network, even with congestion control. With congestion control
you can likely prevent congestion collapse, but you should be fully
capable of driving a network into a state of "mostly useless",
especially a network suffering from buffer bloat inside of the ingress
nodes.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Mon Feb 24 05:18:38 2014
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Thread-Topic: A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

A few questions on  draft-ejzak-dispatch-webrtc-data-channel-sdpneg:

Q1:

Instead of defining a new SDP webrtc-DataChannel attribute, would it be pos=
sible to define the parameters as extension parameters to the SDP sctpmap a=
ttribute (yes, I know the ABNF currently does not allow that)?

Example:

a=3Dsctpmap:1000 webrtc-datachannel 1;stream=3D6;subprotocol=3D"CLUE"

I am not saying this would be good or bad - at this point I just want to un=
derstand whether it would work.


Q2:

Would it be possible, for subprotocol values, use the same IANA registry/va=
lues as for the WebRTC Data Channel Protocol?


Q3:

It would be good to clarify, if only one data channel is requested (or, eve=
n required), that the stream value must be the same in the Offer and Answer=
.


Q4:

I have issues with the max_retr, max_time and unordered parameters.

First, they seem to specify SENDING capabilities, rather than RECEIVING cap=
abilities.

Second, they seem to describe characteristics associated with the subprotoc=
ol, in which case they could be specified in the associated subprotocol spe=
cification.

Regards,

Christer

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">A few questions on&nbsp; draft-=
ejzak-dispatch-webrtc-data-channel-sdpneg:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q1:<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Instead of defining a new SDP w=
ebrtc-DataChannel attribute, would it be possible to define the parameters =
as extension parameters to the SDP sctpmap attribute (yes, I know the ABNF =
currently does not allow that)?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Example:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:65.2pt"><b><span lang=3D"EN-US"=
>a=3Dsctpmap:1000 webrtc-datachannel 1;stream=3D6;subprotocol=3D&#8221;CLUE=
&quot;<o:p></o:p></span></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I am not saying this would be g=
ood or bad &#8211; at this point I just want to understand whether it would=
 work.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q2:<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Would it be possible, for subpr=
otocol values, use the same IANA registry/values as for the WebRTC Data Cha=
nnel Protocol?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q3:<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">It would be good to clarify, if=
 only one data channel is requested (or, even required), that the stream va=
lue must be the same in the Offer and Answer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q4:<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have issues with the max_retr=
, max_time and unordered parameters.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">First, they seem to specify SEN=
DING capabilities, rather than RECEIVING capabilities.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Second, they seem to describe c=
haracteristics associated with the subprotocol, in which case they could be=
 specified in the associated subprotocol specification.<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

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Date: Mon, 24 Feb 2014 08:07:47 -0600
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On Mon, Feb 24, 2014 at 7:18 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> A few questions on  draft-ejzak-dispatch-webrtc-data-channel-sdpneg:
>
>
>
> *Q1:*
>
>
>
> Instead of defining a new SDP webrtc-DataChannel attribute, would it be
> possible to define the parameters as extension parameters to the SDP
> sctpmap attribute (yes, I know the ABNF currently does not allow that)?
>
>
>
> Example:
>
>
>
> *a=sctpmap:1000 webrtc-datachannel 1;stream=6;subprotocol="CLUE"*
>
>
>
> I am not saying this would be good or bad - at this point I just want to
> understand whether it would work.
>

It might work, but you would still need a different mechanism to negotiate
multiple data channels.  Why bother with two different formats when one
will do?


>
>
>
>
> *Q2:*
>
>
>
> Would it be possible, for subprotocol values, use the same IANA
> registry/values as for the WebRTC Data Channel Protocol?
>

This is really more a question for the data channel design itself.  A
decision has been made to use the PPID for framing rather than protocol
identification.  There is some discussion of this in another thread.


>
>
>
>
> *Q3:*
>
>
>
> It would be good to clarify, if only one data channel is requested (or,
> even required), that the stream value must be the same in the Offer and
> Answer.
>

I thought this also came up in an earlier thread and that it is documented
(or should be documented) in the core data channel docs.  My understanding
is that the stream id is always the same in both directions for a
particular data channel.


>
>
>
> *Q4:*
>
>
>
> I have issues with the max_retr, max_time and unordered parameters.
>
>
>
> First, they seem to specify SENDING capabilities, rather than RECEIVING
> capabilities.
>
>
>
> Second, they seem to describe characteristics associated with the
> subprotocol, in which case they could be specified in the associated
> subprotocol specification.
>

This should be a topic for discussion in London.  As it is currently
written, these parameters are of the "take it or leave it" variety, i.e.,
the answerer either accepts the options listed in the offer or rejects the
data channel outright.  This is true for the in-band control protocol
(please correct me if I am wrong on this), so we adopted the same approach
for the SDP-negotiated case.  Some have suggested that we should allow more
flexibility in negotiation of these parameters when using SDP.  If there is
general agreement that it should be done this way, I see no problem with
defining parameter negotiation rules for SDP in the next version of the
draft.

BR, Richard



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

--089e01228d68075bf004f3277f8e
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 Mon, Feb 24, 2014 at 7:18 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@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 lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">A few questions on&nbsp; draft-=
ejzak-dispatch-webrtc-data-channel-sdpneg:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q1:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Instead of defining a new SDP w=
ebrtc-DataChannel attribute, would it be possible to define the parameters =
as extension parameters to the SDP sctpmap attribute (yes, I know the ABNF =
currently does not allow that)?<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Example:<u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:65.2pt"><b><span lang=3D"EN-US"=
>a=3Dsctpmap:1000 webrtc-datachannel 1;stream=3D6;subprotocol=3D&rdquo;CLUE=
&quot;<u></u><u></u></span></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I am not saying this would be g=
ood or bad &ndash; at this point I just want to understand whether it would=
 work.</span></p></div></div></blockquote><div><br></div><div>It might work=
, but you would still need a different mechanism to negotiate multiple data=
 channels. &nbsp;Why bother with two different formats when one will do?</d=
iv>
<div>&nbsp;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"FI" link=3D"bl=
ue" vlink=3D"purple"><div><p class=3D"MsoNormal"><span lang=3D"EN-US"><u></=
u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q2:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Would it be possible, for subpr=
otocol values, use the same IANA registry/values as for the WebRTC Data Cha=
nnel Protocol?</span></p></div></div></blockquote><div><br></div><div>This =
is really more a question for the data channel design itself. &nbsp;A decis=
ion has been made to use the PPID for framing rather than protocol identifi=
cation. &nbsp;There is some discussion of this in another thread.</div>
<div>&nbsp;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"FI" link=3D"bl=
ue" vlink=3D"purple"><div><p class=3D"MsoNormal"><span lang=3D"EN-US"><u></=
u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q3:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">It would be good to clarify, if=
 only one data channel is requested (or, even required), that the stream va=
lue must be the same in the Offer and Answer.</span></p></div></div></block=
quote>
<div><br></div><div>I thought this also came up in an earlier thread and th=
at it is documented (or should be documented) in the core data channel docs=
. &nbsp;My understanding is that the stream id is always the same in both d=
irections for a particular data channel. &nbsp;&nbsp;</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 lang=3D"FI" link=3D"blue=
" vlink=3D"purple"><div><p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=
<u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q4:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have issues with the max_retr=
, max_time and unordered parameters.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">First, they seem to specify SEN=
DING capabilities, rather than RECEIVING capabilities.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Second, they seem to describe c=
haracteristics associated with the subprotocol, in which case they could be=
 specified in the associated subprotocol specification.</span></p></div></d=
iv>
</blockquote><div><br></div><div>This should be a topic for discussion in L=
ondon. &nbsp;As it is currently written, these parameters are of the &quot;=
take it or leave it&quot; variety, i.e., the answerer either accepts the op=
tions listed in the offer or rejects the data channel outright. &nbsp;This =
is true for the in-band control protocol (please correct me if I am wrong o=
n this), so we adopted the same approach for the SDP-negotiated case. &nbsp=
;Some have suggested that we should allow more flexibility in negotiation o=
f these parameters when using SDP. &nbsp;If there is general agreement that=
 it should be done this way, I see no problem with defining parameter negot=
iation rules for SDP in the next version of the draft.</div>
<div><br></div><div>BR, Richard</div><div><br></div><div>&nbsp;</div><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex"><div lang=3D"FI" link=3D"blue" vlink=3D"purple"><d=
iv><p class=3D"MsoNormal">
<span lang=3D"EN-US"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<span class=3D"HOEnZb">=
<font color=3D"#888888"><u></u><u></u></font></span></span></p><span class=
=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<u></u><u></u></span></=
p>
</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>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

>>Q1:
>>
>>=A0Instead of defining a new SDP webrtc-DataChannel attribute, would it b=
e possible to define the parameters as extension parameters to the SDP sctp=
map attribute (yes, I know the ABNF currently does not allow that)?
>>
>>Example:
>>=20
>>          a=3Dsctpmap:1000 webrtc-datachannel 1;stream=3D6;subprotocol=3D=
"CLUE"
>>
>> I am not saying this would be good or bad - at this point I just want to=
 understand whether it would work.
>
> It might work, but you would still need a different mechanism to negotiat=
e multiple data channels. =A0Why bother with two different formats when one=
 will do?

I am sure the syntax could be defined so that you could define multiple cha=
nnels. Or, you could use multiple sctpmap attributes.

Anyway, that is a syntax question, so we don't need to spend time on it now=
.
=A0
>> Q2:
>>=A0
>> Would it be possible, for subprotocol values, use the same IANA registry=
/values as for the WebRTC Data Channel Protocol?
>
> This is really more a question for the data channel design itself. =A0A d=
ecision has been made to use the PPID for framing rather than protocol iden=
tification. =A0There is some discussion of this in another thread.

I am not talking about the PPID - I am talking about the Protocol field in =
the DATA_CHANNEL_OPEN message, which (afaik) specifies the same thing as th=
e subprotocol parameter in your draft.

=A0
>> Q3:
>>=A0
>> It would be good to clarify, if only one data channel is requested (or, =
even required), that the stream value must be the same in the Offer and Ans=
wer.
>
> I thought this also came up in an earlier thread and that it is documente=
d (or should be documented) in the core data channel docs. =A0My understand=
ing is that the stream id is always the same in both directions for a parti=
cular data channel. =A0=A0

Correct. But, there should be an Offer/Answer procedure section in your dra=
ft, which shall say that the stream number must be the same. You can then r=
eference to the data channel doc for justification.

=A0
>> Q4:
>>
>> I have issues with the max_retr, max_time and unordered parameters.
>>=A0
>> First, they seem to specify SENDING capabilities, rather than RECEIVING =
capabilities.=20
>>=A0
>> Second, they seem to describe characteristics associated with the subpro=
tocol, in which case they could be specified in the associated subprotocol =
specification.
>
> This should be a topic for discussion in London. =A0As it is currently wr=
itten, these parameters are of the "take it or leave it" variety, i.e., the=
 answerer either accepts the options listed
> in the offer or rejects the data channel outright. =A0This is true for th=
e in-band control protocol (please correct me if I am wrong on this), so we=
 adopted the same approach for the=20
> SDP-negotiated case. =A0Some have suggested that we should allow more fle=
xibility in negotiation of these parameters when using SDP. =A0If there is =
general agreement that it should be
> done this way, I see no problem with defining parameter negotiation rules=
 for SDP in the next version of the draft.

The difference is that your draft is based on Offer/Answer, which is about =
indicating receiving capabilities (yes, I know we have broken that rule in =
the past).

Regards,

Christer

=A0


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Date: Mon, 24 Feb 2014 08:58:30 -0600
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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--089e013c6ae46aef8f04f32834ed
Content-Type: text/plain; charset=ISO-8859-1

On Mon, Feb 24, 2014 at 8:18 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Hi,
>
> >> Q2:
> >>
> >> Would it be possible, for subprotocol values, use the same IANA
> registry/values as for the WebRTC Data Channel Protocol?
> >
> > This is really more a question for the data channel design itself.  A
> decision has been made to use the PPID for framing rather than protocol
> identification.  There is some discussion of this in another thread.
>
> I am not talking about the PPID - I am talking about the Protocol field in
> the DATA_CHANNEL_OPEN message, which (afaik) specifies the same thing as
> the subprotocol parameter in your draft.
>

This issue did come up in another thread, and the answer is yes.  It should
probably be documented in the core data channel specs and just referenced
here.


>
>
> >> Q3:
> >>
> >> It would be good to clarify, if only one data channel is requested (or,
> even required), that the stream value must be the same in the Offer and
> Answer.
> >
> > I thought this also came up in an earlier thread and that it is
> documented (or should be documented) in the core data channel docs.  My
> understanding is that the stream id is always the same in both directions
> for a particular data channel.
>
> Correct. But, there should be an Offer/Answer procedure section in your
> draft, which shall say that the stream number must be the same. You can
> then reference to the data channel doc for justification.
>

Since there is no disagreement over how it should work, we can clarify as
necessary in a subsequent draft.


>
>
> >> Q4:
> >>
> >> I have issues with the max_retr, max_time and unordered parameters.
> >>
> >> First, they seem to specify SENDING capabilities, rather than RECEIVING
> capabilities.
> >>
> >> Second, they seem to describe characteristics associated with the
> subprotocol, in which case they could be specified in the associated
> subprotocol specification.
> >
> > This should be a topic for discussion in London.  As it is currently
> written, these parameters are of the "take it or leave it" variety, i.e.,
> the answerer either accepts the options listed
> > in the offer or rejects the data channel outright.  This is true for the
> in-band control protocol (please correct me if I am wrong on this), so we
> adopted the same approach for the
> > SDP-negotiated case.  Some have suggested that we should allow more
> flexibility in negotiation of these parameters when using SDP.  If there is
> general agreement that it should be
> > done this way, I see no problem with defining parameter negotiation
> rules for SDP in the next version of the draft.
>
> The difference is that your draft is based on Offer/Answer, which is about
> indicating receiving capabilities (yes, I know we have broken that rule in
> the past).
>

There are many examples of attributes that use different negotiation models
than this.  The only requirement is that the semantics of the negotiation
be defined in the document defining the attribute.

BR, Richard

--089e013c6ae46aef8f04f32834ed
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 Mon, Feb 24, 2014 at 8:18 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@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"><br>
Hi,<br>
<div class=3D""><br></div><div class=3D"">
&gt;&gt; Q2:<br>
&gt;&gt;=A0<br>
&gt;&gt; Would it be possible, for subprotocol values, use the same IANA re=
gistry/values as for the WebRTC Data Channel Protocol?<br>
&gt;<br>
&gt; This is really more a question for the data channel design itself. =A0=
A decision has been made to use the PPID for framing rather than protocol i=
dentification. =A0There is some discussion of this in another thread.<br>
<br>
</div>I am not talking about the PPID - I am talking about the Protocol fie=
ld in the DATA_CHANNEL_OPEN message, which (afaik) specifies the same thing=
 as the subprotocol parameter in your draft.<br></blockquote><div><br>
</div><div>This issue did come up in another thread, and the answer is yes.=
 =A0It should probably be documented in the core data channel specs and jus=
t referenced here.</div><div>=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div class=3D""><br>
=A0<br>
&gt;&gt; Q3:<br>
&gt;&gt;=A0<br>
&gt;&gt; It would be good to clarify, if only one data channel is requested=
 (or, even required), that the stream value must be the same in the Offer a=
nd Answer.<br>
&gt;<br>
&gt; I thought this also came up in an earlier thread and that it is docume=
nted (or should be documented) in the core data channel docs. =A0My underst=
anding is that the stream id is always the same in both directions for a pa=
rticular data channel. =A0=A0<br>

<br>
</div>Correct. But, there should be an Offer/Answer procedure section in yo=
ur draft, which shall say that the stream number must be the same. You can =
then reference to the data channel doc for justification.<br></blockquote>
<div><br></div><div>Since there is no disagreement over how it should work,=
 we can clarify as necessary in a subsequent draft.</div><div>=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">

<div class=3D""><br>
=A0<br>
&gt;&gt; Q4:<br>
&gt;&gt;<br>
&gt;&gt; I have issues with the max_retr, max_time and unordered parameters=
.<br>
&gt;&gt;=A0<br>
&gt;&gt; First, they seem to specify SENDING capabilities, rather than RECE=
IVING capabilities.<br>
&gt;&gt;=A0<br>
&gt;&gt; Second, they seem to describe characteristics associated with the =
subprotocol, in which case they could be specified in the associated subpro=
tocol specification.<br>
&gt;<br>
&gt; This should be a topic for discussion in London. =A0As it is currently=
 written, these parameters are of the &quot;take it or leave it&quot; varie=
ty, i.e., the answerer either accepts the options listed<br>
&gt; in the offer or rejects the data channel outright. =A0This is true for=
 the in-band control protocol (please correct me if I am wrong on this), so=
 we adopted the same approach for the<br>
&gt; SDP-negotiated case. =A0Some have suggested that we should allow more =
flexibility in negotiation of these parameters when using SDP. =A0If there =
is general agreement that it should be<br>
&gt; done this way, I see no problem with defining parameter negotiation ru=
les for SDP in the next version of the draft.<br>
<br>
</div>The difference is that your draft is based on Offer/Answer, which is =
about indicating receiving capabilities (yes, I know we have broken that ru=
le in the past).<br></blockquote><div><br></div><div>There are many example=
s of attributes that use different negotiation models than this. =A0The onl=
y requirement is that the semantics of the negotiation be defined in the do=
cument defining the attribute.</div>
<div><br></div><div>BR, Richard</div><div>=A0</div></div></div></div>

--089e013c6ae46aef8f04f32834ed--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

>>>> Q2:
>>>>=A0
>>>> Would it be possible, for subprotocol values, use the same IANA regist=
ry/values as for the WebRTC Data Channel Protocol?
>>>
>>> This is really more a question for the data channel design itself. =A0A=
 decision has been made to use the PPID for framing rather than protocol id=
entification. =A0There is some discussion of this in another thread.
>> I am not talking about the PPID - I am talking about the Protocol field =
in the DATA_CHANNEL_OPEN message, which (afaik) specifies the same thing as=
 the subprotocol parameter in your draft.
>
> This issue did come up in another thread, and the answer is yes. =A0It sh=
ould probably be documented in the core data channel specs and just referen=
ced here.

Good. Sorry if I missed the previous discussion on the topic.
=A0

>>>> Q3:
>>>>=A0
>>>> It would be good to clarify, if only one data channel is requested (or=
, even required), that the stream value must be the same in the Offer and A=
nswer.
>>>
>>> I thought this also came up in an earlier thread and that it is documen=
ted (or should be documented) in the core data channel docs. =A0My understa=
nding is that the stream id is always the same in both directions for a par=
ticular data channel. =A0=A0
>> Correct. But, there should be an Offer/Answer procedure section in your =
draft, which shall say that the stream number must be the same. You can the=
n reference to the data channel doc for justification.
>>
> Since there is no disagreement over how it should work, we can clarify as=
 necessary in a subsequent draft.
=A0
Good.

=A0
>>>> Q4:
>>>>
>>>> I have issues with the max_retr, max_time and unordered parameters.
>>>>=A0
>>>> First, they seem to specify SENDING capabilities, rather than RECEIVIN=
G capabilities.
>>>>=A0
>>>> Second, they seem to describe characteristics associated with the subp=
rotocol, in which case they could be specified in the associated subprotoco=
l specification.
>>>
>>> This should be a topic for discussion in London. =A0As it is currently =
written, these parameters are of the "take it or leave it" variety, i.e., t=
he answerer either accepts the options listed
>>> in the offer or rejects the data channel outright. =A0This is true for =
the in-band control protocol (please correct me if I am wrong on this), so =
we adopted the same approach for the
>>> SDP-negotiated case. =A0Some have suggested that we should allow more f=
lexibility in negotiation of these parameters when using SDP. =A0If there i=
s general agreement that it should be
>>> done this way, I see no problem with defining parameter negotiation rul=
es for SDP in the next version of the draft.
>> The difference is that your draft is based on Offer/Answer, which is abo=
ut indicating receiving capabilities (yes, I know we have broken that rule =
in the past).
>
> There are many examples of attributes that use different negotiation mode=
ls than this. =A0The only requirement is that the semantics of the negotiat=
ion be defined in the document defining the attribute.

I still don't think the information belong to SDP - they define protocol ch=
aracteristics. The protocol specification should define re-transmission tim=
ers etc.

Anyway, we can argue about it later. At this moment I think the focus shoul=
d be on whether to move forward with the mechanism to begin with, and the m=
ain feature of the mechanism - which is about negotiating usage of channels=
 :)

Regards,

Christer


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This draft seems like a good start to address the issue of interoperability
of audio codecs in WebRTC. As has been pointed out in previous postings,
AMR, AMR-WB and G.722 are particularly attractive codecs which should be
included, but I note that the authors have left the opportunity for
including others.

 

...Paul


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>This draft =
seems like a good start to address the issue of interoperability of =
audio codecs in WebRTC. As has been pointed out in previous postings, =
AMR, AMR-WB and G.722 are particularly attractive codecs which should be =
included, but I note that the authors have left the opportunity for =
including others.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>...Paul<o:p></o:p></p></div></body></html>
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From nobody Mon Feb 24 08:32:20 2014
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Date: Mon, 24 Feb 2014 17:32:14 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Review comments on draft-ietf-rtcweb-data-protocol-03
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Hi,
(as individual)

I have reviewed the WebRTC Data Channel Establishment protocol
(draft-ietf-rtcweb-data-protocol-03) and have some comments:

1. Section 4:
Shouldn't this section discuss the priority field?

2. Section 4:

The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers.

Isn't this unnecessary using the vague word of WebRTC instead of simply
pointing to the DTLS roles of the established data channel?

3. Section 5.1:
      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
         a partial reliable in-order bi-directional Communication
         channel.  User messages might not be transmitted or
         retransmitted after a specified life-time given in milli-
         seconds in the Reliability Parameter.  This life-time starts
         when providing the user message to the Javascript engine.

I think the use of "Javascript engine" in the last sentence is unclear.
Can we provide a implementation neutral definition, like "This life-time
starts when requesting transmission of the user message."?

4. Section 5.1:
   Label: Variable Length (sequence of characters)
      The name of the channel.  This may be an empty string.

   Protocol: Variable Length (sequence of characters)
      The protocol for the channel.  If this is an empty string the
      protocol us unspecified.  If it is an non-empty string, it
      specifies an IANA-registered protocol (see Section 8.4).

Both of these fields are strings, shouldn't a particular encoding be
specified here? Like UTF-8. Secondly, what values are allowed, the full
set of Unicode?

5. Section 6:
All Data Channel Establishment Protocol messages MUST be sent
   requesting ordered delivery and using reliable transmission.

I wonder of the use of requesting ordered delivery and using reliable
transmission, from an SCTP stream perspective, wouldn't using in both
places be appropriate? Or is how object which has been requested to be
transmitted unordered interact in SCTP with the ordered ones?

6. Section 7:

I think this section can be beefed up a bit. First make clear that the
Data Channel's required usage of DTLS ensures that the message integrity
and possible source authentication as well as confidentiality. Then
going over any security risks with a malicous peer using this protocol.
Can a malicous side screw up the peer using this protocol? What are the
implications?

7. Section 8.2:

This registry reserves 0xff without any motivation, please add one.

8. Section 8.3:
If I define a channel behavior that allows mixed ordered and unordered,
how would I register this?

9. Section 8.4:

I have a suggestion for this registries name, instead of "Protocol
   Registry" I propose  "DCEP Protocol Identification Registry"

10. Section 8.4:

There is missing any specification of what type of strings I may
register. Also, no length limitation, although the protocol allows at
maximum of 255 octets of encoded characters.

11. This comments concerns the relation to the Data Channel
specification. So my interpretation is that the WebRTC Data channel
draft has become both the base for this specification as well as the one
that specifies that the DCEP shall be implemented and supported. For
WebRTC I don't think this matters much, but if someone likes to re-use
the basic Data Channel specification, this structure makes it more
difficult and have no need for the bi-directional negotiated parameter
settings that the DCEP provides. In that case one have to sub-set the
data channel specification. I wonder if it would be better to move some
normative statements around, so that the chain of implementation
requirement goes draft-ietf-rtcweb-transports ->
draft-ietf-rtcweb-data-protocol -> draft-ietf-rtcweb-data-channel thus
providing a cleaner and more modular build up of the protocols.

This a suggestion based on that it looks like others, like CLUE is going
to use the Data Channel specification, but I am uncertain if they want
DCEP or not.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From nobody Mon Feb 24 08:54:48 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

Section 4.1.1 says:

"o  "max-bundle": The application will BUNDLE all of its media streams
      on a single transport.  All streams other than the first will be
      marked as bundle-only.  This policy aims to minimize candidate
      gathering and maximize multiplexing, at the cost of less
      compatibility with legacy endpoints."

Q1:

I think it would be good to clarify that the application will BUNDLE all of=
 its media streams *THAT THE APPLICATION IS ABLE TO DE-MULTIPLEX*.

Q2:

The text says that all streams other than the first will be marked as bundl=
e-only. But, will the application have any clue which media type that first=
 stream will be? Should we e.g. specify that it MUST be audio (if audio is =
enabled)?

Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Section 4.1.1 says:<o:p></o:p><=
/span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&#8220;o&nbsp; &quot;max-bundle=
&quot;: <b>The application will BUNDLE all of its media streams<o:p></o:p><=
/b></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; on a single transport.</span></b><span lang=3D"EN-US">&nbsp; All streams=
 other than the first will be<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
marked as bundle-only.&nbsp; This policy aims to minimize candidate<o:p></o=
:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
gathering and maximize multiplexing, at the cost of less<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>compatibility with legacy endpoints.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q1:<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I think it would be good to cla=
rify that the application will BUNDLE all of its media streams *<b>THAT THE=
 APPLICATION IS ABLE TO DE-MULTIPLEX</b>*.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q2:<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The text says that all streams =
other than the first will be marked as bundle-only. But, will the applicati=
on have any clue which media type that first stream will be? Should we e.g.=
 specify that it MUST be audio (if audio
 is enabled)?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

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From nobody Mon Feb 24 09:02:52 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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Date: Mon, 24 Feb 2014 17:02:44 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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From nobody Mon Feb 24 09:07:48 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: JSEP-06: How to create the second offer where only the port numbers are changed?
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Subject: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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Hi,

As specified in BUNDLE, before the remote peer has indicated support of BUN=
DLE, all m- lines (including those within to a bundle group) must have uniq=
ue port numbers.

Then, when the remote peer has indicated support of BUNDLE, a second offer =
(called the BAS offer) needs to be sent, in which the same port number is a=
ssigned to all m- lines within a bundle group.

My question is: how do I create the BAS offer? I don't want to change anyth=
ing - I just want to get an SDP with the BUNDLE port assigned to all m- lin=
es. Is there a flag in createOffer() for that?

Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As specified in BUNDLE, before =
the remote peer has indicated support of BUNDLE, all m- lines (including th=
ose within to a bundle group) must have unique port numbers.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Then, when the remote peer has =
indicated support of BUNDLE, a second offer (called the BAS offer) needs to=
 be sent, in which the same port number is assigned to all m- lines within =
a bundle group.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My question is: how do I create=
 the BAS offer? I don&#8217;t want to change anything &#8211; I just want t=
o get an SDP with the BUNDLE port assigned to all m- lines. Is there a flag=
 in createOffer() for that?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<o:p></o:p></span></p>
</div>
</body>
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On 24 February 2014 02:05, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> My position is that the above mitigation is unlikely to have significant
> impact on this attack.

That was my sense too.

> There are two reasons I wanted this attack to be considered. First, it
> provides a clear requirement on having congestion control as first level
> mitigation.

Congestion control.  Check.

> Secondly, I think this could become a significant issue as data channel
> PeerConnections can be opened without user consent. A malicious JS that
> is sufficient well spread with a well working find and connect could
> establish a large mesh of peer connections that could come close to
> saturate each endpoints local access link, resulting in very heavy loads
> on the network, even with congestion control. With congestion control
> you can likely prevent congestion collapse, but you should be fully
> capable of driving a network into a state of "mostly useless",
> especially a network suffering from buffer bloat inside of the ingress
> nodes.

So the concern is that even with congestion control, a single bad
actor can use more than their "fair" share of network resources.

Sadly, this is already true on the web.

No harm in writing down the potential.  Are you looking for anything
more than that?


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On 24 February 2014 08:54, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> The text says that all streams other than the first will be marked as
> bundle-only. But, will the application have any clue which media type that
> first stream will be? Should we e.g. specify that it MUST be audio (if audio
> is enabled)?

I'd prefer not.  I'm OK with having a recommendation (or RECOMMENDED),
but not MUST.

If anything, the idea here is to select the media section that is
*most important*.  That's going to be application-specific.  Common
usages might have audio being the most important, but that's not a
universal principal.

MUST would also require a higher degree of specification.  Do you
prefer video over application?  Does it matter which application?
What if video is more important?


From nobody Mon Feb 24 11:09:21 2014
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On Feb 23, 2014, at 4:07 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> A few comments on this draft:
>=20
> * Section 5.1:
>=20
>      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
>         a partial reliable in-order bi-directional Communication
>         channel.  User messages might not be transmitted or
>         retransmitted after a specified life-time given in milli-
>         seconds in the Reliability Parameter.  This life-time starts
>         when providing the user message to the Javascript engine.
>=20
> The last sentence above is troubling. This protocol won't always be =
used via Javascript. Can we please have a definition that
OK. What about replacing the last sentence with

This life-time starts when providing the user
message to the protocol stack.

> doesn't depend on that? Is the timing being done by SCTP, or are you =
assuming that a data channel layer on top of SCTP is doing this? Is it =
specific to SCEP, or is it still applicable when the channel has been =
established via external negotiation?
It is actually the time when the message is provided to the SCTP stack, =
but I think
there might be some code running between the SCTP code and the user code =
(let it be
JS or anything else). I'm assuming that there is no substantial =
buffering happening
there.
>=20
> Does use of this option imply that retransmission continues until this =
time limit is reached? Or might it stop after some implementation =
defined number of retransmissions?
The only way the retransmissions don't happen until the time limit is =
reached, is that
SCTP decides that the association is broken because of excessive =
retransmissions. =20
>=20
> The description of the reliability parameter says:
>=20
>      This field is ignored if a reliable channel is used.
>=20
> IMO folk wisdom says that some specific value (e.g., 0) should be =
prescribed to be used in such cases. That makes things more =
deterministic and provides more flexibility in extending the protocol in =
the future.
What about:

For reliable channels this field MUST be set to 0 on the sending side
and MUST be ignored no the receiving side.

>=20
> The name "Protocol" (and "Protocol Length") here is troublesome, =
because it is ambiguous with other sorts of uses. (See my comment about =
this point earlier today wrt draft-ietf-rtcweb-data-channel-07.) Others =
(including draft-ejzak-mmusic-data-channel-sdpneg-00 and websockets) =
call this "subprotocol". Using that would make it a little less =
confusing.
In the W3C specification it is also called protocol, but the text talks =
about sub-protocol. Maybe we should do
the same here.
>=20
> Also, ISTM that all of these things are applicable to externally =
negotiated channels, and so ought to be defined by =
draft-ietf-rtcweb-data-channel. (But of course their use in the =
DATA_CHANNEL_OPEN message belongs here.)
Reliability is discussed there, but I agree, we need to discuss label =
and protocol there, too.
>=20
> * Section 8.4:
>=20
> ISTM there ought to be a template of the minimum information that the =
specification for a registered (sub)protocol MUST (SHOULD?) include. =
E.g.,
>=20
> - what Channel Type(s) are valid for this (sub)protocol
I assumed all...
> - A contact for more information about this protocol
You need to provide a reference for the protocol. Isn't that enough?

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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> >>>> Q4:
> >>>>
> >>>> I have issues with the max_retr, max_time and unordered parameters.
> >>>>
> >>>> First, they seem to specify SENDING capabilities, rather than RECEIV=
ING
> capabilities.
> >>>>
> >>>> Second, they seem to describe characteristics associated with the
> subprotocol, in which case they could be specified in the associated
> subprotocol specification.
> >>>
> >>> This should be a topic for discussion in London. =A0As it is currentl=
y
> written, these parameters are of the "take it or leave it" variety, i.e.,
> the answerer either accepts the options listed
> >>> in the offer or rejects the data channel outright. =A0This is true fo=
r the
> in-band control protocol (please correct me if I am wrong on this), so we
> adopted the same approach for the
> >>> SDP-negotiated case. =A0Some have suggested that we should allow more
> flexibility in negotiation of these parameters when using SDP. =A0If ther=
e is
> general agreement that it should be
> >>> done this way, I see no problem with defining parameter negotiation
> rules for SDP in the next version of the draft.
> >> The difference is that your draft is based on Offer/Answer, which is
> about indicating receiving capabilities (yes, I know we have broken that
> rule in the past).
> >
> > There are many examples of attributes that use different negotiation
> models than this. =A0The only requirement is that the semantics of the
> negotiation be defined in the document defining the attribute.
>=20
> I still don't think the information belong to SDP - they define protocol
> characteristics. The protocol specification should define re-transmission
> timers etc.
>=20
> Anyway, we can argue about it later. At this moment I think the focus sho=
uld
> be on whether to move forward with the mechanism to begin with, and the m=
ain
> feature of the mechanism - which is about negotiating usage of channels :=
)

[Raju] Sorry to continue this discussion here instead of London, which I wo=
n't be attending.

I also believe that we do need these max_retr, max_time and unordered in th=
e SDP.
These map to the data channel init parameters specified in http://dev.w3.or=
g/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel . In case of DCP th=
ese are used by "5.1.  DATA_CHANNEL_OPEN Message" of http://tools.ietf.org/=
html/draft-ietf-rtcweb-data-protocol-03=20
Combination of these parameters will determine the type of=20
DATA_CHANNEL_OPEN open that will be sent.

For non-DCP external negotiation use case, these values have to be communic=
ated to the peer. Since we want to closely follow DCP, which allows only on=
e set of these values to be negotiated for both the directions, SDP offer/a=
nswer will do that same. Receiving end should accept these values, give it =
to data channel stack and echo them back to the offerer.
This way in external negotiation case as well, similar to DCP, both the pee=
rs will have same set of data channel attributes (reliable or partial relia=
ble, ordered or unordered, timed or # of retransmission).

We will make the following changes, if agreed by others, in the next draft:
1. Add a new "reliable/partial reliable" SDP attribute, which is missing no=
w.
   This is needed to map to the exact data channel type using external nego=
tiation.
2. Map these SDP attributes to relevant DCP and PeerConnection sections.
   Need to make updates to SDP whenever DCP and/or PeerConnection is change=
d.
3. Specify either max_retr (retransmit data channel) or max_time (timed dat=
a channel)  =20
   present, but NOT both.

Best Regards
Raju


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] JSEP-06: max-bundle policy questions
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Hi Martin,

I buy the argument that different applications may have different prioritie=
s regarding media type, so a "MUST audio" is not a good idea.

But, should the application then be able to indicate which media type (or, =
even which media source) shall be considered "first"? The browser probably =
has no clue?

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Martin Thomson <martin.thomson@gmail.com> wrote:


On 24 February 2014 08:54, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> The text says that all streams other than the first will be marked as
> bundle-only. But, will the application have any clue which media type tha=
t
> first stream will be? Should we e.g. specify that it MUST be audio (if au=
dio
> is enabled)?

I'd prefer not.  I'm OK with having a recommendation (or RECOMMENDED),
but not MUST.

If anything, the idea here is to select the media section that is
*most important*.  That's going to be application-specific.  Common
usages might have audio being the most important, but that's not a
universal principal.

MUST would also require a higher degree of specification.  Do you
prefer video over application?  Does it matter which application?
What if video is more important?


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On 24 February 2014 11:24, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> But, should the application then be able to indicate which media type (or, even which media source) shall be considered "first"? The browser probably has no clue?

I think that Justin has proposed a way to identify tracks that should
not be bundled.  I recall no objections to this [1].  I'd assume that
with only one track being marked that way, that track could be the one
that gets ports and the others not.


[1]... though I can't find anything in the API spec, and we did have
the discussion at an IETF meeting and nowhere else.  Maybe the W3C
don't know about it yet.


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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-data-protocol-03 - Protocol IANA registration information
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Hi,

>> * Section 8.4:
>>=20
>> ISTM there ought to be a template of the minimum information that the=20
>> specification for a registered (sub)protocol MUST (SHOULD?) include.=20
>> E.g.,
>>=20
>> - what Channel Type(s) are valid for this (sub)protocol
>> I assumed all...
>> - A contact for more information about this protocol
> You need to provide a reference for the protocol. Isn't that enough?

The question is what information that reference needs to contain.

For example, if a protocol requires reliability, the reference needs to spe=
cify that a reliable data channel must be opened.=20

The protocol may also define procedures regarding who is responsible for op=
ening the data channel.=20

Etc etc etc.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Richard Ejzak" <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

I don't see why an SDP based mechanism by default must provide the same cap=
abilities as the DCP. If you need to communicate everything that DCP provid=
es, then you use DCP. Otherwise you may use SDP (or whatever other external=
 mechanism).

OR, you can define a mechanism to communicate low-level information (re-tra=
nsmission timer values etc) within the protocol that you are going to trans=
port on top of the data channel.

Anyway, can you give me an example of a case where you want to use SDP, and=
 where you need to negotiate the re-transmission values etc?

Regards,

Christer


-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Makaraju, Maridi Raju (Raju) [mailto:Raju.Makaraju@alcatel=
-lucent.com]=20
L=E4hetetty: 24. helmikuuta 2014 21:16
Vastaanottaja: Christer Holmberg; Richard Ejzak
Kopio: rtcweb@ietf.org
Aihe: RE: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-chan=
nel-sdpneg-00

> >>>> Q4:
> >>>>
> >>>> I have issues with the max_retr, max_time and unordered parameters.
> >>>>
> >>>> First, they seem to specify SENDING capabilities, rather than=20
> >>>> RECEIVING
> capabilities.
> >>>>
> >>>> Second, they seem to describe characteristics associated with the
> subprotocol, in which case they could be specified in the associated=20
> subprotocol specification.
> >>>
> >>> This should be a topic for discussion in London. =A0As it is=20
> >>> currently
> written, these parameters are of the "take it or leave it" variety,=20
> i.e., the answerer either accepts the options listed
> >>> in the offer or rejects the data channel outright. =A0This is true=20
> >>> for the
> in-band control protocol (please correct me if I am wrong on this), so=20
> we adopted the same approach for the
> >>> SDP-negotiated case. =A0Some have suggested that we should allow=20
> >>> more
> flexibility in negotiation of these parameters when using SDP. =A0If=20
> there is general agreement that it should be
> >>> done this way, I see no problem with defining parameter=20
> >>> negotiation
> rules for SDP in the next version of the draft.
> >> The difference is that your draft is based on Offer/Answer, which=20
> >> is
> about indicating receiving capabilities (yes, I know we have broken=20
> that rule in the past).
> >
> > There are many examples of attributes that use different negotiation
> models than this. =A0The only requirement is that the semantics of the=20
> negotiation be defined in the document defining the attribute.
>=20
> I still don't think the information belong to SDP - they define=20
> protocol characteristics. The protocol specification should define=20
> re-transmission timers etc.
>=20
> Anyway, we can argue about it later. At this moment I think the focus=20
> should be on whether to move forward with the mechanism to begin with,=20
> and the main feature of the mechanism - which is about negotiating=20
> usage of channels :)

[Raju] Sorry to continue this discussion here instead of London, which I wo=
n't be attending.

I also believe that we do need these max_retr, max_time and unordered in th=
e SDP.
These map to the data channel init parameters specified in http://dev.w3.or=
g/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel . In case of DCP th=
ese are used by "5.1.  DATA_CHANNEL_OPEN Message" of http://tools.ietf.org/=
html/draft-ietf-rtcweb-data-protocol-03
Combination of these parameters will determine the type of DATA_CHANNEL_OPE=
N open that will be sent.

For non-DCP external negotiation use case, these values have to be communic=
ated to the peer. Since we want to closely follow DCP, which allows only on=
e set of these values to be negotiated for both the directions, SDP offer/a=
nswer will do that same. Receiving end should accept these values, give it =
to data channel stack and echo them back to the offerer.
This way in external negotiation case as well, similar to DCP, both the pee=
rs will have same set of data channel attributes (reliable or partial relia=
ble, ordered or unordered, timed or # of retransmission).

We will make the following changes, if agreed by others, in the next draft:
1. Add a new "reliable/partial reliable" SDP attribute, which is missing no=
w.
   This is needed to map to the exact data channel type using external nego=
tiation.
2. Map these SDP attributes to relevant DCP and PeerConnection sections.
   Need to make updates to SDP whenever DCP and/or PeerConnection is change=
d.
3. Specify either max_retr (retransmit data channel) or max_time (timed dat=
a channel)  =20
   present, but NOT both.

Best Regards
Raju


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

> Hi,
>=20
> I don't see why an SDP based mechanism by default must provide the same
[Raju] Because, whether browser based or not each side will use a data chan=
nel stack. Independent of DCP used or not, the data channel stack has to ha=
ve these attributes to do various order/unordered, reliab/unreliable, retra=
nsmit/timed modes.
With external negotiation, the same SDP will be used to negotiate these val=
ues (similar to what DCP data channel open does).
This way the only difference between external based negotiation vs. DCP is =
"DCP" itself, neither more nor less; rest of the attributes of data channel=
 are exactly identical on both ends. This is what we want to achieve with e=
xternal negotiation.


> capabilities as the DCP. If you need to communicate everything that DCP
> provides, then you use DCP. Otherwise you may use SDP (or whatever other
> external mechanism).

[Raju] Using DCP means the intermediate elements have no ("easy") visibilit=
y into what channels are being negotiated; such visibility is needed for va=
rious reasons like to have interworking functionality with other protocols,=
 such as MSRP; or regulatory needs; or value added services with billing im=
plications etc.

>=20
> OR, you can define a mechanism to communicate low-level information (re-
> transmission timer values etc) within the protocol that you are going to
> transport on top of the data channel.
[Raju] On top of data channel standard IANA registered protocols, such as M=
SRP, will typically be used; so modifying such protocols for this use case =
is not an option.

>=20
> Anyway, can you give me an example of a case where you want to use SDP, a=
nd
> where you need to negotiate the re-transmission values etc?

[Raju] A simple example would be 2 browsers talking thru an intermedia prox=
y, which wants to know what protocols are being negitiated and used as part=
 of the session.
1. Calling client creates data channel (using http://dev.w3.org/2011/webrtc=
/editor/webrtc.html#idl-def-RTCDataChannel) with desired attributes and set=
s negotiated=3Dtrue. So, calling browser saves the attributes for the data =
channel but won't use DCP.
2. These attributes are sent via SDP to peer client (via proxy).
3. Peer client sets these same attributes while creating data channel and s=
ets negotiated=3Dtrue but won't do DCP.
4. Peer client echo the same attributes back in SDP answer.

Now, data channel stacks on both ends will use same attributes, similar to =
DCP use case, with the only difference being DCP is not used here.

Best Regards
Raju


From nobody Mon Feb 24 14:10:18 2014
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Cc: 'Colin Perkins' <csp@csperkins.org>
Subject: [rtcweb]  [tram] Payload Types assignments
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I suggest to the RTCWEB WG that the below from the September and October
discussions on the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html is
introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC 5285, to =
allow:

=20

(1) WebRTC applications to directly convey QoS related real-time traffic
info to the network at points where RTP flow is directed to by TRAM =
Milstone
3, to be used by *any network element implementing any suitable QoS =
methods
for the particular network* for=20

(2) *all* WebRTC browsers *and* clients, under *all* OSs, and *all* =
current
and future IP network, to achieve best QoE=20

(3) *without* having to force WebRTC into application specific networks
(such as IMS) instead of using the Internet (including OTT).

=20

The only further activity required, is to call for ISPs=92 to review =
whether
the traffic information transferred by RFC 5285 is sufficient for =
current
and future needs in their network as suggested in below repeated
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have an RFC in place.

=20

With TRAM milestone 3 also place,=20

market forces will drive ISPs and browser makers to implement just this,
without even having it MUST-established.

=20

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

=20

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
Stahl
Skickat: den 22 oktober 2013 16:37
Till: 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus Westerlund'
Kopia: 'Colin Perkins'
=C4mne: [rtcweb] [avtext] Payload Types assignments was Re: SV: [mmusic] =
WGLC
of draft-ietf-rtcweb-use-cases-and-requirements-11

=20

Harald, I mostly agree with the quality requirements of different =
real-time
traffic that the WebRTC browser/application may use. But rather than =
asking
the application, let's convey the bandwidth and priority requirements to =
the
network. Just like with the Payload type (that is hard to squeeze that
information into) it must be visible to the network (and not changed by =
the
network, like diffserv bits are). Such marking must also be available =
for
incoming traffic, which is especially important in RSVP type of =
networks,
that has to reserve bandwidth for it. =20

=20

There is actually a good way to show these needs to the network (without
using the PT, or diffserv bits, which aren=92t sufficient anyway).=20

=20

Let's use the RTP header extension field that also is visible outside =
the
encrypted payload. A week ago came
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that
outlines the usage of the extension field for classification of traffic!
This document does not yet outline what to put in there and how to =
encode it
though.

=20

Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 =
discusses
other webrtc usages of the RTP header extension in 5.2 (there can be =
many
header extensions according to RFC 5285) and in 9 there is "WebRTC Use =
of
RTP: Future Extensions".

=20

So, it looks obvious to use the RTP header extension to show the
characteristics and bandwidth requirements to the network. It should not
introduce any backward incompatibilities either.

=20

Such marking is done in every RTP packet so it can be set individually =
for
each stream and could even be changed during a session (e.g. when =
limiting
the bandwidth based on RTCP feedback). RFC 5286 also specifies how RTP
extension header usage can be negotiated in SDP. I think this could be
easily done by the WebRTC browser for "all current and future needs" if
properly specified now.

=20

I suggest that two parameters (e.g. two bytes each) are encoded into the =
RTP
header extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each quality e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

=20

Please note the totally different requirements a diffserv and an RSVP
network have to know, so let=92s put all into these bytes. (E.g. a =
diffserv
network don't need the bandwidth usage, but RSVP reservation networks =
(e.g.
cable and 3G/4G OTT) do. There one should initially reserve the maximum
bandwidth indicated, but can later re-reserve.)

=20

/Karl

=20

PS Microsoft seems to have done work in this field, defining a =
proprietary
attribute =93MS Service Quality=94;=20

However that seems to apply to the TURN server allocation request and =
would
therefore:

--- Apply to the whole UDP flow, and could not be set for each stream
individually (with different requirements), and

--- Does not handle the bandwidth requirement for incoming real-time =
traffic
(required to reserve in RSVP type of networks)

However the quality attributes conveyed and their encoding may  be
considered.

=20

This is 2.2.2.19 MS-Service Quality Attribute from=20

http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20

=20

MS-Service Quality Attribute

The MS-Service Quality attribute is used to convey information about the
data stream that the protocol client is intending to transfer over an
allocated port. The protocol client SHOULD<21> include this attribute as
part of an Allocate request message. A TURN server SHOULD use the
information in this attribute to make decisions about resource =
allocation,
bandwidth prioritization, and data delivery methods. If the attribute is =
not
present in the Allocate request message, the TURN server SHOULD assume =
that
the data stream is audio with best effort delivery. The format of this
attribute is as follows...=20

...

The following stream types are supported in this extension. All other =
stream
types are reserved for future use.

=A7 "0x0001": Audio

=A7 "0x0002": Video

=A7 "0x0003": Supplemental Video

=A7 "0x0004": Data

Service Quality (2 bytes): The service quality level required by the
protocol client for the stream.

The following service quality levels are supported in this extension. =
All
other service quality levels are reserved for future use.

=A7 "0x0000": Best effort delivery.

=A7 "0x0001": Reliable delivery.

=20

=20

-----Ursprungligt meddelande-----

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Harald
Alvestrand

Skickat: den 8 oktober 2013 13:01

Till: rtcweb@ietf.org

=C4mne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] WGLC =
of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20

On 10/08/2013 09:17 AM, Karl Stahl wrote:

> Hej Magnus,

>=20

>> Also, are you really interested in knowing that it is VP9 vs H.264,=20

>> isn't

> the questions this is video of this priority that is important?

>> I think you need to more carefully consider what are the goals you=20

>> try to

> achieve them.

>=20

> Actually, my concern is to get an idea of the maximum bandwidth that=20

> could be required for a WebRTC (ICE) setup media flow. Both voice and=20

> video should be prioritized over data (their individual priority is of =


> less importance as long as there is sufficient bandwidth for both).

=20

You don't know that without knowing what the application is for.

In, for instance, a shooter game with voice backchannels, the movement =
and
event information (data) is MORE time sensitive than the voice data.

=20

>=20

> With diffserv you don=92t need to know the bandwidth requirement, but=20

> with RSVP reservation (like in cable and mobile networks) you need to=20

> know how much to reserve. Voice is like 100's kbit/s, video VP8 or=20

> H.264 is like 3,5 mbps.

=20

Again, without knowing the application, you don't know that.

The application could decide to use QCIF or HD, and the bandwidth =
variation
of screencast (semi-static with sudden, large changes) is completely
different from that of a talking head, which is again completely =
different
from a high-movement scene.

=20

>=20

> To add to the complication of codec variants, the video codecs in=20

> question for WebRTC have variable bandwidth, and when there is a poor=20

> connection we see Chrome reducing the video window size to reduce the
bandwidth used...

>=20

> I think the payload type field at best can reflect a maximum bandwidth =


> to initially reserve bandwidth for, and thereafter make new=20

> reservations if the bandwidth changes during the call. So could we=20

> change RTP to show maximum bandwidth instead of payload type in that=20

> field outside the encrypted payload :) ... Or maybe that is not a =
joke?

=20

I think these ruminations only lead to one conclusion:

=20

You can't tell what the needed bandwidth is up front without asking the
application.

You can't tell what the right priority ranking is without asking the
application.

=20

If you need to know the bandwidth or the priority up front, the =
application
has to tell you. Anything else is pure heuristics.

=20

_______________________________________________

rtcweb mailing list

 <mailto:rtcweb@ietf.org> rtcweb@ietf.org

 <https://www.ietf.org/mailman/listinfo/rtcweb>
https://www.ietf.org/mailman/listinfo/rtcweb


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
suggest to the RTCWEB WG that the below from the September and October =
discussions on the relevant </span><span lang=3DEN-US>[rtcweb] [avtext] =
[mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
is introduced into </span><span lang=3DEN-US>draft-ietf-rtcweb-rtp-usage =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
r usage of RFC 5285, to allow:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) WebRTC applications to directly convey QoS related real-time traffic =
info to the network at points where RTP flow is directed to by TRAM =
Milstone 3, to be used by *<b>any network element implementing any =
suitable QoS methods for the particular network</b>* for =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* clients, under *<b>all</b>* =
OSs, and *<b>all</b>* current and future IP network, to achieve best QoE =
<o:p></o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to force WebRTC into application specific networks (such as IMS) =
instead of using the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only further activity required, is to call for ISPs&#8217; to review =
whether the traffic information transferred by RFC 5285 is sufficient =
for current and future needs in their network as suggested in below =
repeated <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have an RFC in =
place.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place, <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just this, =
without even having it MUST-established.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>F=F6r =
</b>Karl Stahl<br><b>Skickat:</b> den 22 oktober 2013 =
16:37<br><b>Till:</b> 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus =
Westerlund'<br><b>Kopia:</b> 'Colin Perkins'<br><b>=C4mne:</b> [rtcweb] =
[avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p></di=
v></div><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Harald, I mostly agree with the quality requirements of =
different real-time traffic that the WebRTC browser/application may use. =
But rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. &nbsp;<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>There is actually a good way to =
show these needs to the network (without using the PT, or diffserv bits, =
which aren&#8217;t sufficient anyway). <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Let's use the RTP header =
extension field that also is visible outside the encrypted payload. A =
week ago came <a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">h=
ttp://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a> =
&nbsp;that outlines the usage of the extension field for classification =
of traffic! This document does not yet outline what to put in there and =
how to encode it though.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Today's <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http:/=
/tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a> discusses other =
webrtc usages of the RTP header extension in 5.2 (there can be many =
header extensions according to RFC 5285) and in 9 there is &quot;WebRTC =
Use of RTP: Future Extensions&quot;.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>So, it looks obvious to use the =
RTP header extension to show the characteristics and bandwidth =
requirements to the network. It should not introduce any backward =
incompatibilities either.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Such marking is done in every =
RTP packet so it can be set individually for each stream and could even =
be changed during a session (e.g. when limiting the bandwidth based on =
RTCP feedback). RFC 5286 also specifies how RTP extension header usage =
can be negotiated in SDP. I think this could be easily done by the =
WebRTC browser for &quot;all current and future needs&quot; if properly =
specified now.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>I suggest that two parameters (e.g. two bytes each) are =
encoded into the RTP header extension:<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>A) The maximum bandwidth =
requirement: Two bytes could contain everything from some bps for =
real-time text to Gbps for future 3D supersize telepresence&#8230; on a =
logarithmic scale.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>B) The quality characteristics for the stream, with the =
highest bit set to 1, we could allocate a bit each quality =
e.g:<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Best Effort, Audio, Video, Supplemental Video, Gaming, =
Data, Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable =
Delivery, Prioritize X, Variation Y, that could be combined as required =
to describe the stream.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>And with highest bit set to 0, =
there could instead be a number for special usage that does not fit the =
general description of the individual bits.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Please note the totally =
different requirements a diffserv and an RSVP network have to know, so =
let&#8217;s put all into these bytes. (E.g. a diffserv network don't =
need the bandwidth usage, but RSVP reservation networks (e.g. cable and =
3G/4G OTT) do. There one should initially reserve the maximum bandwidth =
indicated, but can later re-reserve.)<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>/Karl<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>PS </span><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>Microsoft seems =
to have done work in this field, defining a proprietary attribute =
&#8220;MS Service Quality&#8221;; <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>However that =
seems to apply to the TURN server allocation request and would =
therefore:<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>--- Apply to =
the whole UDP flow, and could not be set for each stream individually =
(with different requirements), and<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>--- Does not =
handle the bandwidth requirement for incoming real-time traffic =
(required to reserve in RSVP type of networks)<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>However the =
quality attributes conveyed and their encoding may &nbsp;be =
considered.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p=
></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:black'>This is 2.2.2.19 MS-Service Quality Attribute from =
</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:black'><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx" =
title=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).=
aspx">http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).asp=
x</a>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:black'>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>MS-Service =
Quality Attribute</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>The MS-Service =
Quality attribute is used to convey information about the data stream =
that the protocol client is intending to transfer over an allocated =
port. The protocol client SHOULD&lt;21&gt; include this attribute as =
part of an Allocate request message. A TURN server SHOULD use the =
information in this <span =
style=3D'background:yellow;mso-highlight:yellow'>attribute to make =
decisions about resource allocation, bandwidth prioritization, and data =
delivery methods</span>. If the attribute is not present in the Allocate =
request message, the TURN server SHOULD assume that the data stream is =
audio with best effort delivery. The format of this attribute is as =
follows... </span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>...</span></em><=
span lang=3DEN-US style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>The following =
stream types are supported in this extension. All other stream types are =
reserved for future use.</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0001&quo=
t;: Audio</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0002&quo=
t;: Video</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0003&quo=
t;: Supplemental Video</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0004&quo=
t;: Data</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>Service Quality =
(2 bytes): The service quality level required by the protocol client for =
the stream.</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>The following =
service quality levels are supported in this extension. All other =
service quality levels are reserved for future use.</span></em><span =
lang=3DEN-US style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0000&quo=
t;: Best effort delivery.</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol;color:black'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&quot;0x0001&quo=
t;: Reliable delivery.</span></em><span lang=3DEN-US =
style=3D'color:black'><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:black'>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<o:p></o:p></p><p =
class=3DMsoPlainText>Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] F=F6r Harald Alvestrand<o:p></o:p></p><p =
class=3DMsoPlainText>Skickat: den 8 oktober 2013 13:01<o:p></o:p></p><p =
class=3DMsoPlainText>Till: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>=C4mne: Re: [rtcweb] Payload =
Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>On 10/08/2013 09:17 AM, Karl =
Stahl wrote:<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; Hej Magnus,<o:p></o:p></span></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; Also, are you really =
interested in knowing that it is VP9 vs H.264, <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; =
isn't<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; the questions this is video of this priority that is =
important?<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; I think you need to more carefully consider what =
are the goals you <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; try to<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; achieve =
them.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; Actually, my concern is to =
get an idea of the maximum bandwidth that <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; could be required for a =
WebRTC (ICE) setup media flow. Both voice and <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; video should be prioritized =
over data (their individual priority is of <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; less importance as long as =
there is sufficient bandwidth for both).<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You don't know that without =
knowing what the application is for.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>In, for instance, a shooter game =
with voice backchannels, the movement and event information (data) is =
MORE time sensitive than the voice data.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; With diffserv you =
don&#8217;t need to know the bandwidth requirement, but =
<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
with RSVP reservation (like in cable and mobile networks) you need to =
<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
know how much to reserve. Voice is like 100's kbit/s, video VP8 or =
<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
H.264 is like 3,5 mbps.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Again, without knowing the =
application, you don't know that.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>The application could decide to =
use QCIF or HD, and the bandwidth variation of screencast (semi-static =
with sudden, large changes) is completely different from that of a =
talking head, which is again completely different from a high-movement =
scene.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; To add to the complication =
of codec variants, the video codecs in <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; question for WebRTC have =
variable bandwidth, and when there is a poor <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; connection we see Chrome =
reducing the video window size to reduce the bandwidth =
used...<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; I think the payload type =
field at best can reflect a maximum bandwidth <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; to initially reserve =
bandwidth for, and thereafter make new <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; reservations if the =
bandwidth changes during the call. So could we <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; change RTP to show maximum =
bandwidth instead of payload type in that <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; field outside the encrypted =
payload :) ... Or maybe that is not a joke?<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>I think these ruminations only =
lead to one conclusion:<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You can't tell what the needed =
bandwidth is up front without asking the =
application.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>You can't tell what the right priority ranking is without =
asking the application.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>If you need to know the =
bandwidth or the priority up front, the application has to tell you. =
Anything else is pure heuristics.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>_______________________________________________<o:p></o:p></=
span></p><p class=3DMsoPlainText><span lang=3DEN-US>rtcweb mailing =
list<o:p></o:p></span></p><p class=3DMsoPlainText><a =
href=3D"mailto:rtcweb@ietf.org"><span =
lang=3DEN-US>rtcweb@ietf.org</span></a><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoPlainText><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span =
lang=3DEN-US>https://www.ietf.org/mailman/listinfo/rtcweb</span></a><span=
 lang=3DEN-US><o:p></o:p></span></p></div></body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Pal Martinsen \(palmarti\)'" <palmarti@cisco.com>, <tram@ietf.org>, <rtcweb@ietf.org>
References: <20140214030712.30321.21888.idtracker@ietfa.amsl.com> <CAKhHsXGzA=ZTFGTK7ht9hQbfG70iqKrDtxrZCdQNNMzBYZCk8A@mail.gmail.com> <530604ea.c5bf440a.5cfd.ffffde18SMTPIN_ADDED_BROKEN@mx.google.com> <CAKhHsXGsp6ma6Ko9op+YFRqSGM_Ex-jFo_fjz69rN0SEHfNK2A@mail.gmail.com> <CALDtMrKb3_38Rs0vaGnpEvNvTYz8YUTo89STvLJNXfkfdipDSQ@mail.gmail.com> <93BEDDC39A54294B9E78C7860516FA4724AA4422@AZ-US1EXMB06.global.avaya.com> <B7FA2629-6D48-4569-BB62-56395C3EE4BC@cisco.com>
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Cc: 'Oleg Moskalenko' <mom040267@gmail.com>, "'Yoakum, John H \(John\)'" <yoakum@avaya.com>
Subject: Re: [rtcweb] [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
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Hi P=E5l,

=20

You did not comment nor answer, in response to any of:

http://www.ietf.org/mail-archive/web/tram/current/msg00275.html=20

http://www.ietf.org/mail-archive/web/tram/current/msg00273.html=20

whether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the
application/browser to transfer relevant real-time traffic information
(types and bandwidth) into every RTP package by using the already IETF
standardized RTP extension header RCF 5285, which could be used by any
current or future Internet (including OTT) network where WebRTC =
real-time
traffic may flow. It seems to have been standardized already 2008 to =
allow
such usage.

=20

QoS related step C) draft-martinsen-tram-discuss can then be taken out =
of
TRAM, and step D) (that was never intended to be handled by TRAM anyway)
will be handled elsewhere.

=20

I have repeated my suggestion to RTCWEB WG from the October discussions =
on
the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html to
introduce the QoS related step D) into draft-ietf-rtcweb-rtp-usage
suggesting usage of RFC 5285, where traffic information in RTP packets
simply would be filled by any browser under any OS.

=20

Without step B), TRAM =93Enforcing the real-time traffic through the
offered/discovered TURN servers=94, the real-time traffic may happen =
through
the IP default gateways often congested by data traffic and QoS =
insensitive
streaming video and file sharing. Without specific QoS methods at those
points, the network raw bandwidth capacity may have to be 10-folded to
achieve sufficient QoE when WebRTC usage becomes popular. Methods like
DISCUSS addresses such things occurring when STUN instead TURN is used =
(by
allowing flows into such congestion points), but only provides direct
traffic info for outgoing (not incoming) real-time traffic and locally,
therefore are not even applicable to reservation type of networks like =
Cable
Networks and Mobile OTT.=20

=20

That would leave certain network types to investing in raw bandwidth
increase, instead of simply borrowing the bandwidth (from much larger =
data
traffic and QoS insensitive streaming video and file sharing at no =
adverse
effect) and making that borrowed no-cost bandwidth very valuable for the
carriers when delivering to their customers.

I know a few of you Cisco guys are among the ones that have been =
fighting
against the general =93it is all about bandwidth=94 and =93it will go =
away with
time=94-attitude within IETF work, e.g. see
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html, but =
the
pointers given in the current QoS discussion within TRAM:=20

=20

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

=20

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. All networks will *also be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

Newer, fiber only type of networks, can still borrow bandwidth at no =
extra
cost, *by proprietary usage* of RFC 5285 by browsers.=20

=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may has to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

=20

Further explainations about QoS methods and their implementation were =
given
a few days ago in:=20

http://www.ietf.org/mail-archive/web/tram/current/msg00274.html=20

=20

(i), (ii) and (iii) will *seriously delay* usage of telepresence capable =
and
quality demanding WebRTC (bandwidth being around 30 times higher over =
best
effort Internet, than telephony over quality managed networks) and will
*vastly increase cost* for ISPs to offer WebRTC with good QoE.

=20

Below you are giving pointers saying =93They are currently working on a
problem-statement draft and a use-case draft, any input to those would =
be
very helpful.=94 Those are unnecessary, risking leading to (i), (ii) and =
(iii)
above.

=20

So are pointers such as: =93RMCAT where congestion avoidance for RTP is =
being
developed=94. TRAMs Milestone 3 is for the purpose of directing =
real-time
traffic where congestion control isn=92t already in place. Using RFC =
5285
available since 2008, to fill traffic information in RTP packets (step =
D))
is probably a necessity for most future =93congestion control=94 =
discussed.

=20

This chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues
to:=20

http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos=20

focusing on DSCP-mapping without even mentioning RFC 5285 available =
since
2008, to fill traffic information in RTP packets where it  is a =
necessity
and will allow:=20

(1) applications to directly convey QoS related real-time traffic info =
to
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to
be used by *any network element implementing any suitable QoS methods =
for
the particular network* for=20

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks=20

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of using the Internet (including OTT).

=20

The only activity required, is to call for ISPs=92 review whether the =
traffic
information transferred by RFC 5285 is sufficient for current and future
needs in their network as suggested in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have the RFC in place.

With TRAM milestone 3 also place,=20

market forces will drive ISPs and browser makers to implement just that,
without even having it MUST-established.

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

Fr=E5n: tram [mailto:tram-bounces@ietf.org] F=F6r Pal Martinsen =
(palmarti)
Skickat: den 21 februari 2014 10:37
Till: tram@ietf.org
Kopia: Karl Stahl; Oleg Moskalenko; Alan Johnston; Yoakum, John H (John)
=C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt

=20

Hi,

=20

I agree the full QoS discussion should _not_ happen in TRAM. If you are
interested in helping out in that area I suggest you looking into the =
AEON
mailing list at: https://www.ietf.org/mailman/listinfo/aeon . They are
currently working on a problem-statement draft and a use-case draft, any
input to those would be very helpful.
(http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01,
http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00).

=20

That said, STUN have a few nice characteristics that makes it a perfect
candidate for transporting some of the QoS information.  IMHO that would =
be
extending the STUN spec and should be within the TRAM charter.  The main
goal of draft-martinsen-tram-discuss was to show how already existing =
QoS
mechanisms could be transported with STUN to provide more value, and to
start the discussion if TRAM is the appropriate place to have those on =
the
wire format discussions.

=20

.-.

P=E5l-Erik

=20

=20

=20

=20

=20

=20

On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <yoakum@avaya.com> =
wrote:





+1

I fully agree with the comments that QOS should be a low priority for =
the
initial focus of the TRAM efforts.  There are other groups doing QOS =
work
and frankly I engage in WebRTC multimedia interactions daily over the
Internet, enterprise VPNs, and various combinations and seldom suffer
egregious quality issues.  I am more concerned about carriers doing =
things
to regulate or degrade WebRTC flows than a failure of existing Internet
mechanisms to enable them.

=20

Significant focus on QOS before we better enable TURN to be easily used =
in a
browser environment taking advantage or normal web characteristics (as
opposed to historic telephony constructs) would seem to be highly
distracting at this point.

=20

=20

Cheers,

John

=20

AVAYA
1.919.425.8446

=20

From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
Sent: Thursday, February 20, 2014 12:43 PM
To: Alan Johnston
Cc: Karl Stahl; tram@ietf.org
Subject: Re: [tram] Fwd: I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

=20

=20

On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <
<mailto:alan.b.johnston@gmail.com> alan.b.johnston@gmail.com> wrote:

=20

=20

=20

Personally, I am not sure how much QoS is actually in scope for TRAM. =
Have
you been following RMCAT where congestion avoidance for RTP is being
developed?  I see some overlap in your goals and the goals of that work.

-

=20

=20

I'd concentrate on the TURN application-level functionality, for now, =
and
I'd leave QoS for the future discussions.




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

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 P=E5l,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Yo=
u did not comment nor answer, in response to any =
of:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00273.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00273.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>wh=
ether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the =
application/browser to transfer relevant real-time traffic information =
(types and bandwidth) into every RTP package by using the already IETF =
standardized RTP extension header RCF 5285, which could be used by any =
current or future Internet (including OTT) network where WebRTC =
real-time traffic may flow. It seems to have been standardized already =
2008 to allow such usage.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Qo=
S related step C) </span><span lang=3DEN-US>draft-martinsen-tram-discuss =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ca=
n then be taken out of TRAM, and step D) (that was never intended to be =
handled by TRAM anyway) will be handled =
elsewhere.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have repeated my suggestion to RTCWEB WG from the October discussions on =
the relevant </span><span lang=3DEN-US>[rtcweb] [avtext] =
[mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
to introduce the QoS related step D) into </span><span =
lang=3DEN-US>draft-ietf-rtcweb-rtp-usage s</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ug=
gesting usage of RFC 5285, where traffic information in RTP packets =
simply would be filled by any browser under any OS.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
thout step B),</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
TRAM</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
&#8220;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>En=
forcing the real-time traffic through the offered/discovered TURN =
servers&#8221;, the real-time traffic may happen through the IP default =
gateways often congested by data traffic and QoS insensitive streaming =
video and file sharing. Without specific QoS methods at those points, =
the network raw bandwidth capacity may have to be 10-folded to achieve =
sufficient QoE when WebRTC usage becomes popular. Methods like DISCUSS =
addresses such things occurring when STUN instead TURN is used (by =
allowing flows into such congestion points), but only provides direct =
traffic info for outgoing (not incoming) real-time traffic and locally, =
therefore are not even applicable to reservation type of networks like =
Cable Networks and Mobile OTT. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
at would leave certain network types to investing in raw bandwidth =
increase, instead of simply borrowing the bandwidth (from much larger =
data traffic and QoS insensitive streaming video and file sharing at no =
adverse effect) and making that borrowed no-cost bandwidth very valuable =
for the carriers when delivering to their =
customers.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
know a few of you Cisco guys are among the ones that have been fighting =
against the general &#8220;it is all about bandwidth&#8221; and =
&#8220;it will go away with time&#8221;-attitude within IETF work, e.g. =
see <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html</a>, =
but the pointers given in the current QoS discussion within TRAM: =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
) will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time traffic.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i) will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. All networks will *<b>also be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time usage<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ne=
wer, fiber only type of networks, can still borrow bandwidth at no extra =
cost, *<b>by proprietary usage</b>* of RFC 5285 by browsers. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii) will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may has to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther explainations about QoS methods and their implementation were =
given a few days ago in: <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00274.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00274.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
), (ii) and (iii) will *<b>seriously delay</b>* usage of telepresence =
capable and quality demanding WebRTC (bandwidth being around 30 times =
higher over best effort Internet, than telephony over quality managed =
networks) and will *<b>vastly increase cost</b>* for ISPs to offer =
WebRTC with good QoE.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
low you are giving pointers saying &#8220;</span><span lang=3DEN-US>They =
are currently working on a problem-statement draft and a use-case draft, =
any input to those would be very helpful.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221; Those are unnecessary, risking leading to (i), (ii) and (iii) =
above.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
 are pointers such as: &#8220;</span><span lang=3DEN-US>RMCAT where =
congestion avoidance for RTP is being developed</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;. TRAMs Milestone 3 is for the purpose of directing real-time =
traffic where congestion control isn&#8217;t already in place. Using =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>RF=
C 5285 available since 2008, to fill traffic information in RTP packets =
(step D)) is probably a necessity for most future &#8220;congestion =
control&#8221; discussed.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues to: <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos">=
http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
cusing on DSCP-mapping without even mentioning RFC 5285 available since =
2008, to fill traffic information in RTP packets where it =A0is a =
necessity and will allow: <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) applications to directly convey QoS related real-time traffic info to =
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to be used by *<b>any network element implementing any suitable QoS =
methods for the particular network</b>* for <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of using the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only activity required, is to call for ISPs&#8217; review whether the =
traffic information transferred by RFC 5285 is sufficient for current =
and future needs in their network as suggested in <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have the RFC in =
place.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place, <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just that, =
without even having it MUST-established.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> tram =
[mailto:tram-bounces@ietf.org] <b>F=F6r </b>Pal Martinsen =
(palmarti)<br><b>Skickat:</b> den 21 februari 2014 10:37<br><b>Till:</b> =
tram@ietf.org<br><b>Kopia:</b> Karl Stahl; Oleg Moskalenko; Alan =
Johnston; Yoakum, John H (John)<br><b>=C4mne:</b> Re: [tram] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p></div></div=
><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>Hi,<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
agree the full QoS discussion should _not_ happen in TRAM. If you are =
interested in helping out in that area I suggest you looking into the =
AEON mailing list at:&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/aeon">https://www.ietf.org/=
mailman/listinfo/aeon</a> . They are currently working on a =
problem-statement draft and a use-case draft, any input to those would =
be very helpful. (<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01">http://=
tools.ietf.org/html/draft-eckel-aeon-use-cases-01</a>,&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00"=
>http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00</a>).<o=
:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>That said, STUN have a few nice characteristics that =
makes it a perfect candidate for transporting some of the QoS =
information. &nbsp;IMHO that would be extending the STUN spec and should =
be within the TRAM charter. &nbsp;The main goal =
of&nbsp;draft-martinsen-tram-discuss was to show how already existing =
QoS mechanisms could be transported with STUN to provide more value, and =
to start the discussion if TRAM is the appropriate place to have those =
on the wire format discussions.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>.-.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>P=E5l-Erik<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal>On =
20 Feb 2014, at 19:55 pm, Yoakum, John H (John) &lt;<a =
href=3D"mailto:yoakum@avaya.com">yoakum@avaya.com</a>&gt; =
wrote:<o:p></o:p></p></div><p =
class=3DMsoNormal><br><br><o:p></o:p></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I fully agree with the comments that QOS should be a low priority for =
the initial focus of the TRAM efforts.&nbsp; There are other groups =
doing QOS work and frankly I engage in WebRTC multimedia interactions =
daily over the Internet, enterprise VPNs, and various combinations and =
seldom suffer egregious quality issues.&nbsp; I am more concerned about =
carriers doing things to regulate or degrade WebRTC flows than a failure =
of existing Internet mechanisms to enable them.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Significant focus on QOS before we better enable TURN to be easily =
used in a browser environment taking advantage or normal web =
characteristics (as opposed to historic telephony constructs) would seem =
to be highly distracting at this point.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Ch=
eers,</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Jo=
hn</span></i><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:navy'>&nb=
sp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:red'>AVAY=
A</span><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><br></span><span lang=3DEN-US =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif";color:teal'>1.9=
19.425.8446</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span class=3Dapple-converted-space><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&nbsp;</span=
></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>tram [<a =
href=3D"mailto:tram-bounces@ietf.org">mailto:tram-bounces@ietf.org</a>]<s=
pan class=3Dapple-converted-space>&nbsp;</span><b>On Behalf Of<span =
class=3Dapple-converted-space>&nbsp;</span></b>Oleg =
Moskalenko<br><b>Sent:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Thursday, February 20, 2014 =
12:43 PM<br><b>To:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Alan =
Johnston<br><b>Cc:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Karl Stahl; <a =
href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><b>Subject:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Re: [tram] Fwd: I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On Thu, Feb 20, 2014 at 6:26 AM, =
Alan Johnston &lt;<a href=3D"mailto:alan.b.johnston@gmail.com" =
target=3D"_blank"><span =
style=3D'color:purple'>alan.b.johnston@gmail.com</span></a>&gt; =
wrote:<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>Personally, I am not sure how much =
QoS is actually in scope for TRAM. Have you been following RMCAT where =
congestion avoidance for RTP is being developed? &nbsp;I see some =
overlap in your goals and the goals of that =
work.<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span class=3Dhoenzb><span lang=3DEN-US =
style=3D'color:#888888'>-</span></span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#888888'>&nbsp;</span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div></div></div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span lang=3DEN-US>I'd =
concentrate on the TURN application-level functionality, for now, and =
I'd leave QoS for the future =
discussions.<br><br><br><o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif"'>__________=
_____________________________________<br>tram mailing list<br><a =
href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/tram">https://www.ietf.org/=
mailman/listinfo/tram</a><o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div></body></html>
------=_NextPart_000_04EF_01CF31B7.445B3D00--



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Date: Mon, 24 Feb 2014 22:52:04 +0000
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se>	<07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>	<07e401ceb713$bef87a60$3ce96f20$@stahl@intertex.se>	<524AB730.7040809@ericsson.com>	<00b001cebfc1$ba8e89e0$2fab9da0$@stahl@intertex.se>	<525272E8.5050300@ericsson.com>	<050801cec3f6$6172aec0$24580c40$@stahl@intertex.se>	<5253E5EB.8030608@alvestrand.no> <02bf01cecf34$35e174a0$a1a45de0$@stahl@intertex.se> <04dd01cf31ad$0fe62d00$2fb28700$@stahl@intertex.se>
To: Karl Stahl <karl.stahl@intertex.se>
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Subject: Re: [rtcweb] [tram] Payload Types assignments
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Karl,

I strongly disagree with this suggestion. An RTP header extension, =
located at an unknown and variable offset into a packet that does not =
have a well-defined magic number in the header, indicated using a =
dynamically assigned identifier that is conveyed in an out-of-band and =
encrypted signalling channel, is not an appropriate place to put QoS =
information that has to be processed on a per-packet basis. If you want =
DiffServ, you know where to find it.

Colin


On 24 Feb 2014, at 22:09, Karl Stahl <karl.stahl@intertex.se> wrote:
> I suggest to the RTCWEB WG that the below from the September and =
October discussions on the relevant [rtcweb] [avtext] [mmusic] lists =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html is =
introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC 5285, to =
allow:
> =20
> (1) WebRTC applications to directly convey QoS related real-time =
traffic info to the network at points where RTP flow is directed to by =
TRAM Milstone 3, to be used by *any network element implementing any =
suitable QoS methods for the particular network* for
> (2) *all* WebRTC browsers *and* clients, under *all* OSs, and *all* =
current and future IP network, to achieve best QoE
> (3) *without* having to force WebRTC into application specific =
networks (such as IMS) instead of using the Internet (including OTT).
> =20
> The only further activity required, is to call for ISPs=92 to review =
whether the traffic information transferred by RFC 5285 is sufficient =
for current and future needs in their network as suggested in below =
repeated =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
> <snip>
> =85two parameters (e.g. two bytes each) are encoded into the RTP =
header extension:
> =20
> A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic scale.
> =20
> B) The quality characteristics for the stream, with the highest bit =
set to 1, we could allocate a bit each for quality type e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.
> =20
> And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.
> </snip>
> =20
> Then this could be assigned numbers to have an RFC in place.
> =20
> With TRAM milestone 3 also place,
> market forces will drive ISPs and browser makers to implement just =
this, without even having it MUST-established.
> =20
> =93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and
> =93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?=94 and vice versa.
> =20
> Please see further emails soon following this one, for details and =
history.
> =20
> /Karl
> =20
> =20
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl Stahl
> Skickat: den 22 oktober 2013 16:37
> Till: 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus Westerlund'
> Kopia: 'Colin Perkins'
> =C4mne: [rtcweb] [avtext] Payload Types assignments was Re: SV: =
[mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
> =20
> Harald, I mostly agree with the quality requirements of different =
real-time traffic that the WebRTC browser/application may use. But =
rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. =20
> =20
> There is actually a good way to show these needs to the network =
(without using the PT, or diffserv bits, which aren=92t sufficient =
anyway).
> =20
> Let's use the RTP header extension field that also is visible outside =
the encrypted payload. A week ago came =
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that =
outlines the usage of the extension field for classification of traffic! =
This document does not yet outline what to put in there and how to =
encode it though.
> =20
> Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 =
discusses other webrtc usages of the RTP header extension in 5.2 (there =
can be many header extensions according to RFC 5285) and in 9 there is =
"WebRTC Use of RTP: Future Extensions".
> =20
> So, it looks obvious to use the RTP header extension to show the =
characteristics and bandwidth requirements to the network. It should not =
introduce any backward incompatibilities either.
> =20
> Such marking is done in every RTP packet so it can be set individually =
for each stream and could even be changed during a session (e.g. when =
limiting the bandwidth based on RTCP feedback). RFC 5286 also specifies =
how RTP extension header usage can be negotiated in SDP. I think this =
could be easily done by the WebRTC browser for "all current and future =
needs" if properly specified now.
> =20
> I suggest that two parameters (e.g. two bytes each) are encoded into =
the RTP header extension:
> =20
> A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic scale.
> =20
> B) The quality characteristics for the stream, with the highest bit =
set to 1, we could allocate a bit each quality e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.
> =20
> And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.
> =20
> Please note the totally different requirements a diffserv and an RSVP =
network have to know, so let=92s put all into these bytes. (E.g. a =
diffserv network don't need the bandwidth usage, but RSVP reservation =
networks (e.g. cable and 3G/4G OTT) do. There one should initially =
reserve the maximum bandwidth indicated, but can later re-reserve.)
> =20
> /Karl
> =20
> PS Microsoft seems to have done work in this field, defining a =
proprietary attribute =93MS Service Quality=94;
> However that seems to apply to the TURN server allocation request and =
would therefore:
> --- Apply to the whole UDP flow, and could not be set for each stream =
individually (with different requirements), and
> --- Does not handle the bandwidth requirement for incoming real-time =
traffic (required to reserve in RSVP type of networks)
> However the quality attributes conveyed and their encoding may  be =
considered.
> =20
> This is 2.2.2.19 MS-Service Quality Attribute from
> http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20=

> =20
> MS-Service Quality Attribute
> The MS-Service Quality attribute is used to convey information about =
the data stream that the protocol client is intending to transfer over =
an allocated port. The protocol client SHOULD<21> include this attribute =
as part of an Allocate request message. A TURN server SHOULD use the =
information in this attribute to make decisions about resource =
allocation, bandwidth prioritization, and data delivery methods. If the =
attribute is not present in the Allocate request message, the TURN =
server SHOULD assume that the data stream is audio with best effort =
delivery. The format of this attribute is as follows...
> ...
> The following stream types are supported in this extension. All other =
stream types are reserved for future use.
> =A7 "0x0001": Audio
> =A7 "0x0002": Video
> =A7 "0x0003": Supplemental Video
> =A7 "0x0004": Data
> Service Quality (2 bytes): The service quality level required by the =
protocol client for the stream.
> The following service quality levels are supported in this extension. =
All other service quality levels are reserved for future use.
> =A7 "0x0000": Best effort delivery.
> =A7 "0x0001": Reliable delivery.
> =20
> =20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Harald Alvestrand
> Skickat: den 8 oktober 2013 13:01
> Till: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] =
WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
> =20
> On 10/08/2013 09:17 AM, Karl Stahl wrote:
> > Hej Magnus,
> >=20
> >> Also, are you really interested in knowing that it is VP9 vs H.264,
> >> isn't
> > the questions this is video of this priority that is important?
> >> I think you need to more carefully consider what are the goals you
> >> try to
> > achieve them.
> >=20
> > Actually, my concern is to get an idea of the maximum bandwidth that
> > could be required for a WebRTC (ICE) setup media flow. Both voice =
and
> > video should be prioritized over data (their individual priority is =
of
> > less importance as long as there is sufficient bandwidth for both).
> =20
> You don't know that without knowing what the application is for.
> In, for instance, a shooter game with voice backchannels, the movement =
and event information (data) is MORE time sensitive than the voice data.
> =20
> >=20
> > With diffserv you don=92t need to know the bandwidth requirement, =
but
> > with RSVP reservation (like in cable and mobile networks) you need =
to
> > know how much to reserve. Voice is like 100's kbit/s, video VP8 or
> > H.264 is like 3,5 mbps.
> =20
> Again, without knowing the application, you don't know that.
> The application could decide to use QCIF or HD, and the bandwidth =
variation of screencast (semi-static with sudden, large changes) is =
completely different from that of a talking head, which is again =
completely different from a high-movement scene.
> =20
> >=20
> > To add to the complication of codec variants, the video codecs in
> > question for WebRTC have variable bandwidth, and when there is a =
poor
> > connection we see Chrome reducing the video window size to reduce =
the bandwidth used...
> >=20
> > I think the payload type field at best can reflect a maximum =
bandwidth
> > to initially reserve bandwidth for, and thereafter make new
> > reservations if the bandwidth changes during the call. So could we
> > change RTP to show maximum bandwidth instead of payload type in that
> > field outside the encrypted payload :) ... Or maybe that is not a =
joke?
> =20
> I think these ruminations only lead to one conclusion:
> =20
> You can't tell what the needed bandwidth is up front without asking =
the application.
> You can't tell what the right priority ranking is without asking the =
application.
> =20
> If you need to know the bandwidth or the priority up front, the =
application has to tell you. Anything else is pure heuristics.
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;">Karl,<div><br></div><div>I strongly disagree with =
this suggestion. An RTP header extension, located at an unknown and =
variable offset into a packet that does not have a well-defined magic =
number in the header, indicated using a dynamically assigned identifier =
that is conveyed in an out-of-band and encrypted signalling channel, is =
not an appropriate place to put QoS information that has to be processed =
on a per-packet basis. If you want DiffServ, you know where to find =
it.</div><div><br></div><div>Colin</div><div><br></div><div><br><div><div>=
On 24 Feb 2014, at 22:09, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:</div><blockquote type=3D"cite"><meta http-equiv=3D"Content-Type" =
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</o:shapelayout></xml><![endif]--><div lang=3D"SV" link=3D"blue" =
vlink=3D"purple"><div class=3D"WordSection1"><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">I suggest to the RTCWEB WG that the below from the =
September and October discussions on the relevant </span><span =
lang=3D"EN-US">[rtcweb] [avtext] [mmusic]</span><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue"> lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html"=
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
is introduced into </span><span lang=3D"EN-US">draft-ietf-rtcweb-rtp-usage=
 </span><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">for usage of RFC 5285, to =
allow:<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(1) WebRTC applications to directly convey QoS related =
real-time traffic info to the network at points where RTP flow is =
directed to by TRAM Milstone 3, to be used by *<b>any network element =
implementing any suitable QoS methods for the particular network</b>* =
for <o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(2) *<b>all</b>* WebRTC browsers *<b>and</b>* clients, =
under *<b>all</b>* OSs, and *<b>all</b>* current and future IP network, =
to achieve best QoE <o:p></o:p></span></p><p class=3D"MsoNormal"><b><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(3) *without* </span></b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">having to force WebRTC into application specific =
networks (such as IMS) instead of using the Internet (including =
OTT).<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">The only further activity required, is to call for =
ISPs=92 to review whether the traffic information transferred by RFC =
5285 is sufficient for current and future needs in their network as =
suggested in below repeated <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html"=
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a><o:p=
></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&lt;snip&gt;<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">=85two parameters (e.g. two bytes each) are encoded into the RTP =
header extension:<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic =
scale.<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">B) The quality characteristics for the stream, with the highest =
bit set to 1, we could allocate a bit each for quality type =
e.g:<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&lt;/snip&gt;<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Then this could be assigned numbers to have an RFC in =
place.<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">With TRAM milestone 3 also place, =
<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">market forces will drive ISPs and browser makers to =
implement just this, without even having it =
MUST-established.<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">=93Who does not want a =93WebRTC-Ready=94 Internet =
access?=94 and<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">=93Who wants to use Chrome, if Firefox, Internet =
Explorer or Safari comes with much better QoE?=94 and vice =
versa.<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Please see further emails soon following this one, for =
details and history.<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">/Karl<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><div><div =
style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm"><p class=3D"MsoNormal"><b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;">Fr=E5n:</span></b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;"> <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>=
] <b>F=F6r </b>Karl Stahl<br><b>Skickat:</b> den 22 oktober 2013 =
16:37<br><b>Till:</b> 'Harald Alvestrand'; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; 'Magnus =
Westerlund'<br><b>Kopia:</b> 'Colin Perkins'<br><b>=C4mne:</b> [rtcweb] =
[avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p></div=
></div><p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">Harald, I mostly agree with =
the quality requirements of different real-time traffic that the WebRTC =
browser/application may use. But rather than asking the application, =
let's convey the bandwidth and priority requirements to the network. =
Just like with the Payload type (that is hard to squeeze that =
information into) it must be visible to the network (and not changed by =
the network, like diffserv bits are). Such marking must also be =
available for incoming traffic, which is especially important in RSVP =
type of networks, that has to reserve bandwidth for it. =
&nbsp;<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">There is actually a good way to show these needs to the =
network (without using the PT, or diffserv bits, which aren=92t =
sufficient anyway). <o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">Let's use the RTP header extension field that also is =
visible outside the encrypted payload. A week ago came <a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">ht=
tp://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a> =
&nbsp;that outlines the usage of the extension field for classification =
of traffic! This document does not yet outline what to put in there and =
how to encode it though.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">Today's <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http://=
tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a> discusses other =
webrtc usages of the RTP header extension in 5.2 (there can be many =
header extensions according to RFC 5285) and in 9 there is "WebRTC Use =
of RTP: Future Extensions".<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">So, it looks obvious to use =
the RTP header extension to show the characteristics and bandwidth =
requirements to the network. It should not introduce any backward =
incompatibilities either.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">Such marking is done in =
every RTP packet so it can be set individually for each stream and could =
even be changed during a session (e.g. when limiting the bandwidth based =
on RTCP feedback). RFC 5286 also specifies how RTP extension header =
usage can be negotiated in SDP. I think this could be easily done by the =
WebRTC browser for "all current and future needs" if properly specified =
now.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">I suggest that two parameters (e.g. two bytes each) are =
encoded into the RTP header extension:<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">A) The maximum bandwidth =
requirement: Two bytes could contain everything from some bps for =
real-time text to Gbps for future 3D supersize telepresence=85 on a =
logarithmic scale.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">B) The quality characteristics for the stream, with the =
highest bit set to 1, we could allocate a bit each quality =
e.g:<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">Best Effort, Audio, Video, Supplemental Video, Gaming, =
Data, Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable =
Delivery, Prioritize X, Variation Y, that could be combined as required =
to describe the stream.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">And with highest bit set to =
0, there could instead be a number for special usage that does not fit =
the general description of the individual bits.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">Please note the totally =
different requirements a diffserv and an RSVP network have to know, so =
let=92s put all into these bytes. (E.g. a diffserv network don't need =
the bandwidth usage, but RSVP reservation networks (e.g. cable and 3G/4G =
OTT) do. There one should initially reserve the maximum bandwidth =
indicated, but can later re-reserve.)<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">/Karl<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">PS </span><span lang=3D"EN-US"=
 style=3D"font-family: Calibri, sans-serif;">Microsoft seems to have =
done work in this field, defining a proprietary attribute =93MS Service =
Quality=94; <o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">However that =
seems to apply to the TURN server allocation request and would =
therefore:<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">--- Apply to =
the whole UDP flow, and could not be set for each stream individually =
(with different requirements), and<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"font-family: =
Calibri, sans-serif;">--- Does not handle the bandwidth requirement for =
incoming real-time traffic (required to reserve in RSVP type of =
networks)<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">However the =
quality attributes conveyed and their encoding may &nbsp;be =
considered.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">&nbsp;</span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"">This is 2.2.2.19 MS-Service Quality Attribute from =
</span><o:p></o:p></p><p class=3D"MsoNormal"><span style=3D""><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).as=
px" =
title=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx">http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx<=
/a>&nbsp;<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
style=3D"">&nbsp;<o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">MS-Service =
Quality Attribute</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">The =
MS-Service Quality attribute is used to convey information about the =
data stream that the protocol client is intending to transfer over an =
allocated port. The protocol client SHOULD&lt;21&gt; include this =
attribute as part of an Allocate request message. A TURN server SHOULD =
use the information in this <span =
style=3D"background:yellow;mso-highlight:yellow">attribute to make =
decisions about resource allocation, bandwidth prioritization, and data =
delivery methods</span>. If the attribute is not present in the Allocate =
request message, the TURN server SHOULD assume that the data stream is =
audio with best effort delivery. The format of this attribute is as =
follows... </span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">...</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">The following =
stream types are supported in this extension. All other stream types are =
reserved for future use.</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
style=3D"font-family: Symbol;">=A7 </span></em><em><span lang=3D"EN-US" =
style=3D"font-family: Calibri, sans-serif;">"0x0001": =
Audio</span></em><span lang=3D"EN-US" style=3D""><o:p></o:p></span></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family: Symbol;">=A7 =
</span></em><em><span lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">"0x0002": Video</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
style=3D"font-family: Symbol;">=A7 </span></em><em><span lang=3D"EN-US" =
style=3D"font-family: Calibri, sans-serif;">"0x0003": Supplemental =
Video</span></em><span lang=3D"EN-US" style=3D""><o:p></o:p></span></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family: Symbol;">=A7 =
</span></em><em><span lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">"0x0004": Data</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">Service =
Quality (2 bytes): The service quality level required by the protocol =
client for the stream.</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" style=3D"font-family: Calibri, sans-serif;">The following =
service quality levels are supported in this extension. All other =
service quality levels are reserved for future use.</span></em><span =
lang=3D"EN-US" style=3D""><o:p></o:p></span></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family: Symbol;">=A7 =
</span></em><em><span lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">"0x0000": Best effort delivery.</span></em><span =
lang=3D"EN-US" style=3D""><o:p></o:p></span></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family: Symbol;">=A7 =
</span></em><em><span lang=3D"EN-US" style=3D"font-family: Calibri, =
sans-serif;">"0x0001": Reliable delivery.</span></em><span lang=3D"EN-US" =
style=3D""><o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" style=3D"">&nbsp;<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText">-----Ursprungligt =
meddelande-----<o:p></o:p></p><p class=3D"MsoPlainText">Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>=
] F=F6r Harald Alvestrand<o:p></o:p></p><p class=3D"MsoPlainText">Skickat:=
 den 8 oktober 2013 13:01<o:p></o:p></p><p class=3D"MsoPlainText">Till: =
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">=C4mne: Re: [rtcweb] Payload =
Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">On 10/08/2013 09:17 AM, Karl =
Stahl wrote:<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt; Hej Magnus,<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; Also, are you =
really interested in knowing that it is VP9 vs H.264, =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;&gt; isn't<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; the questions this is =
video of this priority that is important?<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; I think you need to =
more carefully consider what are the goals you <o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; try =
to<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt; achieve them.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; Actually, my concern is =
to get an idea of the maximum bandwidth that <o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; could be required for a =
WebRTC (ICE) setup media flow. Both voice and <o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; video should be =
prioritized over data (their individual priority is of =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
less importance as long as there is sufficient bandwidth for =
both).<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">You don't know that without knowing what the application =
is for.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">In, for instance, a shooter game with voice backchannels, =
the movement and event information (data) is MORE time sensitive than =
the voice data.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; With diffserv you don=92t=
 need to know the bandwidth requirement, but <o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; with RSVP reservation =
(like in cable and mobile networks) you need to <o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; know how much to =
reserve. Voice is like 100's kbit/s, video VP8 or =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
H.264 is like 3,5 mbps.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">Again, without knowing the =
application, you don't know that.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">The application could decide =
to use QCIF or HD, and the bandwidth variation of screencast =
(semi-static with sudden, large changes) is completely different from =
that of a talking head, which is again completely different from a =
high-movement scene.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; To add to the =
complication of codec variants, the video codecs in =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
question for WebRTC have variable bandwidth, and when there is a poor =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
connection we see Chrome reducing the video window size to reduce the =
bandwidth used...<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;<o:p>&nbsp;</o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; I think the payload =
type field at best can reflect a maximum bandwidth =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
to initially reserve bandwidth for, and thereafter make new =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
reservations if the bandwidth changes during the call. So could we =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
change RTP to show maximum bandwidth instead of payload type in that =
<o:p></o:p></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
field outside the encrypted payload :) ... Or maybe that is not a =
joke?<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">I think these ruminations only lead to one =
conclusion:<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">You can't tell what the needed bandwidth is up front =
without asking the application.<o:p></o:p></span></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">You can't tell what the =
right priority ranking is without asking the =
application.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">If you need to know the bandwidth or the priority up =
front, the application has to tell you. Anything else is pure =
heuristics.<o:p></o:p></span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">_______________________________________________<o:p></o:p><=
/span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">rtcweb mailing =
list<o:p></o:p></span></p><p class=3D"MsoPlainText"><a =
href=3D"mailto:rtcweb@ietf.org"><span =
lang=3D"EN-US">rtcweb@ietf.org</span></a><span =
lang=3D"EN-US"><o:p></o:p></span></p><p class=3D"MsoPlainText"><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span =
lang=3D"EN-US">https://www.ietf.org/mailman/listinfo/rtcweb</span></a><spa=
n =
lang=3D"EN-US"><o:p></o:p></span></p></div></div></blockquote></div><br><d=
iv apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div><div><br></d=
iv></span><br class=3D"Apple-interchange-newline">

</div>
<br></div></body></html>=

--Apple-Mail=_88B6A6AA-91D0-4A95-AD5B-F4C66DD48D6D--


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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Simon Perreault'" <simon.perreault@viagenie.ca>, "'Ted Hardie'" <ted.ietf@gmail.com>, "'Gonzalo Camarillo'" <gonzalo.camarillo@ericsson.com>, <rtcweb@ietf.org>, <tram@ietf.org>
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Subject: [rtcweb] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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To the TRAM WG and RTCWEB WG and ADs:

=20

It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed =
and
encouraged to route quality demanding WebRTC media into their IP pipes =
that
are capable of transporting real-time traffic without quality issues, =
using
TURN servers.=20

=20

The subject from the TRAM mailing initiation spells out:

*Milestone 3: TURN server auto-discovery mechanism for enterprise and =
ISPs*

The TRAM WG must assure that TURN servers offered by ISPs/NSPs can be =
used
for the above purpose.

(The objective of the TRAM WG milestone 3 is *not only* to resolve
restrictive enterprise NAT/firewall traversal problem using TURN =
servers.)

I request that this is explicitly clarified in the TRAM charter.

=20

The QoS discussions that has appeared in this TRAM WG, resulting from=20

draft-martinsen-tram-discuss-00.txt

and=20

draft-thomson-tram-turn-bandwidth-00.txt=20

has been confusing (to say the least), risking that the general QoS =
attitude
within IETF work =93it is all about bandwidth=94 and =93it will go away =
with
time=94:=20

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. That all networks *will be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

*while* newer, fiber only type of networks, still can borrow bandwidth =
at no
extra cost, *by proprietary usage* of RFC 5285 by browsers.=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may have to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

=20

Further explanations about QoS methods and their implementation were =
given a
few days ago in:=20

http://www.ietf.org/mail-archive/web/tram/current/msg00274.html=20

=20

(i), (ii) and (iii) risk *seriously delaying* usage of telepresence =
capable
and quality demanding WebRTC (bandwidth being around 30 times higher =
over
best effort Internet, than telephony over quality managed networks) and =
will
*vastly increase cost* for ISPs to offer WebRTC with good QoE.

=20

I am therefore advising that draft-martinsen-tram-discuss-00.txt and
draft-thomson-tram-turn-bandwidth-00.txt=20

are withdrawn from TRAM. draft-martinsen-tram-discuss-00.txt can be
reintroduced after clarification in the draft, that it is not about QoS =
as
clarified by
http://www.ietf.org/mail-archive/web/tram/current/msg00276.html.

=20

There are further details about this in the email
http://www.ietf.org/mail-archive/web/tram/current/msg00303.html also =
copied
below.

=20

It is therein explained that my repeated suggestion to RTCWEB WG
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html from =
the
October discussions on the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html to
introduce the QoS related usage of RFC 5285 into =
draft-ietf-rtcweb-rtp-usage
in combinations with the Milestone 3 objectives of TRAM, immediately =
would
allow:

(1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in
addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to
quickly allow us to finally enjoy the high quality and connectivity
(NAT/firewall traversal) of real-time communication services now =
possible,
for =20

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks=20

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of the Internet (including OTT).

=20

The only activity required, is to call for ISPs=92 review whether the =
traffic
information transferred by RFC 5285 is sufficient for the needs of their
networks.

=20

WebRTC *can be* used for existing application specific IP networks (such =
as
IMS) for services existing on these networks, but with the introduction =
of
the TRAM objectives above and the proposed traffic information in every =
RTP
packet http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html =
as
made possible by RFC 5285 available since 2008:=20

(a) the same as well as other real-time services using IETF RTP traffic,
*can also (often much better) be* implemented over the Internet =
(including
OTT)

(b) also considering needs of QoS, media routing, control, quality
measurements, SLA implementation, usage, accounting, billing and =
SP-peering,


that so far has been attributed application specific networks, but in
practice and in reality have shown difficulties to implement (e.g. =
SP=92s
peering of beyond POTS services, resulting in that the 50 year old, =
pre-AM
radio quality telephony service, still is the only truly global
communication service (and today often is used to initiate other better
proprietary communication services).

=20

Application specific networks such as the IMS network have their own =
methods
of handling QoS, that not in anyway will be made less efficient by the
methods proposed here. Application specific networks *can use and will =
also
benefit* from the flow control and traffic requirement marking of RTP
streams proposed here.

=20

WebRTC *will be* used over the Internet (including OTT) and *will =
probably
also be* used over the application specific IMS network for the services
existing there through a gateway function from the Internet.=20

=20

With RFC 5285 QoS usage and with TRAM milestone 3 in place (which I =
suggest
the WGs to expedite),=20

market forces will drive ISPs and browser makers to implement just that,
without even having it MUST-established.

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

The TRAM WG and RTCWEB WG chairs and ADs are advised to review whether =
IETF
standard work in these WGs by contributors having an interest in more =
than
one of current or emerging: ISPs, carrier equipment vendors or web =
browser
makers, may discriminate any with an interest only in one of these. The =
same
should apply to non-activity by WG contributors to remedy such
discrimination opened by allowed proprietary usage of RFCs such as RFC =
5285.

=20

One input for such review are responses directed to me in the September =
=96
October discussion on the RTCWEB list and on the TRAM list from my entry
there February 8
http://www.ietf.org/mail-archive/web/tram/current/msg00132.html.

=20

Please see just sent emails:

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html=20

http://www.ietf.org/mail-archive/web/tram/current/msg00303.html=20

and further emails soon following this one, for details and history.

=20

/Karl =20

=20

Charter charter-ietf-tram-01
<https://datatracker.ietf.org/doc/charter-ietf-tram/>
https://datatracker.ietf.org/doc/charter-ietf-tram/=20

=20

=20

Fr=E5n: Karl Stahl [mailto:karl.stahl@intertex.se]=20
Skickat: den 24 februari 2014 23:22
Till: 'Pal Martinsen (palmarti)'; 'tram@ietf.org'; 'rtcweb@ietf.org'
Kopia: 'Oleg Moskalenko'; 'Alan Johnston'; 'Yoakum, John H (John)'
=C4mne: SV: [tram] [rtcweb] I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

Hi P=E5l,

=20

You did not comment nor answer, in response to any of:

http://www.ietf.org/mail-archive/web/tram/current/msg00275.html=20

http://www.ietf.org/mail-archive/web/tram/current/msg00273.html=20

whether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the
application/browser to transfer relevant real-time traffic information
(types and bandwidth) into every RTP package by using the already IETF
standardized RTP extension header RCF 5285, which could be used by any
current or future Internet (including OTT) network where WebRTC =
real-time
traffic may flow. It seems to have been standardized already 2008 to =
allow
such usage.

=20

QoS related step C) draft-martinsen-tram-discuss can then be taken out =
of
TRAM, and step D) (that was never intended to be handled by TRAM anyway)
will be handled elsewhere.

=20

I have repeated my suggestion to RTCWEB WG from the October discussions =
on
the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html to
introduce the QoS related step D) into draft-ietf-rtcweb-rtp-usage
suggesting usage of RFC 5285, where traffic information in RTP packets
simply would be filled by any browser under any OS.

=20

Without step B), TRAM =93Enforcing the real-time traffic through the
offered/discovered TURN servers=94, the real-time traffic may happen =
through
the IP default gateways often congested by data traffic and QoS =
insensitive
streaming video and file sharing. Without specific QoS methods at those
points, the network raw bandwidth capacity may have to be 10-folded to
achieve sufficient QoE when WebRTC usage becomes popular. Methods like
DISCUSS addresses such things occurring when STUN instead TURN is used =
(by
allowing flows into such congestion points), but only provides direct
traffic info for outgoing (not incoming) real-time traffic and locally,
therefore are not even applicable to reservation type of networks like =
Cable
Networks and Mobile OTT.=20

=20

That would leave certain network types to investing in raw bandwidth
increase, instead of simply borrowing the bandwidth (from much larger =
data
traffic and QoS insensitive streaming video and file sharing at no =
adverse
effect) and making that borrowed no-cost bandwidth very valuable for the
carriers when delivering to their customers.

=20

I know a few of you Cisco guys are among the ones that have been =
fighting
against the general =93it is all about bandwidth=94 and =93it will go =
away with
time=94-attitude within IETF work, e.g. see
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html, but =
the
pointers given in the current QoS discussion within TRAM:=20

=20

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

=20

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. All networks will *also be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

Newer, fiber only type of networks, can still borrow bandwidth at no =
extra
cost, *by proprietary usage* of RFC 5285 by browsers.=20

=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may has to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

=20

Further explanations about QoS methods and their implementation were =
given a
few days ago in:=20

http://www.ietf.org/mail-archive/web/tram/current/msg00274.html=20

=20

(i), (ii) and (iii) will *seriously delay* usage of telepresence capable =
and
quality demanding WebRTC (bandwidth being around 30 times higher over =
best
effort Internet, than telephony over quality managed networks) and will
*vastly increase cost* for ISPs to offer WebRTC with good QoE.

=20

Below you are giving pointers saying =93They are currently working on a
problem-statement draft and a use-case draft, any input to those would =
be
very helpful.=94 Those are unnecessary, risking leading to (i), (ii) and =
(iii)
above.

=20

So are pointers such as: =93RMCAT where congestion avoidance for RTP is =
being
developed=94. TRAMs Milestone 3 is for the purpose of directing =
real-time
traffic where congestion control isn=92t already in place. Using RFC =
5285
available since 2008, to fill traffic information in RTP packets (step =
D))
is probably a necessity for most future =93congestion control=94 =
discussed.

=20

This chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues
to:=20

http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos=20

focusing on DSCP-mapping without even mentioning RFC 5285 available =
since
2008, to fill traffic information in RTP packets where it  is a =
necessity
and will allow:=20

(1) applications to directly convey QoS related real-time traffic info =
to
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to
be used by *any network element implementing any suitable QoS methods =
for
the particular network* for=20

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks=20

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of using the Internet (including OTT).

=20

The only activity required, is to call for ISPs=92 review whether the =
traffic
information transferred by RFC 5285 is sufficient for current and future
needs in their network as suggested in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have the RFC in place.

With TRAM milestone 3 also place,=20

market forces will drive ISPs and browser makers to implement just that,
without even having it MUST-established.

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

Fr=E5n: Pal Martinsen (palmarti) [mailto:palmarti@cisco.com]=20
Skickat: den 21 februari 2014 10:37
Till: tram@ietf.org
Kopia: Oleg Moskalenko; Alan Johnston; Karl Stahl; Yoakum, John H (John)
=C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt

=20

Hi,

=20

I agree the full QoS discussion should _not_ happen in TRAM. If you are
interested in helping out in that area I suggest you looking into the =
AEON
mailing list at: https://www.ietf.org/mailman/listinfo/aeon . They are
currently working on a problem-statement draft and a use-case draft, any
input to those would be very helpful.
(http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01,
http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00).

=20

That said, STUN have a few nice characteristics that makes it a perfect
candidate for transporting some of the QoS information.  IMHO that would =
be
extending the STUN spec and should be within the TRAM charter.  The main
goal of draft-martinsen-tram-discuss was to show how already existing =
QoS
mechanisms could be transported with STUN to provide more value, and to
start the discussion if TRAM is the appropriate place to have those on =
the
wire format discussions.

=20

.-.

P=E5l-Erik

=20

=20

On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <
<mailto:yoakum@avaya.com> yoakum@avaya.com> wrote:





+1

I fully agree with the comments that QOS should be a low priority for =
the
initial focus of the TRAM efforts.  There are other groups doing QOS =
work
and frankly I engage in WebRTC multimedia interactions daily over the
Internet, enterprise VPNs, and various combinations and seldom suffer
egregious quality issues.  I am more concerned about carriers doing =
things
to regulate or degrade WebRTC flows than a failure of existing Internet
mechanisms to enable them.

=20

Significant focus on QOS before we better enable TURN to be easily used =
in a
browser environment taking advantage or normal web characteristics (as
opposed to historic telephony constructs) would seem to be highly
distracting at this point.

=20

=20

Cheers,

John

=20

AVAYA
1.919.425.8446

=20

From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
Sent: Thursday, February 20, 2014 12:43 PM
To: Alan Johnston
Cc: Karl Stahl; tram@ietf.org
Subject: Re: [tram] Fwd: I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

=20

=20

On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <
<mailto:alan.b.johnston@gmail.com> alan.b.johnston@gmail.com> wrote:

=20

=20

=20

Personally, I am not sure how much QoS is actually in scope for TRAM. =
Have
you been following RMCAT where congestion avoidance for RTP is being
developed?  I see some overlap in your goals and the goals of that work.

-

=20

=20

I'd concentrate on the TURN application-level functionality, for now, =
and
I'd leave QoS for the future discussions.




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

=20


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>To=
 the TRAM WG and RTCWEB WG and ADs:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 must be a clear objective of the TRAM WG that ISPs/NSPs are allowed and =
encouraged to route quality demanding WebRTC media into their IP pipes =
that are capable of transporting real-time traffic without quality =
issues, using TURN servers. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e subject from the TRAM mailing initiation spells =
out:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>*<=
b>Milestone 3: TURN server auto-discovery mechanism for enterprise and =
ISPs</b>*<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e TRAM WG must assure that TURN servers offered by ISPs/NSPs can be used =
for the above purpose.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(T=
he objective of the TRAM WG milestone 3 is *<b>not only</b>* to resolve =
restrictive enterprise NAT/firewall traversal problem using TURN =
servers.)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
request that this is explicitly clarified in the TRAM =
charter.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e QoS discussions that has appeared in this TRAM WG, resulting from =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>draft-martinsen-tram-discuss-00.txt<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>draft-thomson-tram-turn-bandwidth-00.txt =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
s been confusing (to say the least), risking that the</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
general QoS attitude within IETF work &#8220;it is all about =
bandwidth&#8221; and &#8220;it will go away with time&#8221;: =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
) will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time traffic.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i) will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. That all networks *<b>will be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time usage<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>*<=
b>while</b>* newer, fiber only type of networks, still can borrow =
bandwidth at no extra cost, *<b>by proprietary usage</b>* of RFC 5285 by =
browsers. <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii) will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may have to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther explanations about QoS methods and their implementation were given =
a few days ago in: <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00274.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00274.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
), (ii) and (iii) risk *<b>seriously delaying</b>* usage of telepresence =
capable and quality demanding WebRTC (bandwidth being around 30 times =
higher over best effort Internet, than telephony over quality managed =
networks) and will *<b>vastly increase cost</b>* for ISPs to offer =
WebRTC with good QoE.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
am therefore advising that </span><span =
lang=3DEN-US>draft-martinsen-tram-discuss-00.txt </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d </span><span lang=3DEN-US>draft-thomson-tram-turn-bandwidth-00.txt =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ar=
e withdrawn from TRAM. </span><span =
lang=3DEN-US>draft-martinsen-tram-discuss-00.txt </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ca=
n be reintroduced after clarification in the draft, that it is not about =
QoS as clarified by <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00276.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00276.html</a>.<o:p>=
</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
ere are further details about this in the email <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00303.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00303.html</a> also =
copied below.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 is therein explained that my repeated suggestion to RTCWEB WG <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html</a> =
from the October discussions on the relevant </span><span =
lang=3DEN-US>[rtcweb] [avtext] [mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
to introduce the QoS related usage of RFC 5285 into </span><span =
lang=3DEN-US>draft-ietf-rtcweb-rtp-usage </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>in=
 combinations with the Milestone 3 objectives of TRAM, immediately would =
allow:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) *<b>all</b>* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE =
(in addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to quickly allow us to finally enjoy the high quality and =
connectivity (NAT/firewall traversal) of real-time communication =
services now possible, for =A0<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only activity required, is to call for ISPs&#8217; review whether the =
traffic information transferred by RFC 5285 is sufficient for the needs =
of their networks.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>We=
bRTC <b>*can be* </b>used for existing application specific IP networks =
(such as IMS) for services existing on these networks, but with the =
introduction of the TRAM objectives above and the proposed traffic =
information in every RTP packet <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
as made possible by RFC 5285 available since 2008: =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(a=
) the same as well as other real-time services using IETF RTP traffic, =
*<b>can also (often much better) be</b>* implemented over the Internet =
(including OTT)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(b=
) also considering needs of QoS, media routing, control, quality =
measurements, SLA implementation, usage, accounting, billing and =
SP-peering, =A0=A0<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>th=
at so far has been attributed application specific networks, but in =
practice and in reality have shown difficulties to implement (e.g. =
SP&#8217;s peering of beyond POTS services, resulting in that the 50 =
year old, pre-AM radio quality telephony service, still is the only =
truly global communication service (and today often is used to initiate =
other better proprietary communication =
services).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ap=
plication specific networks such as the IMS network have their own =
methods of handling QoS, that not in anyway will be made less efficient =
by the methods proposed here. Application specific networks <b>*can use =
and will also benefit*</b> from the flow control and traffic requirement =
marking of RTP streams proposed here.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>We=
bRTC *<b>will be</b>* used over the Internet (including OTT) and =
*<b>will probably also be</b>* used over the application specific IMS =
network for the services existing there through a gateway function from =
the Internet. <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th RFC 5285 QoS usage and with TRAM milestone 3 in place (which I =
suggest the WGs to expedite), <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just that, =
without even having it MUST-established.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e TRAM WG and RTCWEB WG chairs and ADs are advised to review whether =
IETF standard work in these WGs by contributors having an interest in =
more than one of current or emerging: ISPs, carrier equipment vendors or =
web browser makers, may discriminate any with an interest only in one of =
these. The same should apply to non-activity by WG contributors to =
remedy such discrimination opened by allowed proprietary usage of RFCs =
such as RFC 5285.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>On=
e input for such review are responses directed to me in the September =
&#8211; October discussion on the RTCWEB list and on the TRAM list from =
my entry there February 8 <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00132.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00132.html</a>.<o:p>=
</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see just sent emails:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11542.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00303.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00303.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d further emails soon following this one, for details and =
history.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl =A0<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:13.5pt;font-family:"Arial","sans-serif"'>Charter =
charter-ietf-tram-01 </span></b><a =
href=3D"https://datatracker.ietf.org/doc/charter-ietf-tram/"><span =
lang=3DEN-US>https://datatracker.ietf.org/doc/charter-ietf-tram/</span></=
a> <span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Karl Stahl =
[mailto:karl.stahl@intertex.se] <br><b>Skickat:</b> den 24 februari 2014 =
23:22<br><b>Till:</b> 'Pal Martinsen (palmarti)'; 'tram@ietf.org'; =
'rtcweb@ietf.org'<br><b>Kopia:</b> 'Oleg Moskalenko'; 'Alan Johnston'; =
'Yoakum, John H (John)'<br><b>=C4mne:</b> SV: [tram] [rtcweb] I-D =
Action: draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 P=E5l,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Yo=
u did not comment nor answer, in response to any =
of:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00273.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00273.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>wh=
ether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the =
application/browser to transfer relevant real-time traffic information =
(types and bandwidth) into every RTP package by using the already IETF =
standardized RTP extension header RCF 5285, which could be used by any =
current or future Internet (including OTT) network where WebRTC =
real-time traffic may flow. It seems to have been standardized already =
2008 to allow such usage.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Qo=
S related step C) </span><span lang=3DEN-US>draft-martinsen-tram-discuss =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ca=
n then be taken out of TRAM, and step D) (that was never intended to be =
handled by TRAM anyway) will be handled =
elsewhere.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have repeated my suggestion to RTCWEB WG from the October discussions on =
the relevant </span><span lang=3DEN-US>[rtcweb] [avtext] =
[mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
to introduce the QoS related step D) into </span><span =
lang=3DEN-US>draft-ietf-rtcweb-rtp-usage s</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ug=
gesting usage of RFC 5285, where traffic information in RTP packets =
simply would be filled by any browser under any OS.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
thout step B), TRAM &#8220;Enforcing the real-time traffic through the =
offered/discovered TURN servers&#8221;, the real-time traffic may happen =
through the IP default gateways often congested by data traffic and QoS =
insensitive streaming video and file sharing. Without specific QoS =
methods at those points, the network raw bandwidth capacity may have to =
be 10-folded to achieve sufficient QoE when WebRTC usage becomes =
popular. Methods like DISCUSS addresses such things occurring when STUN =
instead TURN is used (by allowing flows into such congestion points), =
but only provides direct traffic info for outgoing (not incoming) =
real-time traffic and locally, therefore are not even applicable to =
reservation type of networks like Cable Networks and Mobile OTT. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
at would leave certain network types to investing in raw bandwidth =
increase, instead of simply borrowing the bandwidth (from much larger =
data traffic and QoS insensitive streaming video and file sharing at no =
adverse effect) and making that borrowed no-cost bandwidth very valuable =
for the carriers when delivering to their =
customers.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
know a few of you Cisco guys are among the ones that have been fighting =
against the general &#8220;it is all about bandwidth&#8221; and =
&#8220;it will go away with time&#8221;-attitude within IETF work, e.g. =
see <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html</a>, =
but the pointers given in the current QoS discussion within TRAM: =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
) will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time traffic.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i) will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. All networks will *<b>also be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time usage<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ne=
wer, fiber only type of networks, can still borrow bandwidth at no extra =
cost, *<b>by proprietary usage</b>* of RFC 5285 by browsers. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii) will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may has to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther explanations about QoS methods and their implementation were given =
a few days ago in: <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00274.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00274.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
), (ii) and (iii) will *<b>seriously delay</b>* usage of telepresence =
capable and quality demanding WebRTC (bandwidth being around 30 times =
higher over best effort Internet, than telephony over quality managed =
networks) and will *<b>vastly increase cost</b>* for ISPs to offer =
WebRTC with good QoE.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
low you are giving pointers saying &#8220;</span><span lang=3DEN-US>They =
are currently working on a problem-statement draft and a use-case draft, =
any input to those would be very helpful.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221; Those are unnecessary, risking leading to (i), (ii) and (iii) =
above.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
 are pointers such as: &#8220;</span><span lang=3DEN-US>RMCAT where =
congestion avoidance for RTP is being developed</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;. TRAMs Milestone 3 is for the purpose of directing real-time =
traffic where congestion control isn&#8217;t already in place. Using RFC =
5285 available since 2008, to fill traffic information in RTP packets =
(step D)) is probably a necessity for most future &#8220;congestion =
control&#8221; discussed.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues to: <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos">=
http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
cusing on DSCP-mapping without even mentioning RFC 5285 available since =
2008, to fill traffic information in RTP packets where it &nbsp;is a =
necessity and will allow: <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) applications to directly convey QoS related real-time traffic info to =
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to be used by *<b>any network element implementing any suitable QoS =
methods for the particular network</b>* for <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of using the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only activity required, is to call for ISPs&#8217; review whether the =
traffic information transferred by RFC 5285 is sufficient for current =
and future needs in their network as suggested in <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have the RFC in =
place.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place, <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just that, =
without even having it MUST-established.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Pal =
Martinsen (palmarti) [mailto:palmarti@cisco.com] <br><b>Skickat:</b> den =
21 februari 2014 10:37<br><b>Till:</b> tram@ietf.org<br><b>Kopia:</b> =
Oleg Moskalenko; Alan Johnston; Karl Stahl; Yoakum, John H =
(John)<br><b>=C4mne:</b> Re: [tram] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p></div></div=
><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>Hi,<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
agree the full QoS discussion should _not_ happen in TRAM. If you are =
interested in helping out in that area I suggest you looking into the =
AEON mailing list at:&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/aeon">https://www.ietf.org/=
mailman/listinfo/aeon</a> . They are currently working on a =
problem-statement draft and a use-case draft, any input to those would =
be very helpful. (<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01">http://=
tools.ietf.org/html/draft-eckel-aeon-use-cases-01</a>,&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00"=
>http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00</a>).<o=
:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>That said, STUN have a few nice characteristics that =
makes it a perfect candidate for transporting some of the QoS =
information. &nbsp;IMHO that would be extending the STUN spec and should =
be within the TRAM charter. &nbsp;The main goal =
of&nbsp;draft-martinsen-tram-discuss was to show how already existing =
QoS mechanisms could be transported with STUN to provide more value, and =
to start the discussion if TRAM is the appropriate place to have those =
on the wire format discussions.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>.-.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>P=E5l-Erik<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On 20 Feb 2014, at 19:55 pm, =
Yoakum, John H (John) &lt;</span><a =
href=3D"mailto:yoakum@avaya.com"><span =
lang=3DEN-US>yoakum@avaya.com</span></a><span lang=3DEN-US>&gt; =
wrote:<o:p></o:p></span></p></div><p class=3DMsoNormal><span =
lang=3DEN-US><br><br><o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I fully agree with the comments that QOS should be a low priority for =
the initial focus of the TRAM efforts.&nbsp; There are other groups =
doing QOS work and frankly I engage in WebRTC multimedia interactions =
daily over the Internet, enterprise VPNs, and various combinations and =
seldom suffer egregious quality issues.&nbsp; I am more concerned about =
carriers doing things to regulate or degrade WebRTC flows than a failure =
of existing Internet mechanisms to enable them.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Significant focus on QOS before we better enable TURN to be easily =
used in a browser environment taking advantage or normal web =
characteristics (as opposed to historic telephony constructs) would seem =
to be highly distracting at this point.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Ch=
eers,</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Jo=
hn</span></i><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:navy'>&nb=
sp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:red'>AVAY=
A</span><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><br></span><span lang=3DEN-US =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif";color:teal'>1.9=
19.425.8446</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span class=3Dapple-converted-space><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&nbsp;</span=
></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>tram [<a =
href=3D"mailto:tram-bounces@ietf.org">mailto:tram-bounces@ietf.org</a>]<s=
pan class=3Dapple-converted-space>&nbsp;</span><b>On Behalf Of<span =
class=3Dapple-converted-space>&nbsp;</span></b>Oleg =
Moskalenko<br><b>Sent:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Thursday, February 20, 2014 =
12:43 PM<br><b>To:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Alan =
Johnston<br><b>Cc:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Karl Stahl; <a =
href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><b>Subject:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Re: [tram] Fwd: I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On Thu, Feb 20, 2014 at 6:26 AM, =
Alan Johnston &lt;<a href=3D"mailto:alan.b.johnston@gmail.com" =
target=3D"_blank"><span =
style=3D'color:purple'>alan.b.johnston@gmail.com</span></a>&gt; =
wrote:<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>Personally, I am not sure how much =
QoS is actually in scope for TRAM. Have you been following RMCAT where =
congestion avoidance for RTP is being developed? &nbsp;I see some =
overlap in your goals and the goals of that =
work.<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span class=3Dhoenzb><span lang=3DEN-US =
style=3D'color:#888888'>-</span></span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#888888'>&nbsp;</span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div></div></div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span lang=3DEN-US>I'd =
concentrate on the TURN application-level functionality, for now, and =
I'd leave QoS for the future =
discussions.<br><br><br><o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif"'>__________=
_____________________________________<br>tram mailing list<br><a =
href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/tram">https://www.ietf.org/=
mailman/listinfo/tram</a><o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div></body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Colin Perkins'" <csp@csperkins.org>
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Subject: Re: [rtcweb] [tram] Payload Types assignments
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Colin,

=20

If the below were the case, it would be =93DPI guesswork=94 that I also =
advice
against. RTP doesn=92t even have unique protocol header within UDP =96 =
it can
even be confused with other UDP payload.=20

=20

However, if the RTP is captured in a TURN-flow, as in TRAM Milestone 3, =
the
network point that this flow is directed to and can apply QoS methods
relevant to the network (which is not diffserve in Mobile OTT and Cable
Networks) has not a too difficult tasks. Linking each RTP-flow by its ID =
and
sequence number, and picking exactly the right traffic type and =
bandwidth
parameters is doable (see inline below, we at Ingate do it already)!

=20

Also DiffServ DSCP-bits are seldom maintained crossing network =
boundaries,
thus carrying no relevant information at the receiving end, while the =
RTP
extension header remains unchanged end-to-end. In DSCP-bits, there is no
bandwidth requirement information when entering networks requiring
reservation. That is always (and dynamically set) available in the RTP
extension header for each packet.

=20

And, I am sure you are aware of the difficulties of getting DSCP-bits
through OS sockets, which is even worse with multiple streams over the =
same
UDP-port

=20

Further, defining the QoS RTP extension header as in RFC5285, does not =
in
anyway conflict with other RTP extension headers or DSCP transfer or
settings.

=20

/Karl

=20

Fr=E5n: Colin Perkins [mailto:csp@csperkins.org]=20
Skickat: den 24 februari 2014 23:52
Till: Karl Stahl
Kopia: rtcweb@ietf.org; Magnus Westerlund; tram@ietf.org; Harald =
Alvestrand
=C4mne: Re: [rtcweb] [tram] Payload Types assignments

=20

Karl,

=20

I strongly disagree with this suggestion. An RTP header extension, =
located
at an unknown and variable offset into a packet that does not have a
well-defined magic number in the header,=20

This is how to find the traffic info:

   The payload of the classifier header extension element can be encoded

   using either the one-byte or two-byte header defined in [rfc5285
<http://tools.ietf.org/html/rfc5285> ] and

   shown below in Figure 1 and 2 below.

=20

      0                   1                   2                   3

      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

     |  ID   | len=3D1 |   Namespace   |    Value      |    0 (pad)    |

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

=20

             Figure 1: Classifier Using the One-Byte Header

=20

      0                   1                   2                   3

      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

     |      ID       |    len=3D2      |   Namespace   |    Value      |

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

=20

=20

             Figure 2: Classifier Using the Two-Byte Header

=20

indicated using a dynamically assigned identifier that is conveyed in an
out-of-band=20

--- It is the TURN flow!!! Requested by the ICE protocol for the =
specific
media to come!

and encrypted signalling channel,=20

--- Only the payload is encrypted in SRTP =96 leaving id, seq no and =
header
ext to be used as intended!

--- Thus being the appropriate place=85

is not an appropriate place to put QoS information that has to be =
processed
on a per-packet basis.=20

If you want DiffServ, you know where to find it.

--- True, but if it cannot be used=85

Colin

=20

=20

On 24 Feb 2014, at 22:09, Karl Stahl <karl.stahl@intertex.se> wrote:

I suggest to the RTCWEB WG that the below from the September and October
discussions on the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html is
introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC 5285, to =
allow:

=20

(1) WebRTC applications to directly convey QoS related real-time traffic
info to the network at points where RTP flow is directed to by TRAM =
Milstone
3, to be used by *any network element implementing any suitable QoS =
methods
for the particular network* for=20

(2) *all* WebRTC browsers *and* clients, under *all* OSs, and *all* =
current
and future IP network, to achieve best QoE=20

(3) *without* having to force WebRTC into application specific networks
(such as IMS) instead of using the Internet (including OTT).

=20

The only further activity required, is to call for ISPs=92 to review =
whether
the traffic information transferred by RFC 5285 is sufficient for =
current
and future needs in their network as suggested in below repeated
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have an RFC in place.

=20

With TRAM milestone 3 also place,=20

market forces will drive ISPs and browser makers to implement just this,
without even having it MUST-established.

=20

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

=20

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
Stahl
Skickat: den 22 oktober 2013 16:37
Till: 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus Westerlund'
Kopia: 'Colin Perkins'
=C4mne: [rtcweb] [avtext] Payload Types assignments was Re: SV: [mmusic] =
WGLC
of draft-ietf-rtcweb-use-cases-and-requirements-11

=20

Harald, I mostly agree with the quality requirements of different =
real-time
traffic that the WebRTC browser/application may use. But rather than =
asking
the application, let's convey the bandwidth and priority requirements to =
the
network. Just like with the Payload type (that is hard to squeeze that
information into) it must be visible to the network (and not changed by =
the
network, like diffserv bits are). Such marking must also be available =
for
incoming traffic, which is especially important in RSVP type of =
networks,
that has to reserve bandwidth for it. =20

=20

There is actually a good way to show these needs to the network (without
using the PT, or diffserv bits, which aren=92t sufficient anyway).=20

=20

Let's use the RTP header extension field that also is visible outside =
the
encrypted payload. A week ago came
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that
outlines the usage of the extension field for classification of traffic!
This document does not yet outline what to put in there and how to =
encode it
though.

=20

Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 =
discusses
other webrtc usages of the RTP header extension in 5.2 (there can be =
many
header extensions according to RFC 5285) and in 9 there is "WebRTC Use =
of
RTP: Future Extensions".

=20

So, it looks obvious to use the RTP header extension to show the
characteristics and bandwidth requirements to the network. It should not
introduce any backward incompatibilities either.

=20

Such marking is done in every RTP packet so it can be set individually =
for
each stream and could even be changed during a session (e.g. when =
limiting
the bandwidth based on RTCP feedback). RFC 5286 also specifies how RTP
extension header usage can be negotiated in SDP. I think this could be
easily done by the WebRTC browser for "all current and future needs" if
properly specified now.

=20

I suggest that two parameters (e.g. two bytes each) are encoded into the =
RTP
header extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each quality e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

=20

Please note the totally different requirements a diffserv and an RSVP
network have to know, so let=92s put all into these bytes. (E.g. a =
diffserv
network don't need the bandwidth usage, but RSVP reservation networks =
(e.g.
cable and 3G/4G OTT) do. There one should initially reserve the maximum
bandwidth indicated, but can later re-reserve.)

=20

/Karl

=20

PS Microsoft seems to have done work in this field, defining a =
proprietary
attribute =93MS Service Quality=94;=20

However that seems to apply to the TURN server allocation request and =
would
therefore:

--- Apply to the whole UDP flow, and could not be set for each stream
individually (with different requirements), and

--- Does not handle the bandwidth requirement for incoming real-time =
traffic
(required to reserve in RSVP type of networks)

However the quality attributes conveyed and their encoding may  be
considered.

=20

This is 2.2.2.19 MS-Service Quality Attribute from=20

http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20

=20

MS-Service Quality Attribute

The MS-Service Quality attribute is used to convey information about the
data stream that the protocol client is intending to transfer over an
allocated port. The protocol client SHOULD<21> include this attribute as
part of an Allocate request message. A TURN server SHOULD use the
information in this attribute to make decisions about resource =
allocation,
bandwidth prioritization, and data delivery methods. If the attribute is =
not
present in the Allocate request message, the TURN server SHOULD assume =
that
the data stream is audio with best effort delivery. The format of this
attribute is as follows...=20

...

The following stream types are supported in this extension. All other =
stream
types are reserved for future use.

=A7 "0x0001": Audio

=A7 "0x0002": Video

=A7 "0x0003": Supplemental Video

=A7 "0x0004": Data

Service Quality (2 bytes): The service quality level required by the
protocol client for the stream.

The following service quality levels are supported in this extension. =
All
other service quality levels are reserved for future use.

=A7 "0x0000": Best effort delivery.

=A7 "0x0001": Reliable delivery.

=20

=20

-----Ursprungligt meddelande-----

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Harald
Alvestrand

Skickat: den 8 oktober 2013 13:01

Till: rtcweb@ietf.org

=C4mne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] WGLC =
of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20

On 10/08/2013 09:17 AM, Karl Stahl wrote:

> Hej Magnus,

>=20

>> Also, are you really interested in knowing that it is VP9 vs H.264,=20

>> isn't

> the questions this is video of this priority that is important?

>> I think you need to more carefully consider what are the goals you=20

>> try to

> achieve them.

>=20

> Actually, my concern is to get an idea of the maximum bandwidth that=20

> could be required for a WebRTC (ICE) setup media flow. Both voice and=20

> video should be prioritized over data (their individual priority is of =


> less importance as long as there is sufficient bandwidth for both).

=20

You don't know that without knowing what the application is for.

In, for instance, a shooter game with voice backchannels, the movement =
and
event information (data) is MORE time sensitive than the voice data.

=20

>=20

> With diffserv you don=92t need to know the bandwidth requirement, but=20

> with RSVP reservation (like in cable and mobile networks) you need to=20

> know how much to reserve. Voice is like 100's kbit/s, video VP8 or=20

> H.264 is like 3,5 mbps.

=20

Again, without knowing the application, you don't know that.

The application could decide to use QCIF or HD, and the bandwidth =
variation
of screencast (semi-static with sudden, large changes) is completely
different from that of a talking head, which is again completely =
different
from a high-movement scene.

=20

>=20

> To add to the complication of codec variants, the video codecs in=20

> question for WebRTC have variable bandwidth, and when there is a poor=20

> connection we see Chrome reducing the video window size to reduce the
bandwidth used...

>=20

> I think the payload type field at best can reflect a maximum bandwidth =


> to initially reserve bandwidth for, and thereafter make new=20

> reservations if the bandwidth changes during the call. So could we=20

> change RTP to show maximum bandwidth instead of payload type in that=20

> field outside the encrypted payload :) ... Or maybe that is not a =
joke?

=20

I think these ruminations only lead to one conclusion:

=20

You can't tell what the needed bandwidth is up front without asking the
application.

You can't tell what the right priority ranking is without asking the
application.

=20

If you need to know the bandwidth or the priority up front, the =
application
has to tell you. Anything else is pure heuristics.

=20

_______________________________________________

rtcweb mailing list

 <mailto:rtcweb@ietf.org> rtcweb@ietf.org

 <https://www.ietf.org/mailman/listinfo/rtcweb>
https://www.ietf.org/mailman/listinfo/rtcweb

=20

=20

--=20

Colin Perkins

http://csperkins.org/

=20

=20

=20


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Co=
lin,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>If=
 the below were the case, it would be &#8220;DPI guesswork&#8221; that I =
also advice against. RTP doesn&#8217;t even have unique protocol header =
within UDP &#8211; it can even be confused with other UDP payload. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
wever, if the RTP is captured in a TURN-flow, as in TRAM Milestone 3, =
the network point that this flow is directed to and can apply QoS =
methods relevant to the network (which is not diffserve in Mobile OTT =
and Cable Networks) has not a too difficult tasks. Linking each RTP-flow =
by its ID and sequence number, and picking exactly the right traffic =
type and bandwidth parameters is doable (see inline below, we at Ingate =
do it already)!<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Al=
so DiffServ DSCP-bits are seldom maintained crossing network boundaries, =
thus carrying no relevant information at the receiving end, while the =
RTP extension header remains unchanged end-to-end. In DSCP-bits, there =
is no bandwidth requirement information when entering networks requiring =
reservation. That is always (and dynamically set) available in the RTP =
extension header for each packet.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>An=
d, I am sure you are aware of the difficulties of getting DSCP-bits =
through OS sockets, which is even worse with multiple streams over the =
same UDP-port<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther, defining the QoS RTP extension header as in RFC5285, does not in =
anyway conflict with other RTP extension headers or DSCP transfer or =
settings.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Colin =
Perkins [mailto:csp@csperkins.org] <br><b>Skickat:</b> den 24 februari =
2014 23:52<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> rtcweb@ietf.org; =
Magnus Westerlund; tram@ietf.org; Harald Alvestrand<br><b>=C4mne:</b> =
Re: [rtcweb] [tram] Payload Types =
assignments<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Karl,<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
strongly disagree with this suggestion. An RTP header extension, located =
at an unknown and variable offset into a packet that does not have a =
well-defined magic number in the header, <span =
style=3D'color:blue'><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is is how to find the traffic info:</span><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0 The payload =
of the classifier header extension element can be =
encoded<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0 using either =
the one-byte or two-byte header defined in [<a =
href=3D"http://tools.ietf.org/html/rfc5285" title=3D"&quot;A General =
Mechanism for RTP Header Extensions&quot;">rfc5285</a>] =
and<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0 shown below =
in Figure 1 and 2 below.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0=A0 =
0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
1=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
2=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
3<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0=A0 0 1 =
2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 =
1<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 |=A0 =
ID=A0=A0 | len=3D1 |=A0=A0 Namespace=A0=A0 |=A0=A0=A0 =
Value=A0=A0=A0=A0=A0 |=A0=A0=A0 0 (pad)=A0=A0=A0 =
|<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Figure 1: Classifier Using =
the One-Byte Header<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0=A0 =
0=A0=A0=A0=A0 =
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A01=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0 =
2=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
3<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0=A0 0 1 =
2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 =
1<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 =
|=A0=A0=A0=A0=A0 ID=A0=A0=A0=A0=A0=A0 |=A0=A0=A0 len=3D2=A0=A0=A0=A0=A0 =
|=A0=A0 Namespace=A0=A0 |=A0=A0=A0 Value=A0=A0=A0=A0=A0 =
|<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier New"'>=A0=A0=A0=A0 =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New"'>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Figure 2: Classifier Using =
the Two-Byte Header<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>indicated using a dynamically assigned identifier that is =
conveyed in an out-of-band <span =
style=3D'color:blue'><o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- It is the TURN flow!!! Requested by the ICE protocol for the specific =
media to come!<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>and encrypted signalling channel, <span =
style=3D'color:blue'><o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Only the payload is encrypted in SRTP &#8211; leaving id, seq no and =
header ext to be used as intended!<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Thus being the appropriate place&#8230;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>is not an appropriate place to put =
QoS information that has to be processed on a per-packet basis. <span =
style=3D'color:blue'><o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>If you want DiffServ, you know =
where to find it.<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>--- True, but =
if it cannot be used&#8230;</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal>Colin<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal>On =
24 Feb 2014, at 22:09, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
suggest to the RTCWEB WG that the below from the September and October =
discussions on the relevant </span><span lang=3DEN-US>[rtcweb] [avtext] =
[mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
is introduced into </span><span lang=3DEN-US>draft-ietf-rtcweb-rtp-usage =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
r usage of RFC 5285, to allow:</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) WebRTC applications to directly convey QoS related real-time traffic =
info to the network at points where RTP flow is directed to by TRAM =
Milstone 3, to be used by *<b>any network element implementing any =
suitable QoS methods for the particular network</b>* for =
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* clients, under *<b>all</b>* =
OSs, and *<b>all</b>* current and future IP network, to achieve best QoE =
</span><o:p></o:p></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to force WebRTC into application specific networks (such as IMS) =
instead of using the Internet (including OTT).</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only further activity required, is to call for ISPs&#8217; to review =
whether the traffic information transferred by RFC 5285 is sufficient =
for current and future needs in their network as suggested in below =
repeated <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a></=
span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
/span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have an RFC in =
place.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place, </span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just this, =
without even having it MUST-established.</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] <b>F=F6r </b>Karl Stahl<br><b>Skickat:</b> den 22 oktober 2013 =
16:37<br><b>Till:</b> 'Harald Alvestrand'; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; 'Magnus =
Westerlund'<br><b>Kopia:</b> 'Colin Perkins'<br><b>=C4mne:</b> [rtcweb] =
[avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p></di=
v></div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Harald, I mostly agree with the quality requirements of =
different real-time traffic that the WebRTC browser/application may use. =
But rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. &nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>There is actually a good way to =
show these needs to the network (without using the PT, or diffserv bits, =
which aren&#8217;t sufficient anyway). </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Let's use the RTP header =
extension field that also is visible outside the encrypted payload. A =
week ago came <a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">h=
ttp://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a> =
&nbsp;that outlines the usage of the extension field for classification =
of traffic! This document does not yet outline what to put in there and =
how to encode it though.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Today's <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http:/=
/tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a> discusses other =
webrtc usages of the RTP header extension in 5.2 (there can be many =
header extensions according to RFC 5285) and in 9 there is &quot;WebRTC =
Use of RTP: Future Extensions&quot;.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>So, it looks obvious to use the =
RTP header extension to show the characteristics and bandwidth =
requirements to the network. It should not introduce any backward =
incompatibilities either.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Such marking is done in every =
RTP packet so it can be set individually for each stream and could even =
be changed during a session (e.g. when limiting the bandwidth based on =
RTCP feedback). RFC 5286 also specifies how RTP extension header usage =
can be negotiated in SDP. I think this could be easily done by the =
WebRTC browser for &quot;all current and future needs&quot; if properly =
specified now.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>I suggest that two parameters (e.g. two bytes each) are =
encoded into the RTP header extension:</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>A) The maximum bandwidth =
requirement: Two bytes could contain everything from some bps for =
real-time text to Gbps for future 3D supersize telepresence&#8230; on a =
logarithmic scale.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>B) The quality characteristics for the stream, with the =
highest bit set to 1, we could allocate a bit each quality =
e.g:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Best Effort, Audio, Video, Supplemental Video, Gaming, =
Data, Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable =
Delivery, Prioritize X, Variation Y, that could be combined as required =
to describe the stream.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>And with highest bit set to 0, =
there could instead be a number for special usage that does not fit the =
general description of the individual bits.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Please note the totally =
different requirements a diffserv and an RSVP network have to know, so =
let&#8217;s put all into these bytes. (E.g. a diffserv network don't =
need the bandwidth usage, but RSVP reservation networks (e.g. cable and =
3G/4G OTT) do. There one should initially reserve the maximum bandwidth =
indicated, but can later re-reserve.)</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>/Karl</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>PS </span><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>Microsoft seems to have =
done work in this field, defining a proprietary attribute &#8220;MS =
Service Quality&#8221;; </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>However that seems to apply =
to the TURN server allocation request and would =
therefore:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>--- Apply to =
the whole UDP flow, and could not be set for each stream individually =
(with different requirements), and</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>--- Does not handle the =
bandwidth requirement for incoming real-time traffic (required to =
reserve in RSVP type of networks)</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>However the quality =
attributes conveyed and their encoding may &nbsp;be =
considered.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&nbsp;</span><o:p></o:p></p>=
<p class=3DMsoNormal><span lang=3DEN-US>This is 2.2.2.19 MS-Service =
Quality Attribute from </span><o:p></o:p></p><p class=3DMsoNormal><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx" =
title=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).=
aspx">http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).asp=
x</a>&nbsp;<o:p></o:p></p><p class=3DMsoNormal>&nbsp;<o:p></o:p></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>MS-Service Quality =
Attribute</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>The MS-Service =
Quality attribute is used to convey information about the data stream =
that the protocol client is intending to transfer over an allocated =
port. The protocol client SHOULD&lt;21&gt; include this attribute as =
part of an Allocate request message. A TURN server SHOULD use the =
information in this <span =
style=3D'background:yellow;mso-highlight:yellow'>attribute to make =
decisions about resource allocation, bandwidth prioritization, and data =
delivery methods</span>. If the attribute is not present in the Allocate =
request message, the TURN server SHOULD assume that the data stream is =
audio with best effort delivery. The format of this attribute is as =
follows... </span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>...</span></em><o:p></o:p></=
p><p class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>The following stream types =
are supported in this extension. All other stream types are reserved for =
future use.</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0001&quot;: =
Audio</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0002&quot;: =
Video</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0003&quot;: =
Supplemental Video</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0004&quot;: =
Data</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>Service =
Quality (2 bytes): The service quality level required by the protocol =
client for the stream.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>The following service =
quality levels are supported in this extension. All other service =
quality levels are reserved for future use.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0000&quot;: Best =
effort delivery.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0001&quot;: =
Reliable delivery.</span></em><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<o:p></o:p></p><p =
class=3DMsoPlainText>Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] F=F6r Harald Alvestrand<o:p></o:p></p><p =
class=3DMsoPlainText>Skickat: den 8 oktober 2013 13:01<o:p></o:p></p><p =
class=3DMsoPlainText>Till: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>=C4mne: Re: [rtcweb] Payload =
Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>On 10/08/2013 09:17 AM, Karl =
Stahl wrote:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; Hej Magnus,</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; Also, are you really =
interested in knowing that it is VP9 vs H.264, </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; =
isn't</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; the questions this is video of this priority that is =
important?</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; I think you need to more carefully consider what =
are the goals you </span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; try to</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; achieve =
them.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; Actually, my concern is to =
get an idea of the maximum bandwidth that </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; could be required for a =
WebRTC (ICE) setup media flow. Both voice and </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; video should be prioritized =
over data (their individual priority is of </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; less importance as long as =
there is sufficient bandwidth for both).</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You don't know that without =
knowing what the application is for.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>In, for instance, a shooter game =
with voice backchannels, the movement and event information (data) is =
MORE time sensitive than the voice data.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; With diffserv you =
don&#8217;t need to know the bandwidth requirement, but =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
with RSVP reservation (like in cable and mobile networks) you need to =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
know how much to reserve. Voice is like 100's kbit/s, video VP8 or =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
H.264 is like 3,5 mbps.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Again, without knowing the =
application, you don't know that.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>The application could decide to =
use QCIF or HD, and the bandwidth variation of screencast (semi-static =
with sudden, large changes) is completely different from that of a =
talking head, which is again completely different from a high-movement =
scene.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; To add to the complication =
of codec variants, the video codecs in </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; question for WebRTC have =
variable bandwidth, and when there is a poor </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; connection we see Chrome =
reducing the video window size to reduce the bandwidth =
used...</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; I think the payload type =
field at best can reflect a maximum bandwidth </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; to initially reserve =
bandwidth for, and thereafter make new </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; reservations if the =
bandwidth changes during the call. So could we </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; change RTP to show maximum =
bandwidth instead of payload type in that </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; field outside the encrypted =
payload :) ... Or maybe that is not a joke?</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>I think these ruminations only =
lead to one conclusion:</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You can't tell what the needed =
bandwidth is up front without asking the =
application.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>You can't tell what the right priority ranking is without =
asking the application.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>If you need to know the =
bandwidth or the priority up front, the application has to tell you. =
Anything else is pure heuristics.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>_______________________________________________</span><o:p><=
/o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>rtcweb mailing =
list</span><o:p></o:p></p><p class=3DMsoPlainText><a =
href=3D"mailto:rtcweb@ietf.org"><span =
lang=3DEN-US>rtcweb@ietf.org</span></a><o:p></o:p></p><p =
class=3DMsoPlainText><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span =
lang=3DEN-US>https://www.ietf.org/mailman/listinfo/rtcweb</span></a><o:p>=
</o:p></p></div></blockquote></div><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p><div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>--&nbsp;<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>Colin Perkins<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a><o:p></o:p></span=
></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div></div></body></html>
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From nobody Mon Feb 24 23:09:46 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Some editorial comments on section 5
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On Feb 11, 2014, at 7:12 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> Some editorial comments:
>=20
> Q1:
>=20
> Section 5 says:
>=20
> 	"SCTP as specified in [RFC4960] MUST be used in
>   	combination with the extension defined in [RFC3758] and provides =
the
>   	following interesting features for transporting non-media data
>   	between browsers:"
>=20
> I suggest to say:
>=20
> 	"The SCTP extension defined in [RFC3758] MUST be supported, and
>     	provides the following features:
But this implies that the listed features are provided by the extension
defined in RFC 3758, which is not true. The features are provided by
the combination of RFC 4960 and RFC 3758.
>=20
>=20
> Q2:
>=20
> Section 5 says:
>=20
> 	"Each SCTP user message contains a so called Payload Protocol
>   	Identifier (PPID) that is passed..."
>=20
> I suggest to remove "so called".
Removed.
>=20
>=20
> Q3:
>=20
> Section 5 says:
>=20
> 	"This value can be used to multiplex multiple protocols
>   	over a single SCTP association.  The sender provides for each
>   	protocol a specific PPID and the receiver can demultiplex the
>   	messages based on the received PPID."
>=20
> I think we discussed this earlier. The PPID value CAN NOT (at least =
not as used by WebRTC) be used to multiplex multiple protocols over a =
single SCTP association. If you create multiple channels, for multiple =
protocols, the same PPID values may be used for each protocol.
>=20
> The PPID CAN, however, be used within a data channel, e.g. to =
demultiplex the DCP from channel data.
I guess the problem is that I consider 'protocol' as something running =
on top of SCTP
like DCEP, Binary of String data, you consider 'protocol' as something =
running
on top of Binary of String data and the 'protocol' being identified in =
the JS protocol string.

What about using:

<t>Each SCTP user message contains a Payload Protocol Identifier (PPID)
that is passed to SCTP by its upper layer on the sending side and
provided to its upper layer on the receiving side.
The PPID can be used to multiplex/demultiplex multiple upper layers over
a single SCTP association.
In the WebRTP context, the PPID is used to distinguish between
UTF-8 encoded user data,
binary encoded userdata and
the Data Channel Establishment Protocol defined in
<xref target=3D'I-D.ietf-rtcweb-data-protocol'/>.
Please note that the PPID is not accessable via the Javascript API.</t>

Does this address your issue?
>=20
>=20
> Q4:
>=20
> Section 5 says:
>=20
> 	"When using DTLS as the lower layer, only single homed SCTP
>   	associations MUST be used, since DTLS does not expose any =
address
>   	management to its upper layer."
>=20
> I suggest to change "MUST be used" to "are supported".
OK.
>=20
>=20
> Q5:
>=20
> Sometimes you say "SCTP" and sometimes "SCTP as defined in [RFC4960]".
>=20
> I suggest to e.g. use the reference on the first occurrence, and then =
only use "SCTP".
I double checked:
SCTP as defined in [RFC4960] is used in
* In the Introduction (first occurrence) talking about SCTP / UDP
* In the Introduction to make sure which specifications are used: RFC =
4960 and RFC 3758
* In section 5 when used with RFC 2119 language.
* In section 6 to make clear what relates to RFC 4960 and what to the =
NDATA extension.

All of these are intentional. Which one do you think is not necessary?
>=20
>=20
> Regards,
>=20
> Christer
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Tue Feb 25 00:00:30 2014
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To: "'Alan Johnston'" <alan.b.johnston@gmail.com>, <rtcweb@ietf.org>, <tram@ietf.org>, "'Bernard Aboba'" <bernard_aboba@hotmail.com>, "'David Singer'" <singer@apple.com>, "'Harald Alvestrand'" <harald@alvestrand.no>
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Subject: Re: [rtcweb] [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
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This is a multi-part message in MIME format.

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Hi Alan,

=20

I have in previous email
http://www.ietf.org/mail-archive/web/tram/current/msg00304.html =
suggested
that both DISCUSS and draft-thomson-tram-turn-bandwidth-00.txt is taken =
off
the TRAM WG, but I think is shall be reintroduced after you clarify as =
you
do in=20

http://www.ietf.org/mail-archive/web/tram/current/msg00300.html :

Hi Karl, Thanks for your comments and feedback on the draft.

You are correct in saying that the BANDWIDTH extension is not about QoS. =
 It
is about fairness between users of a TURN server, and a TURN server =
being
able to indicate rate limiting policy to users. in the draft.

=20

The =93confusion=94 (to say least, see below) surrounding QoS within =
IETF has
lead me to protest and address the TRAM WG and RTCWEB WG and ADs with =
the
advise in =
http://www.ietf.org/mail-archive/web/tram/current/msg00304.html :

=93The TRAM WG and RTCWEB WG chairs and ADs are advised to review =
whether IETF
standard work in these WGs by contributors having an interest in more =
than
one of current or emerging: ISPs, carrier equipment vendors or web =
browser
makers, may discriminate any with an interest only in one of these. The =
same
should apply to non-activity by WG contributors to remedy such
discrimination opened by allowed proprietary usage of RFCs such as RFC
5285.=94=20

=20

This is because the introduction of the QoS related usage of RFC 5285 =
into
draft-ietf-rtcweb-rtp-usage in combinations with the Milestone 3 =
objectives
of TRAM, immediately would allow for:

(1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in
addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to
quickly allow us to finally enjoy the high quality and connectivity
(NAT/firewall traversal) of real-time communication services now =
possible,
for =20

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks=20

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of the Internet (including OTT).

=20

While the =93confusion=94 surrounding the QoS discussion in TRAM and =
RTCWEB
combined with the general QoS attitude within IETF =93it is all about
bandwidth=94 and =93it will go away with time=94 otherwise:

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. That all networks *will be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

*while* newer, fiber only type of networks, still can borrow bandwidth =
at no
extra cost, *by proprietary usage* of RFC 5285 by browsers.=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may have to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

=20

As Good doers, we should not allow this to happen.

=20

/Karl

=20

=20

Fr=E5n: Alan Johnston [mailto:alan.b.johnston@gmail.com]=20
Skickat: den 21 februari 2014 18:59
Till: Pal Martinsen (palmarti)
Kopia: tram@ietf.org; Oleg Moskalenko; Karl Stahl; Yoakum, John H (John)
=C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt

=20

I guess the way I would expect this to progress is for QoS discussions =
to
happen in another working group.  Once that other working group came to
consensus on an approach, and if that approach required STUN or TURN
extensions, then we would discuss mechanisms and possible milestones in
TRAM.

=20

- Alan -

=20

On Fri, Feb 21, 2014 at 3:37 AM, Pal Martinsen (palmarti)
<palmarti@cisco.com> wrote:

Hi,

=20

I agree the full QoS discussion should _not_ happen in TRAM. If you are
interested in helping out in that area I suggest you looking into the =
AEON
mailing list at: https://www.ietf.org/mailman/listinfo/aeon . They are
currently working on a problem-statement draft and a use-case draft, any
input to those would be very helpful.
(http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01,
http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00).

=20

That said, STUN have a few nice characteristics that makes it a perfect
candidate for transporting some of the QoS information.  IMHO that would =
be
extending the STUN spec and should be within the TRAM charter.  The main
goal of draft-martinsen-tram-discuss was to show how already existing =
QoS
mechanisms could be transported with STUN to provide more value, and to
start the discussion if TRAM is the appropriate place to have those on =
the
wire format discussions.

=20

.-.

P=E5l-Erik

=20

=20

On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <
<mailto:yoakum@avaya.com> yoakum@avaya.com> wrote:





+1

I fully agree with the comments that QOS should be a low priority for =
the
initial focus of the TRAM efforts.  There are other groups doing QOS =
work
and frankly I engage in WebRTC multimedia interactions daily over the
Internet, enterprise VPNs, and various combinations and seldom suffer
egregious quality issues.  I am more concerned about carriers doing =
things
to regulate or degrade WebRTC flows than a failure of existing Internet
mechanisms to enable them.

=20

Significant focus on QOS before we better enable TURN to be easily used =
in a
browser environment taking advantage or normal web characteristics (as
opposed to historic telephony constructs) would seem to be highly
distracting at this point.

=20

=20

Cheers,

John

=20

AVAYA
1.919.425.8446

=20

From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
Sent: Thursday, February 20, 2014 12:43 PM
To: Alan Johnston
Cc: Karl Stahl; tram@ietf.org
Subject: Re: [tram] Fwd: I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

=20

=20

On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <
<mailto:alan.b.johnston@gmail.com> alan.b.johnston@gmail.com> wrote:

=20

=20

=20

Personally, I am not sure how much QoS is actually in scope for TRAM. =
Have
you been following RMCAT where congestion avoidance for RTP is being
developed?  I see some overlap in your goals and the goals of that work.

-

=20

=20

I'd concentrate on the TURN application-level functionality, for now, =
and
I'd leave QoS for the future discussions.

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

=20

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Alan,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have in previous email <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00304.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00304.html</a> =
suggested that both DISCUSS and draft-thomson-tram-turn-bandwidth-00.txt =
is taken off the TRAM WG, but I think is shall be reintroduced after you =
clarify as you do in <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00300.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00300.html</a> =
:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>Hi Karl, =
Thanks for your comments and feedback on the =
draft.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>You =
are correct in saying that the BANDWIDTH extension is not about QoS. =
&nbsp;It is about fairness between users of a TURN server, and a TURN =
server being able to indicate rate limiting policy to users. =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>in=
 the draft.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e &#8220;confusion&#8221; (to say least, see below) surrounding QoS =
within IETF has lead me to protest and address </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>th=
e TRAM WG and RTCWEB WG and ADs with the advise in <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00304.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00304.html</a> =
:<o:p></o:p></span></p><p class=3DMsoNormal><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span></i><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e TRAM WG and RTCWEB WG chairs and ADs are advised to review whether =
IETF standard work in these WGs by contributors having an interest in =
more than one of current or emerging: ISPs, carrier equipment vendors or =
web browser makers, may discriminate any with an interest only in one of =
these. The same should apply to non-activity by WG contributors to =
remedy such discrimination opened by allowed proprietary usage of RFCs =
such as RFC 5285.</span></i><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221; </span></i><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></i></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is is because the </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>in=
troduction of the QoS related usage of RFC 5285 into </span><span =
lang=3DEN-US>draft-ietf-rtcweb-rtp-usage </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>in=
 combinations with the Milestone 3 objectives of TRAM, immediately would =
allow for:<o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
*<b>all</b>* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in =
addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to quickly allow us to finally enjoy the high quality and =
connectivity (NAT/firewall traversal) of real-time communication =
services now possible, for &nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
*<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wh=
ile the &#8220;confusion&#8221; surrounding the QoS discussion in TRAM =
and RTCWEB combined with the </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ge=
neral QoS attitude within IETF &#8220;it is all about bandwidth&#8221; =
and &#8220;it will go away with time&#8221; </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ot=
herwise:</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time traffic.<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. That all networks *<b>will be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time usage<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>*<=
b>while</b>* newer, fiber only type of networks, still can borrow =
bandwidth at no extra cost, *<b>by proprietary usage</b>* of RFC 5285 by =
browsers. <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may have to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>As=
 Good doers, we should not allow this to happen.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><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=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Alan =
Johnston [mailto:alan.b.johnston@gmail.com] <br><b>Skickat:</b> den 21 =
februari 2014 18:59<br><b>Till:</b> Pal Martinsen =
(palmarti)<br><b>Kopia:</b> tram@ietf.org; Oleg Moskalenko; Karl Stahl; =
Yoakum, John H (John)<br><b>=C4mne:</b> Re: [tram] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>I guess =
the way I would expect this to progress is for QoS discussions to happen =
in another working group. &nbsp;Once that other working group came to =
consensus on an approach, and if that approach required STUN or TURN =
extensions, then we would discuss mechanisms and possible milestones in =
TRAM.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>- =
Alan -<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Fri, Feb 21, 2014 at 3:37 AM, Pal Martinsen =
(palmarti) &lt;<a href=3D"mailto:palmarti@cisco.com" =
target=3D"_blank">palmarti@cisco.com</a>&gt; =
wrote:<o:p></o:p></p><div><div><p =
class=3DMsoNormal>Hi,<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
agree the full QoS discussion should _not_ happen in TRAM. If you are =
interested in helping out in that area I suggest you looking into the =
AEON mailing list at:&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/aeon" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/aeon</a> . They =
are currently working on a problem-statement draft and a use-case draft, =
any input to those would be very helpful. (<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01" =
target=3D"_blank">http://tools.ietf.org/html/draft-eckel-aeon-use-cases-0=
1</a>,&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00"=
 =
target=3D"_blank">http://tools.ietf.org/html/draft-eckel-aeon-problem-sta=
tement-00</a>).<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>That said, STUN have a few nice characteristics that =
makes it a perfect candidate for transporting some of the QoS =
information. &nbsp;IMHO that would be extending the STUN spec and should =
be within the TRAM charter. &nbsp;The main goal =
of&nbsp;draft-martinsen-tram-discuss was to show how already existing =
QoS mechanisms could be transported with STUN to provide more value, and =
to start the discussion if TRAM is the appropriate place to have those =
on the wire format discussions.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>.-.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>P=E5l-Erik<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On 20 Feb 2014, at 19:55 pm, =
Yoakum, John H (John) &lt;</span><a href=3D"mailto:yoakum@avaya.com" =
target=3D"_blank"><span lang=3DEN-US>yoakum@avaya.com</span></a><span =
lang=3DEN-US>&gt; wrote:<o:p></o:p></span></p></div><p =
class=3DMsoNormal><span =
lang=3DEN-US><br><br><o:p></o:p></span></p><div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I fully agree with the comments that QOS should be a low priority for =
the initial focus of the TRAM efforts.&nbsp; There are other groups =
doing QOS work and frankly I engage in WebRTC multimedia interactions =
daily over the Internet, enterprise VPNs, and various combinations and =
seldom suffer egregious quality issues.&nbsp; I am more concerned about =
carriers doing things to regulate or degrade WebRTC flows than a failure =
of existing Internet mechanisms to enable them.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Significant focus on QOS before we better enable TURN to be easily =
used in a browser environment taking advantage or normal web =
characteristics (as opposed to historic telephony constructs) would seem =
to be highly distracting at this point.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Ch=
eers,</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Jo=
hn</span></i><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:navy'>&nb=
sp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:red'>AVAY=
A</span><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><br></span><span lang=3DEN-US =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif";color:teal'><a =
href=3D"tel:1.919.425.8446" =
target=3D"_blank">1.919.425.8446</a></span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&nbsp;tram =
[<a href=3D"mailto:tram-bounces@ietf.org" =
target=3D"_blank">mailto:tram-bounces@ietf.org</a>]&nbsp;<b>On Behalf =
Of&nbsp;</b>Oleg Moskalenko<br><b>Sent:</b>&nbsp;Thursday, February 20, =
2014 12:43 PM<br><b>To:</b>&nbsp;Alan Johnston<br><b>Cc:</b>&nbsp;Karl =
Stahl; <a href=3D"mailto:tram@ietf.org" =
target=3D"_blank">tram@ietf.org</a><br><b>Subject:</b>&nbsp;Re: [tram] =
Fwd: I-D Action: draft-thomson-tram-turn-bandwidth-00.txt</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On Thu, Feb 20, 2014 at 6:26 AM, =
Alan Johnston &lt;<a href=3D"mailto:alan.b.johnston@gmail.com" =
target=3D"_blank"><span =
style=3D'color:purple'>alan.b.johnston@gmail.com</span></a>&gt; =
wrote:<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div><div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>Personally, I am not sure how much =
QoS is actually in scope for TRAM. Have you been following RMCAT where =
congestion avoidance for RTP is being developed? &nbsp;I see some =
overlap in your goals and the goals of that =
work.<o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#888888'>-</span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#888888'>&nbsp;</span><span =
lang=3DEN-US><o:p></o:p></span></p></div></div></div></div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span lang=3DEN-US>I'd =
concentrate on the TURN application-level functionality, for now, and =
I'd leave QoS for the future =
discussions.<o:p></o:p></span></p></div></div></div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif"'>__________=
_____________________________________<br>tram mailing list<br><a =
href=3D"mailto:tram@ietf.org" target=3D"_blank">tram@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/tram" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/tram</a><o:p></o:=
p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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From nobody Tue Feb 25 00:39:47 2014
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Date: Tue, 25 Feb 2014 09:39:41 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Karl Stahl <karl.stahl@intertex.se>
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Karl,
(As individual)

I also wish for more usable QoS mechanisms. However, I don't see that
being achieved by your proposal. In addition I agree with Colin about
the issues with putting the information in the RTP header extension. I
would also note that this would be very RTP specific, and not at all
help with the data channels multiple streams and their priorities. There
might be data channel information that is more crucial than any of the
RTP media stream packets.

You are pushing for a small piece in the middle. A piece that will not
help with the more general issue of QoS. How does the application and
the multiple ISPs that carries the traffic reach an agreement on what
properties that can be provided, that the application in this instance
have the right to request those properties and that any cost is
correctly associated with the user or the user's agent in regards to
carrying the associated cost.

When it comes to setting DSCP from user land in the OS, that is
restricted due to the security implications. If those implications where
resolved, then OS could open up those interfaces.  There are many
interlinked reasons why things look like they do today. I don't believe
in tugging on a single random thread in ball of yarn and hope that it
comes out without any knots and ties on it.

Show me the framework for the QoS functions you have in mind that at
least has less issues than the currently deployed and take care of at
least some of the bigger issues. If that requires information in the RTP
streams for some reasons, then let us talk about how to best encoded it.
But, we need that framework first, the architecture that makes this a
better solution than the current Diffserv architecture or any other QoS
architecture.

Cheers

Magnus Westerlund

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From nobody Tue Feb 25 01:08:54 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Richard Ejzak" <richard.ejzak@gmail.com>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi Raju,

>> Anyway, can you give me an example of a case where you want to use=20
>> SDP, and where you need to negotiate the re-transmission values etc?
>
> [Raju] A simple example would be 2 browsers talking thru an intermedia pr=
oxy, which wants to know what protocols are
> being negitiated and used as part of the session.

Sure. I don't question the use-case/need for using SDP to negotiate channel=
 usages.

> 1. Calling client creates data channel (using http://dev.w3.org/2011/webr=
tc/editor/webrtc.html#idl-def-RTCDataChannel) with desired
> attributes and sets negotiated=3Dtrue. So, calling browser saves the attr=
ibutes for the data channel but won't use DCP.
> 2. These attributes are sent via SDP to peer client (via proxy).
> 3. Peer client sets these same attributes while creating data channel and=
 sets negotiated=3Dtrue but won't do DCP.
> 4. Peer client echo the same attributes back in SDP answer.
>
> Now, data channel stacks on both ends will use same attributes, similar t=
o DCP use case, with the only difference being DCP is not used here.

My question was regarding the re-transmission values etc.=20

Why do the peers need to negotiate those? Why would an intermediary need to=
 know about those? Why not simply define those in the protocol description,=
 and each endpoint supporting the protocol will then know what values to us=
e?

Regards,

Christer



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On 2014-02-24 19:58, Martin Thomson wrote:
> On 24 February 2014 02:05, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> My position is that the above mitigation is unlikely to have significant
>> impact on this attack.
> 
> That was my sense too.
> 
>> There are two reasons I wanted this attack to be considered. First, it
>> provides a clear requirement on having congestion control as first level
>> mitigation.
> 
> Congestion control.  Check.
> 
>> Secondly, I think this could become a significant issue as data channel
>> PeerConnections can be opened without user consent. A malicious JS that
>> is sufficient well spread with a well working find and connect could
>> establish a large mesh of peer connections that could come close to
>> saturate each endpoints local access link, resulting in very heavy loads
>> on the network, even with congestion control. With congestion control
>> you can likely prevent congestion collapse, but you should be fully
>> capable of driving a network into a state of "mostly useless",
>> especially a network suffering from buffer bloat inside of the ingress
>> nodes.
> 
> So the concern is that even with congestion control, a single bad
> actor can use more than their "fair" share of network resources.
> 
> Sadly, this is already true on the web.
> 
> No harm in writing down the potential.  Are you looking for anything
> more than that?

No, I think the main thing is to have something to point at why
congestion control is a MUST. Secondly, at least make people aware that
this mechanism may give a malicious entity more potential for driving
traffic load where they want to cause it than some of the previous
mechanism, due to the ease of finding endpoint and keeping traffic more
localized than in other DDoS attacks. But, sure it will more difficult
to drive the targeted network into completely uselessness.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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On 2014-02-19 19:11, Harald Alvestrand wrote:
> On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
>> On 2014-02-18 23:58, Harald Alvestrand wrote:
>>> I may be a little simple-minded, but if we have a recommendation that
>>> receivers MUST be able to receive all packetizations of G.711 and OPUS
>>> up to 200 ms per packet, and that receivers should signal this by
>>> sending "a=maxptime:200" in their SDP, what situations exist where
>>> interoperability is not going to be achieved?
>> Interoperability is going to be achieve in the direction towards this
>> endpoint. The question is if we achieve interoperability in the
>> direction from that endpoint.
> 
> So, given that there is nothing in the G.711 specification about which
> sample sizes a G.711 receiver has to support - the right approach seems
> to be to amend the G.711 MIME registration with this information.

Yes, it would for the future. But considering the wide-spread deployment
of this payload format I don't see that having any short term effect on
the interoperability.

Also, the recommendations are actually context dependent. Thus, it is
not obvious that a single recommendation is suitable.

> 
> This is not a WebRTC issue; it is an absence of specification for the
> non-WebRTC devices that use G.711. The RTCWEB specifications can
> reasonably be expected to point to existing information about this
> issue; it is not reasonable (in my mind) that the RTCWEB WG decide what
> these values should be.
> 

I would argue for this consideration based on that this is done in the
WebRTC context, not any other context.

Cheers

Magnus Westerlund

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On 2014-02-19 19:47, Harald Alvestrand wrote:
> On 02/19/2014 11:08 AM, Magnus Westerlund wrote:
>> Hi,
>>
>> I have reviewed the Transports for RTCWEB
>> (draft-ietf-rtcweb-transports-02)
> Thanks!
>>  and have the following comments.
>>
>> 1. Section 1.
>>
>>  This document focuses on the data
>>    transport protocos that are used by conforming implementations.
>>
>> Missing "l" in protocos.
> Fixed.
>>
>> 2. Section 3.1:
>>    For both protocols, IPv4 and IPv6 support is assumed; applications
>>    MUST be able to utilize both IPv4 and IPv6 where available.
>>
>> What "applications" are intended in this sentence. I get a feeling that
>> "applications" here could be replaced by "WebRTC implementations"
> 
> Javascript applications. I think I should add the word "application" to
> -overview's vocabulary; it is actually used in that meaning in that
> document (also in the forms "Web application" and "browser-based
> application".

Sorry, then I find the above requirement a bit strangely targeted. I
think we can require that IPv4 and IPv6 support for the applicable
protocol pieces in what we define. But, on the JS application level?
Also what explicit addressing information is visible on that level.

It might be that this requirement needs to be split or at least apply to
multiple levels.

> 
>>
>>
>> 3. Section 3.1:
>>    For UDP, this specification assumes the ability to set the DSCP code
>>    point of the sockets opened on a per-packet basis, in order to
>>    achieve the prioritizations described in
>>    [I-D.dhesikan-tsvwg-rtcweb-qos] when multiple media types are
>>    multiplexed.  It does not assume that the DSCP codepoints will be
>>    honored, and does assume that they may be zeroed or changed, since
>>    this is a local configuration issue.
>>
>> I wonder if this should be moved into 3.2 section instead. At least the
>> part in 3.1 should focus on the DSCP setting part on the transport flow.
>> The implications of it working or not should be moved to 3.2. A forward
>> pointer to that second can be used instead.
> 
> I think it fits better in 3.1 - this is about the assumptions that the
> specifcation makes about the parts of the system that it is not a
> specification for.

I am fine with the first sentence staying, it is the second part that
needs more discussion and I think it should be moved and expanded on.
Thus a forward pointer may be appropriate here.

> 
>>
>> 4. Order of Section 3.2 and 3.3.
>>
>> I think the order should be swapped between 3.2. and 3.3. That way the
>> Multiplexing section can provide definitions of all the stream concepts
>> that exists in WebRTC and how they can be combined. More about this later.
>>
>> 5. Content of Section 3.2
>>
>> This section needs to be expanded to start with a general discussion of
>> how QoS can be applied to get a benefit. But, I think one need to be
>> clear that it can't be assumed to be available in implementations or
>> WebRTC based applications.
> 
> I want to push back on this. It seems inappropriate that the RTCWEB QoS
> specification start off with a tutorial on what QoS is and how one can
> use it to gain a benefit. That material should be available elsewhere.
> (Recommendations?)

I would like to agree with you, however we are putting together the
pieces in a way that stands some of the previous assumptions on their
heads. Thus, we don't have easy access to pointers.

I can see this discussion going into
draft-ietf-avtcore-multiplex-guidelines when it concerns RTP. However,
you also have the data channel to consider. Thus, there exist no
references that I know that discusses this particular combination and
its implications.

> 
>>
>> It needs to also discuss flow-based QoS mechanism. What are the
>> implications if that is what is available and what choices does one have
>> to address this by configuring the multiplexing different.
> 
> I do not want to discuss material for which there are no references.
> What particular flow-based mechanisms do you have in mind, and what are
> the relevant references?

RSVP we can start with, but I think equally applicable is 3GPP QoS. But
finding the right reference here might be a bit problematic.

I think one can cover the general property of flow based QoS in a single
sentence. However, that usage has implications for the peer connection
and its configuration. We have text in the rtp-usage that makes it clear
about the requirement to be able to send multiple RTP sessions on
different flows. But the combination with the data channel is not
covered. Thus, I think that belongs in the transports document. The
question is how much lead in material you need for that discussion?

> 
>>  I think this
>> is the logical place to take the main discussion of that, as it can
>> address both RTP streams and the data channel together. Please consider
>> pointers to appropriate discussion in Section 12.1.3 in
>> draft-ietf-rtcweb-rtp-usage.
> 
> I do not think this document should repeat any material from rtp-usage,
> but rtp-usage does not consider any aspect of QoS marking for multiple
> streams using the same 5-tuple, nor does it say anything about
> flow-based or DPI-based mechanisms apart from noting their existence.

No, I am not talking about repeating. I am discussing the requirements
on being able to separate the data channel on transport level (UDP) from
the RTP sessions over their RTP, as well as being able to put them on
common UDP flows.

> 
> Agree that the multiplexing aspect should be discussed somewhere, but
> I'm not sure where.
>>
>> I do note that writing this section appropriately is difficult until the
>> content of draft-dhesikan-tsvwg-rtcweb-qos has firmed up.
> 
> True.
>>
>> 6. Content of Section 3.3:
>>
>> I think this section should focus on making clear what different types
>> of streams that exist in the WebRTC context, and what choices of
>> multiplexing that are possible. Basically ensure that the full set of
>> combinations are made clear.
> 
> Isn't this material that really should be in
> draft-ietf-avtcore-rtp-multi-stream or

This is not appropriate place, this only discusses multiple SSRCs in the
same RTP session.

> draft-ietf-avtcore-multiplex-guidelines?

This one definitely needs such discussion yes.

Also draft-ietf-avtcore-multi-media-rtp-session has some parts of this,
due to the potential for different needs based on media type.

> 
> Again, this seems the wrong draft for having a discussion of those issues.

However, adding the data channel into this, is something WebRTC
specific, thus you are not getting away from dealing with it.

>>
>> A. All RTP packet streams and data channel over a single UDP flow.
>>
>> B. All RTP packet streams over one UDP flow and the data channel over
>> another UDP flow.
>>
>> C. The RTP packet streams over different UDP flows based on content type
>> and separate data channel.
>>
>> D. Each RTP packet stream over its own UDP flow (RTP session) and the
>> data channel over its own UDP flow.
>>
>> The above with the additional dimension, that each RTP sessions, RTCP
>> can be in its own UDP flow.
>>
>> There is also a question if the data channel can be aggregated with only
>> one RTP session, where there are multiple ones configured.
>>
>> The QoS related discussion of this can be moved to the QoS section to
>> keep that focused.
>>
>> However, the discussion of the pro and cons with having one or more flow
>> is good and could even be expanded to list reasons such as legacy
>> interop over a gateway.
>>
>> 7. Section 3.3:
>>    RTCWEB implementations MUST support the ability to send and receive
>>    multiple SSRCs on the same transport, and MUST support the ability to
>>    send and receive multiple SSRCs on multiple simultaneous transports,
>>    including the ability to send and receive audio and video on the same
>>    transport.
>>
>> I think this sentence is restating what is already required by Section
>> 4.1 in draft-ietf-rtcweb-rtp-usage: If you want to enforce this, I
>> suggest to do that by reference to that text rather than using RFC 2119
>> normative statements.
> 
> Yes, rtp-usage contains these requirements. I can delete it from here.
> Thanks!
> 
> However, this removes the handle on which I hung the discussion of why
> one would want to multiplex or not, which I inserted because you
> requested it. This discussion also seems to be present in rtp-usage, so
> it seems appropriate to delete that too.

As I argue above, there is a need to take a full WebRTC level discussion
of the multiplexing issues. I think the appropriate here is to try to
ensure the AVTCORE documents contains the relevant pieces so you know
what to reference and then try to put togheter a high level RTP + Data
channel multiplexing considerations in this document.

> 
>>
>>
>> 8. Section 3.4:
>>    ICE [RFC5245] MUST be supported.  The implementation MUST be a full
>>    ICE implementation, not ICE-Lite.
>>
>>    In order to deal with situations where both parties are behind NATs
>>    which perform endpoint-dependent mapping (as defined in [RFC5128]
>>    section 2.4), TURN [RFC5766] MUST be supported.
>>
>> These paragraph implies STUN and TURN server configuration. Should that
>> configuration question be discussed with the appropriate pointers to the
>> API?
> This document has no pointers to the API at the moment, and I do not
> want to add them - it's a layering violation of sorts. We can add
> pointers to JSEP, which is kind of close to the API.
> 
> But the main point seems to be that STUN and TURN servers MUST be
> configurable, both from the browser setup and from the (JS) application.
> I'm adding a sentence saying that.

That is probably sufficient.

> 
>>
>> 9. Section 3.4:
>>    In order to deal with firewalls that block all UDP traffic, TURN
>>    using TCP between the client and the server MUST be supported, and
>>    TURN using TLS between the client and the server MUST be supported.
>>    See [RFC5766] section 2.1 for details.
>>
>> I think the following part: "TURN using TLS between" should be changed
>> over "TURN using TLS over TCP between" to make it clear that this is
>> using TLS/TCP rather than TLS/FOO.
> Done.
>>
>> 10. Section 3.4:
>>    TURN TCP candidates [RFC6062] SHOULD be supported; this allows
>>    applications to achieve peer-to-peer communication when both parties
>>    are behind UDP-blocking firewalls using a single TURN server.  (In
>>    this case, one can also achieve communication using two TURN servers
>>    that use TCP between the server and the client, and UDP between the
>>    TURN servers.)
>>
>> My understanding of RFC6062 is that it allows one to establish a vanilla
>> TCP connection on the far side of the TURN server as well as connecting
>> that one to an almost vanilla TCP connection (just a initial TURN
>> ConnectionBind request) that A establish to the TURN server, i.e.
>> WebRTC-A -(Turn/TCP)-> TURN_Server -(TCP)-> WebRTC-B.
>>
>> The end-result is that a relayed vanilla TCP connection is established
>> between the WebRTC endpoints (A and B). As the WebRTC transport so far
>> defined only works with datagrams, i.e. DTLS or RTP/RTCP packets they
>> can't be sent natively over a TCP byte stream. Thus, if this mode of
>> operation is to be supported a framing is required on this leg.
>>
>> A possibility here could be RFC 4571 if one want to go this way.
> 
> I added a 4571 reference, and a note saying this is up for discussion.
> If we frame RTP over TCP, we also have to decide what we do about the SCTP.

Yes, I get the impression that the people arguing for the WebRTC over
IP/TCP has not thought through the implications and I definitely like a
discussion of this.

Also, in which context are you wondering over SCTP in the above?

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Some editorial comments on section 5
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Hi,

>> Q1:
>>=20
>> Section 5 says:
>>=20
>> 	"SCTP as specified in [RFC4960] MUST be used in
>>   	combination with the extension defined in [RFC3758] and provides the
>>   	following interesting features for transporting non-media data
>>   	between browsers:"
>>=20
>> I suggest to say:
>>=20
>> 	"The SCTP extension defined in [RFC3758] MUST be supported, and
>>     	provides the following features:
>>
> But this implies that the listed features are provided by the extension d=
efined in RFC 3758, which is not true. The features are provided by the com=
bination of RFC 4960 and RFC 3758.

Correct. My mistake.

So, my suggestion would then be to remove "as specified in" and "interestin=
g".

 	"SCTP [RFC4960] MUST be used in combination with the extension defined in=
 [RFC3758] and=20
	provides the following features for transporting non-media data between br=
owsers:"

The reference implicitly means "as specified in", in my opinon :)

----------

>> Q3:
>>=20
>> Section 5 says:
>>=20
>> 	"This value can be used to multiplex multiple protocols
>>   	over a single SCTP association.  The sender provides for each
>>   	protocol a specific PPID and the receiver can demultiplex the
>>   	messages based on the received PPID."
>>=20
>> I think we discussed this earlier. The PPID value CAN NOT (at least not =
as used by WebRTC) be used to multiplex multiple protocols=20
>> over a single SCTP association. If you create multiple channels, for mul=
tiple protocols, the same PPID values may be used for each protocol.
>>=20
>> The PPID CAN, however, be used within a data channel, e.g. to demultiple=
x the DCP from channel data.
>
> I guess the problem is that I consider 'protocol' as something running on=
 top of SCTP like DCEP, Binary of String data, you consider 'protocol'=20
> as something running on top of Binary of String data and the 'protocol' b=
eing identified in the JS protocol string.
>
> What about using:
>
> <t>Each SCTP user message contains a Payload Protocol Identifier (PPID) t=
hat is passed to SCTP by its upper layer on the sending side and provided t=
o its upper layer on the receiving side.
> The PPID can be used to multiplex/demultiplex multiple upper layers over =
a single SCTP association.
> In the WebRTP context, the PPID is used to distinguish between
> UTF-8 encoded user data, binary encoded userdata and the Data Channel Est=
ablishment Protocol defined in <xref target=3D'I-D.ietf-rtcweb-data-protoco=
l'/>.
> Please note that the PPID is not accessable via the Javascript API.</t>
>
> Does this address your issue?

Instead of talking about SCTP associations, could we talk about data channe=
ls? I.e. the PPID can be used to multiplex/demultiplex multiple upper layer=
s within a single data channel?

Because, if you have multiple data channels on a single SCTP association, y=
ou will use the stream IDs  - not the PPIDs - to multiplex/demultiplex betw=
een the channels.

----------

>> Q5:
>>=20
>> Sometimes you say "SCTP" and sometimes "SCTP as defined in [RFC4960]".
>>=20
>> I suggest to e.g. use the reference on the first occurrence, and then on=
ly use "SCTP".
>
> I double checked:
> SCTP as defined in [RFC4960] is used in
> * In the Introduction (first occurrence) talking about SCTP / UDP
> * In the Introduction to make sure which specifications are used: RFC 496=
0 and RFC 3758

In the 2nd paragraph of the Introduction, the text says:

	"SCTP as specified in [RFC4960] with the partial reliability extension def=
ined in [RFC3758]"

Could you remove "as specified in", and say?

	"SCTP [RFC4960] with the partial reliability extension defined in [RFC3758=
]"


> * In section 5 when used with RFC 2119 language.

I still think you could remove "as specified in", and say:

	"SCTP [RFC4960] MUST be used in combination with the extension defined..."

> * In section 6 to make clear what relates to RFC 4960 and what to the NDA=
TA extension.
>
> All of these are intentional. Which one do you think is not necessary?

In section 6.6, I think you could remove "specified in", and say:

	"The SCTP base protocol [RFC4960]..."


I am not religious about these things, but again: in my opinion adding the =
reference means "as specified in", and makes the text easier to read :)

Regards,

Christer


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On 2014-02-23 10:54, Barry Dingle wrote:
> In reference to your first _Section 1 comments_, I agree that we need to
> clearly differentiate the SRTP media and the SCTP media. (Note - SRTP -
> not RTP !)

So SRTP is an extension of RTP, in fact it is a profile, or rather two
(RTP/SAVP or RTP/SAVPF). I wrote RTP because I meant no specific RTP
configuration here, it can be any of the profiles and any of the
extensions. I know and fully support that RTCWEB uses RTP/SAVPF for RTP
transported media, however in the wider context and possible other usage
of the defined mechanism this may or may not be true. Therefore I see a
need to be generic in that discussion. If one want to emphasize the SRTP
part, then use (S)RTP.

> 
> I agree that there could be real-time media that can be sent over SCTP
> Channels are still meet the criteria for 'real-time' as being 'received
> within a few hundred milliseconds of being generated' - see WebRTC
> Overview section 2.4.
> 
> Maybe we need to differentiate the Channel types - and then refer to the
> data going over those Channels. For instance,if we define "SRTP
> Channels" and "SCTP Channels", we could then refer to 'SRTP Channel
> media' which would have a specific meaning.

Well, around RTP we aren't discussing "channels" at all. Thus, this
becomes very ambiguous to what it means, unless you have a definition to
go along with it.

In the context of a PeerConnection we can have one or more RTP session.

Also, I don't know what an SCTP channel is. There is an SCTP
association, which is something between two endpoints. On top of this we
have the SCTP Data channels. So what are you missing or want from these
definitions?


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>
Thread-Topic: [tram] [rtcweb] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
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On 24 Feb 2014, at 23:22 pm, Karl Stahl <karl.stahl@intertex.se<mailto:karl=
.stahl@intertex.se>> wrote:

Hi P=E5l,

You did not comment nor answer, in response to any of:
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html
http://www.ietf.org/mail-archive/web/tram/current/msg00273.html

Sorry about that. I did not notice any clear items to answer or comment on.

whether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the appli=
cation/browser to transfer relevant real-time traffic information (types an=
d bandwidth) into every RTP package by using the already IETF standardized =
RTP extension header RCF 5285, which could be used by any current or future=
 Internet (including OTT) network where WebRTC real-time traffic may flow. =
It seems to have been standardized already 2008 to allow such usage.

draft-martinsen-tram-discuss is intended to solve internal application stre=
am prioritisation pr flow, i.e. it is more important for me to get the vide=
o stream across than my application stream at this moment in time. Its inte=
ntion is to describe the 5-topple the STUN message was sent on.

Moving to a pr packet prioritisation is interesting as it can solve the bun=
dle problem, and mark the relative importance of specific video frames (I-F=
rames for example). Since its pr packet the network element might not keep =
stream state so it might be easier that way. But the information needs to b=
e in a fixed offset for quick network element processing. And we want to de=
scribe other streams than just RTP. That makes RTP header extensions unsuit=
able.

QoS related step C) draft-martinsen-tram-discuss can then be taken out of T=
RAM, and step D) (that was never intended to be handled by TRAM anyway) wil=
l be handled elsewhere.

I was hoping to argue that the draft is only new signalling of old concepts=
 (DSCP and ECN). In the end it is the WG that decides if there is enough in=
terest.


I have repeated my suggestion to RTCWEB WG from the October discussions on =
the relevant [rtcweb] [avtext] [mmusic] lists http://www.ietf.org/mail-arch=
ive/web/rtcweb/current/msg09129.html to introduce the QoS related step D) i=
nto draft-ietf-rtcweb-rtp-usage suggesting usage of RFC 5285, where traffic=
 information in RTP packets simply would be filled by any browser under any=
 OS.

Without step B), TRAM =93Enforcing the real-time traffic through the offere=
d/discovered TURN servers=94, the real-time traffic may happen through the =
IP default gateways often congested by data traffic and QoS insensitive str=
eaming video and file sharing. Without specific QoS methods at those points=
, the network raw bandwidth capacity may have to be 10-folded to achieve su=
fficient QoE when WebRTC usage becomes popular. Methods like DISCUSS addres=
ses such things occurring when STUN instead TURN is used (by allowing flows=
 into such congestion points), but only provides direct traffic info for ou=
tgoing (not incoming) real-time traffic and locally, therefore are not even=
 applicable to reservation type of networks like Cable Networks and Mobile =
OTT.

If both clients supports draft-martinsen-tram-discuss you would get an bi-d=
irectional end to end description of the flow that works through NAT.
The information in the STUN packet can be used to remark DSCP bits to whate=
ver values that is of best use for the network the packet is flowing though=
.

I dont see how TURN gives that ability. Once the channel is bound you only =
have a channel number in front of the UDP payload. And TURN is designed to =
carry alls sort of traffic. For this to work there must be some sort of onl=
y webRTC traffic is allowed through this TURN server. (And Ill bet bitorren=
t or others find a way to mimic webRTC to set up only a data channel..)


That would leave certain network types to investing in raw bandwidth increa=
se, instead of simply borrowing the bandwidth (from much larger data traffi=
c and QoS insensitive streaming video and file sharing at no adverse effect=
) and making that borrowed no-cost bandwidth very valuable for the carriers=
 when delivering to their customers.
I know a few of you Cisco guys are among the ones that have been fighting a=
gainst the general =93it is all about bandwidth=94 and =93it will go away w=
ith time=94-attitude within IETF work, e.g. see http://www.ietf.org/mail-ar=
chive/web/rtcweb/current/msg09031.html, but the pointers given in the curre=
nt QoS discussion within TRAM:

(i) will *not allow* ISP=92s to use already available and currently deploya=
ble quality IP pipes for real-time traffic to also be used for WebRTC gener=
ated real-time traffic.

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile OT=
T) to borrow bandwidth from data traffic and QoS insensitive streaming vide=
o and file sharing. All networks will *also be inhibited* from the very com=
mon and used method of simply providing an extra IP pipe (often provided ov=
er the same wire but level-2 separated) dedicated for real-time usage
(*by resisting TRAM implementation of step B*)
Newer, fiber only type of networks, can still borrow bandwidth at no extra =
cost, *by proprietary usage* of RFC 5285 by browsers.

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase th=
at may has to be 10-folded to reach sufficient QoE when WebRTC usage become=
s popular (if at all possible, since unmanaged IP pipes intermittently are =
filled, whatever bandwidth is available).

Further explainations about QoS methods and their implementation were given=
 a few days ago in:
http://www.ietf.org/mail-archive/web/tram/current/msg00274.html

(i), (ii) and (iii) will *seriously delay* usage of telepresence capable an=
d quality demanding WebRTC (bandwidth being around 30 times higher over bes=
t effort Internet, than telephony over quality managed networks) and will *=
vastly increase cost* for ISPs to offer WebRTC with good QoE.

Below you are giving pointers saying =93They are currently working on a pro=
blem-statement draft and a use-case draft, any input to those would be very=
 helpful.=94 Those are unnecessary, risking leading to (i), (ii) and (iii) =
above.

We find them quite useful. It allows us to engage with potential users and =
see if there are real problems to solve. If the customer/users do not see a=
 problem there is little incentive for us to invest in creating a solution =
that nobody wants. That said, it is always the danger of crating a to compl=
icated solution that solves to many problems.

So are pointers such as: =93RMCAT where congestion avoidance for RTP is bei=
ng developed=94. TRAMs Milestone 3 is for the purpose of directing real-tim=
e traffic where congestion control isn=92t already in place. UsingRFC 5285 =
available since 2008, to fill traffic information in RTP packets (step D)) =
is probably a necessity for most future =93congestion control=94 discussed.

There are some good example of RTP header extension usage. For example draf=
t-avtext-berger-framemarking. But that information may very well be encrypt=
ed and unavailable to the network element.


.-.
P=E5l-Erik







This chicken-and-egg circular also applies to RTCWEB direction of QoS issue=
s to:
http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos
focusing on DSCP-mapping without even mentioning RFC 5285 available since 2=
008, to fill traffic information in RTP packets where it  is a necessity an=
d will allow:
(1) applications to directly convey QoS related real-time traffic info to t=
he network at points where RTP flow is directed to by TRAM Milstone 3, to b=
e used by *any network element implementing any suitable QoS methods for th=
e particular network* for
(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), under=
 *all* OSs, and *all* current and future IP networks
(3) *without* having to be forced into application specific networks (PSTN,=
 IMS) instead of using the Internet (including OTT).

The only activity required, is to call for ISPs=92 review whether the traff=
ic information transferred by RFC 5285 is sufficient for current and future=
 needs in their network as suggested in http://www.ietf.org/mail-archive/we=
b/rtcweb/current/msg09129.html
<snip>
=85two parameters (e.g. two bytes each) are encoded into the RTP header ext=
ension:

A) The maximum bandwidth requirement: Two bytes could contain everything fr=
om some bps for real-time text to Gbps for future 3D supersize telepresence=
=85 on a logarithmic scale.

B) The quality characteristics for the stream, with the highest bit set to =
1, we could allocate a bit each for quality type e.g:
Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay Insensit=
ive (e.g. video streaming), Minimum Delay, Reliable Delivery, Prioritize X,=
 Variation Y, that could be combined as required to describe the stream.

And with highest bit set to 0, there could instead be a number for special =
usage that does not fit the general description of the individual bits.
</snip>

Then this could be assigned numbers to have the RFC in place.
With TRAM milestone 3 also place,
market forces will drive ISPs and browser makers to implement just that, wi=
thout even having it MUST-established.
=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and
=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari comes w=
ith much better QoE?=94 and vice versa.
 l
Please see further emails soon following this one, for details and history.

/Karl

Fr=E5n: tram [mailto:tram-bounces@ietf.org] F=F6r Pal Martinsen (palmarti)
Skickat: den 21 februari 2014 10:37
Till: tram@ietf.org<mailto:tram@ietf.org>
Kopia: Karl Stahl; Oleg Moskalenko; Alan Johnston; Yoakum, John H (John)
=C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt

Hi,

I agree the full QoS discussion should _not_ happen in TRAM. If you are int=
erested in helping out in that area I suggest you looking into the AEON mai=
ling list at: https://www.ietf.org/mailman/listinfo/aeon . They are current=
ly working on a problem-statement draft and a use-case draft, any input to =
those would be very helpful. (http://tools.ietf.org/html/draft-eckel-aeon-u=
se-cases-01, http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-=
00).

That said, STUN have a few nice characteristics that makes it a perfect can=
didate for transporting some of the QoS information.  IMHO that would be ex=
tending the STUN spec and should be within the TRAM charter.  The main goal=
 of draft-martinsen-tram-discuss was to show how already existing QoS mecha=
nisms could be transported with STUN to provide more value, and to start th=
e discussion if TRAM is the appropriate place to have those on the wire for=
mat discussions.

.-.
P=E5l-Erik






On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <yoakum@avaya.com<mailto=
:yoakum@avaya.com>> wrote:


+1
I fully agree with the comments that QOS should be a low priority for the i=
nitial focus of the TRAM efforts.  There are other groups doing QOS work an=
d frankly I engage in WebRTC multimedia interactions daily over the Interne=
t, enterprise VPNs, and various combinations and seldom suffer egregious qu=
ality issues.  I am more concerned about carriers doing things to regulate =
or degrade WebRTC flows than a failure of existing Internet mechanisms to e=
nable them.

Significant focus on QOS before we better enable TURN to be easily used in =
a browser environment taking advantage or normal web characteristics (as op=
posed to historic telephony constructs) would seem to be highly distracting=
 at this point.


Cheers,
John

AVAYA
1.919.425.8446

From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
Sent: Thursday, February 20, 2014 12:43 PM
To: Alan Johnston
Cc: Karl Stahl; tram@ietf.org<mailto:tram@ietf.org>
Subject: Re: [tram] Fwd: I-D Action: draft-thomson-tram-turn-bandwidth-00.t=
xt



On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <alan.b.johnston@gmail.com<m=
ailto:alan.b.johnston@gmail.com>> wrote:



Personally, I am not sure how much QoS is actually in scope for TRAM. Have =
you been following RMCAT where congestion avoidance for RTP is being develo=
ped?  I see some overlap in your goals and the goals of that work.
-


I'd concentrate on the TURN application-level functionality, for now, and I=
'd leave QoS for the future discussions.


_______________________________________________
tram mailing list
tram@ietf.org<mailto:tram@ietf.org>
https://www.ietf.org/mailman/listinfo/tram


--_000_B788D545BDC047A7BE1B76E2A5F60509ciscocom_
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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;">
<br>
<div>
<div>On 24 Feb 2014, at 23:22 pm, Karl Stahl &lt;<a href=3D"mailto:karl.sta=
hl@intertex.se">karl.stahl@intertex.se</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Hi P=E5l,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">You did not comment nor answer, in response to any of:<o:=
p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;"><a href=3D"http://www.ietf.org/mail-archive/web/tram/curr=
ent/msg00275.html" style=3D"color: purple; text-decoration: underline;">htt=
p://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a><o:p></o:p>=
</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;"><a href=3D"http://www.ietf.org/mail-archive/web/tram/curr=
ent/msg00273.html" style=3D"color: purple; text-decoration: underline;">htt=
p://www.ietf.org/mail-archive/web/tram/current/msg00273.html</a><o:p></o:p>=
</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Sorry about that. I did not notice any clear items to answer or commen=
t on.&nbsp;</div>
<br>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">whether step D) can obsolete step C) (DISCUSS/MALICE) by =
allowing the application/browser to transfer relevant real-time traffic inf=
ormation (types and bandwidth) into
 every RTP package by using the already IETF standardized RTP extension hea=
der RCF 5285, which could be used by any current or future Internet (includ=
ing OTT) network where WebRTC real-time traffic may flow. It seems to have =
been standardized already 2008 to
 allow such usage.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
</div>
</div>
</blockquote>
<div>draft-martinsen-tram-discuss is intended to solve internal application=
 stream prioritisation pr flow, i.e. it is more important for me to get the=
 video stream across than my application stream at this moment in time. Its=
 intention is to describe the 5-topple
 the STUN message was sent on.&nbsp;</div>
<div><br>
</div>
<div>Moving to a pr packet prioritisation is interesting as it can solve th=
e bundle problem, and mark the relative importance of specific video frames=
 (I-Frames for example). Since its pr packet the network element might not =
keep stream state so it might be
 easier that way. But the information needs to be in a fixed offset for qui=
ck network element processing. And we want to describe other streams than j=
ust RTP. That makes RTP header extensions unsuitable.&nbsp;</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">QoS related step C)<span class=3D"Apple-converted-space">=
&nbsp;</span></span><span lang=3D"EN-US">draft-martinsen-tram-discuss<span =
class=3D"Apple-converted-space">&nbsp;</span></span><span lang=3D"EN-US" st=
yle=3D"font-size: 10pt; font-family: Arial, sans-serif; color: blue;">can
 then be taken out of TRAM, and step D) (that was never intended to be hand=
led by TRAM anyway) will be handled elsewhere.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>I was hoping to argue that the draft is only new signalling of old con=
cepts (DSCP and ECN). In the end it is the WG that decides if there is enou=
gh interest.&nbsp;</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">I have repeated my suggestion to RTCWEB WG from the Octob=
er discussions on the relevant<span class=3D"Apple-converted-space">&nbsp;<=
/span></span><span lang=3D"EN-US">[rtcweb] [avtext]
 [mmusic]</span><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family:=
 Arial, sans-serif; color: blue;"><span class=3D"Apple-converted-space">&nb=
sp;</span>lists<span class=3D"Apple-converted-space">&nbsp;</span><a href=
=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html" styl=
e=3D"color: purple; text-decoration: underline;">http://www.ietf.org/mail-a=
rchive/web/rtcweb/current/msg09129.html</a><span class=3D"Apple-converted-s=
pace">&nbsp;</span>to
 introduce the QoS related step D) into<span class=3D"Apple-converted-space=
">&nbsp;</span></span><span lang=3D"EN-US">draft-ietf-rtcweb-rtp-usage s</s=
pan><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans=
-serif; color: blue;">uggesting usage of RFC
 5285, where traffic information in RTP packets simply would be filled by a=
ny browser under any OS.</span><span lang=3D"EN-US"><o:p></o:p></span></div=
>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Without step B),</span><span lang=3D"EN-US" style=3D"font=
-size: 10pt; font-family: Arial, sans-serif; color: blue;"><span class=3D"A=
pple-converted-space">&nbsp;</span>TRAM</span><span lang=3D"EN-US" style=3D=
"font-size: 10pt; font-family: Arial, sans-serif; color: blue;"><span class=
=3D"Apple-converted-space">&nbsp;</span>=93</span><span lang=3D"EN-US" styl=
e=3D"font-size: 10pt; font-family: Arial, sans-serif; color: blue;">Enforci=
ng
 the real-time traffic through the offered/discovered TURN servers=94, the =
real-time traffic may happen through the IP default gateways often congeste=
d by data traffic and QoS insensitive streaming video and file sharing. Wit=
hout specific QoS methods at those
 points, the network raw bandwidth capacity may have to be 10-folded to ach=
ieve sufficient QoE when WebRTC usage becomes popular. Methods like DISCUSS=
 addresses such things occurring when STUN instead TURN is used (by allowin=
g flows into such congestion points),
 but only provides direct traffic info for outgoing (not incoming) real-tim=
e traffic and locally, therefore are not even applicable to reservation typ=
e of networks like Cable Networks and Mobile OTT.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>If both clients supports draft-martinsen-tram-discuss you would get an=
 bi-directional end to end description of the flow that works through NAT.<=
/div>
<div>The information in the STUN packet can be used to remark DSCP bits to =
whatever values that is of best use for the network the packet is flowing t=
hough.</div>
<div><br>
</div>
<div>I dont see how TURN gives that ability. Once the channel is bound you =
only have a channel number in front of the UDP payload. And TURN is designe=
d to carry alls sort of traffic. For this to work there must be some sort o=
f only webRTC traffic is allowed
 through this TURN server. (And Ill bet bitorrent or others find a way to m=
imic webRTC to set up only a data channel..)</div>
<br>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">That would leave certain network types to investing in ra=
w bandwidth increase, instead of simply borrowing the bandwidth (from much =
larger data traffic and QoS insensitive
 streaming video and file sharing at no adverse effect) and making that bor=
rowed no-cost bandwidth very valuable for the carriers when delivering to t=
heir customers.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;"><o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">I know a few of you Cisco guys are among the ones that ha=
ve been fighting against the general =93it is all about bandwidth=94 and =
=93it will go away with time=94-attitude within
 IETF work, e.g. see<span class=3D"Apple-converted-space">&nbsp;</span><a h=
ref=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html" s=
tyle=3D"color: purple; text-decoration: underline;">http://www.ietf.org/mai=
l-archive/web/rtcweb/current/msg09031.html</a>,
 but the pointers given in the current QoS discussion within TRAM:<o:p></o:=
p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(i) will *<b>not allow</b>* ISP=92s to use already availa=
ble and currently deployable quality IP pipes for real-time traffic to also=
 be used for WebRTC generated real-time
 traffic.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(ii) will *<b>not allow</b>* some network types (e.g. Cab=
le Networks and Mobile OTT) to borrow bandwidth from data traffic and QoS i=
nsensitive streaming video and file
 sharing. All networks will *<b>also be inhibited</b>* from the very common=
 and used method of simply providing an extra IP pipe (often provided over =
the same wire but level-2 separated) dedicated for real-time usage<o:p></o:=
p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(<b>*by resisting TRAM implementation of step B</b>*)<o:p=
></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Newer, fiber only type of networks, can still borrow band=
width at no extra cost, *<b>by proprietary usage</b>* of RFC 5285 by browse=
rs.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(iii) will leave most ISPs to *<b>only use</b>* raw bandw=
idth capacity increase that may has to be 10-folded to reach sufficient QoE=
 when WebRTC usage becomes popular (if
 at all possible, since unmanaged IP pipes intermittently are filled, whate=
ver bandwidth is available).<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Further explainations about QoS methods and their impleme=
ntation were given a few days ago in:<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;"><a href=3D"http://www.ietf.org/mail-archive/web/tram/curr=
ent/msg00274.html" style=3D"color: purple; text-decoration: underline;">htt=
p://www.ietf.org/mail-archive/web/tram/current/msg00274.html</a><o:p></o:p>=
</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(i), (ii) and (iii) will *<b>seriously delay</b>* usage o=
f telepresence capable and quality demanding WebRTC (bandwidth being around=
 30 times higher over best effort Internet,
 than telephony over quality managed networks) and will *<b>vastly increase=
 cost</b>* for ISPs to offer WebRTC with good QoE.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Below you are giving pointers saying =93</span><span lang=
=3D"EN-US">They are currently working on a problem-statement draft and a us=
e-case draft, any input to those would be
 very helpful.</span><span lang=3D"EN-US" style=3D"font-size: 10pt; font-fa=
mily: Arial, sans-serif; color: blue;">=94 Those are unnecessary, risking l=
eading to (i), (ii) and (iii) above.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
</div>
</div>
</blockquote>
<div>We find them quite useful. It allows us to engage with potential users=
 and see if there are real problems to solve. If the customer/users do not =
see a problem there is little incentive for us to invest in creating a solu=
tion that nobody wants. That said,
 it is always the danger of crating a to complicated solution that solves t=
o many problems. &nbsp;</div>
<br>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">So are pointers such as: =93</span><span lang=3D"EN-US">R=
MCAT where congestion avoidance for RTP is being developed</span><span lang=
=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-serif; color:=
 blue;">=94.
 TRAMs Milestone 3 is for the purpose of directing real-time traffic where =
congestion control isn=92t already in place. Using</span><span lang=3D"EN-U=
S" style=3D"font-size: 10pt; font-family: Arial, sans-serif; color: blue;">=
RFC 5285 available since 2008, to fill
 traffic information in RTP packets (step D)) is probably a necessity for m=
ost future =93congestion control=94 discussed.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
</div>
</div>
</blockquote>
<div>There are some good example of RTP header extension usage. For example=
&nbsp;draft-avtext-berger-framemarking. But that information may very well =
be encrypted and unavailable to the network element.</div>
<div><br>
</div>
<div><br>
</div>
.-.</div>
<div>P=E5l-Erik</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">This chicken-and-egg circular also applies to RTCWEB dire=
ction of QoS issues to:<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;"><a href=3D"http://datatracker.ietf.org/doc/draft-dhesikan=
-tsvwg-rtcweb-qos" style=3D"color: purple; text-decoration: underline;">htt=
p://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos</a><o:p></o:p>=
</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">focusing on DSCP-mapping without even mentioning RFC 5285=
 available since 2008, to fill traffic information in RTP packets where it =
&nbsp;is a necessity and will allow:<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(1) applications to directly convey QoS related real-time=
 traffic info to the network at points where RTP flow is directed to by TRA=
M Milstone 3, to be used by *<b>any
 network element implementing any suitable QoS methods for the particular n=
etwork</b>* for<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">(2) *<b>all</b>* WebRTC browsers *<b>and</b>* dedicated c=
lients (not using WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current=
 and future IP networks<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<b><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-=
serif; color: blue;">(3) *without*<span class=3D"Apple-converted-space">&nb=
sp;</span></span></b><span lang=3D"EN-US" style=3D"font-size: 10pt; font-fa=
mily: Arial, sans-serif; color: blue;">having to
 be forced into application specific networks (PSTN, IMS) instead of using =
the Internet (including OTT).<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">The only activity required, is to call for ISPs=92 review=
 whether the traffic information transferred by RFC 5285 is sufficient for =
current and future needs in their network
 as suggested in<span class=3D"Apple-converted-space">&nbsp;</span><a href=
=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html" styl=
e=3D"color: purple; text-decoration: underline;">http://www.ietf.org/mail-a=
rchive/web/rtcweb/current/msg09129.html</a><o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">&lt;snip&gt;<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">=85two parameters (e.g. two bytes each) are encoded into the RTP heade=
r extension:<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">A) The maximum bandwidth requirement: Two bytes could contain everythi=
ng from some bps for real-time text to Gbps for future 3D supersize telepre=
sence=85 on a logarithmic scale.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">B) The quality characteristics for the stream, with the highest bit se=
t to 1, we could allocate a bit each for quality type e.g:<o:p></o:p></span=
></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay Ins=
ensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, Prioriti=
ze X, Variation Y, that could be combined
 as required to describe the stream.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">And with highest bit set to 0, there could instead be a number for spe=
cial usage that does not fit the general description of the individual bits=
.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if;">&lt;/snip&gt;<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Then this could be assigned numbers to have the RFC in pl=
ace.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">With TRAM milestone 3 also place,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">market forces will drive ISPs and browser makers to imple=
ment just that, without even having it MUST-established.<o:p></o:p></span><=
/div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">=93Who does not want a =93WebRTC-Ready=94 Internet access=
?=94 and<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">=93Who wants to use Chrome, if Firefox, Internet Explorer=
 or Safari comes with much better QoE?=94 and vice versa.<o:p></o:p></span>=
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span>l</div>
</div>
</div>
</blockquote>
</div>
<div>
<blockquote type=3D"cite">
<div lang=3D"SV" link=3D"blue" vlink=3D"purple" style=3D"font-family: Helve=
tica; font-size: 12px; font-style: normal; font-variant: normal; font-weigh=
t: 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;">
<div class=3D"WordSection1" style=3D"page: WordSection1;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">Please see further emails soon following this one, for de=
tails and history.<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">/Karl<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: blue;">&nbsp;</span></div>
<div>
<div style=3D"border-style: solid none none; border-top-color: rgb(181, 196=
, 223); border-top-width: 1pt; padding: 3pt 0cm 0cm;">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif;">Fr=E5n=
:</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif=
;"><span class=3D"Apple-converted-space">&nbsp;</span>tram [<a href=3D"mail=
to:tram-bounces@ietf.org" style=3D"color: purple; text-decoration: underlin=
e;">mailto:tram-bounces@ietf.org</a>]<span class=3D"Apple-converted-space">=
&nbsp;</span><b>F=F6r<span class=3D"Apple-converted-space">&nbsp;</span></b=
>Pal
 Martinsen (palmarti)<br>
<b>Skickat:</b><span class=3D"Apple-converted-space">&nbsp;</span>den 21 fe=
bruari 2014 10:37<br>
<b>Till:</b><span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"m=
ailto:tram@ietf.org">tram@ietf.org</a><br>
<b>Kopia:</b><span class=3D"Apple-converted-space">&nbsp;</span>Karl Stahl;=
 Oleg Moskalenko; Alan Johnston; Yoakum, John H (John)<br>
<b>=C4mne:</b><span class=3D"Apple-converted-space">&nbsp;</span>Re: [tram]=
 I-D Action: draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></di=
v>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
Hi,<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
I agree the full QoS discussion should _not_ happen in TRAM. If you are int=
erested in helping out in that area I suggest you looking into the AEON mai=
ling list at:&nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/aeon" s=
tyle=3D"color: purple; text-decoration: underline;">https://www.ietf.org/ma=
ilman/listinfo/aeon</a><span class=3D"Apple-converted-space">&nbsp;</span>.
 They are currently working on a problem-statement draft and a use-case dra=
ft, any input to those would be very helpful. (<a href=3D"http://tools.ietf=
.org/html/draft-eckel-aeon-use-cases-01" style=3D"color: purple; text-decor=
ation: underline;">http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01=
</a>,&nbsp;<a href=3D"http://tools.ietf.org/html/draft-eckel-aeon-problem-s=
tatement-00" style=3D"color: purple; text-decoration: underline;">http://to=
ols.ietf.org/html/draft-eckel-aeon-problem-statement-00</a>).<o:p></o:p></d=
iv>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
That said, STUN have a few nice characteristics that makes it a perfect can=
didate for transporting some of the QoS information. &nbsp;IMHO that would =
be extending the STUN spec and should be within the TRAM charter. &nbsp;The=
 main goal of&nbsp;draft-martinsen-tram-discuss
 was to show how already existing QoS mechanisms could be transported with =
STUN to provide more value, and to start the discussion if TRAM is the appr=
opriate place to have those on the wire format discussions.<o:p></o:p></div=
>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
.-.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
P=E5l-Erik<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<o:p>&nbsp;</o:p></div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) &lt;<a href=3D"mailto:yo=
akum@avaya.com" style=3D"color: purple; text-decoration: underline;">yoakum=
@avaya.com</a>&gt; wrote:<o:p></o:p></div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<br>
<br>
<o:p></o:p></div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">&#43;1</span><span lang=3D"EN-US"><o:p></o:=
p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">I fully agree with the comments that QOS sh=
ould be a low priority for the initial focus of the TRAM efforts.&nbsp; The=
re are other groups doing QOS work and frankly
 I engage in WebRTC multimedia interactions daily over the Internet, enterp=
rise VPNs, and various combinations and seldom suffer egregious quality iss=
ues.&nbsp; I am more concerned about carriers doing things to regulate or d=
egrade WebRTC flows than a failure of
 existing Internet mechanisms to enable them.</span><span lang=3D"EN-US"><o=
:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">&nbsp;</span><span lang=3D"EN-US"><o:p></o:=
p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">Significant focus on QOS before we better e=
nable TURN to be easily used in a browser environment taking advantage or n=
ormal web characteristics (as opposed
 to historic telephony constructs) would seem to be highly distracting at t=
his point.</span><span lang=3D"EN-US"><o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">&nbsp;</span><span lang=3D"EN-US"><o:p></o:=
p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">&nbsp;</span><span lang=3D"EN-US"><o:p></o:=
p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-ser=
if; color: navy;">Cheers,</span><span lang=3D"EN-US"><o:p></o:p></span></di=
v>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<i><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Arial, sans-=
serif; color: navy;">John</span></i><span lang=3D"EN-US"><o:p></o:p></span>=
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 8pt; font-family: Arial, sans-seri=
f; color: navy;">&nbsp;</span><span lang=3D"EN-US"><o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 8pt; font-family: Arial, sans-seri=
f; color: red;">AVAYA</span><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);"><br>
</span><span lang=3D"EN-US" style=3D"font-size: 7.5pt; font-family: Arial, =
sans-serif; color: teal;">1.919.425.8446</span><span lang=3D"EN-US"><o:p></=
o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, sans-s=
erif; color: rgb(31, 73, 125);">&nbsp;</span><span lang=3D"EN-US"><o:p></o:=
p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<b><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Tahoma, sans=
-serif;">From:</span></b><span class=3D"apple-converted-space"><span lang=
=3D"EN-US" style=3D"font-size: 10pt; font-family: Tahoma, sans-serif;">&nbs=
p;</span></span><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family:=
 Tahoma, sans-serif;">tram
 [<a href=3D"mailto:tram-bounces@ietf.org" style=3D"color: purple; text-dec=
oration: underline;">mailto:tram-bounces@ietf.org</a>]<span class=3D"apple-=
converted-space">&nbsp;</span><b>On Behalf Of<span class=3D"apple-converted=
-space">&nbsp;</span></b>Oleg Moskalenko<br>
<b>Sent:</b><span class=3D"apple-converted-space">&nbsp;</span>Thursday, Fe=
bruary 20, 2014 12:43 PM<br>
<b>To:</b><span class=3D"apple-converted-space">&nbsp;</span>Alan Johnston<=
br>
<b>Cc:</b><span class=3D"apple-converted-space">&nbsp;</span>Karl Stahl;<sp=
an class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailto:tram@ietf=
.org" style=3D"color: purple; text-decoration: underline;">tram@ietf.org</a=
><br>
<b>Subject:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [tram=
] Fwd: I-D Action: draft-thomson-tram-turn-bandwidth-00.txt</span><span lan=
g=3D"EN-US"><o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font=
-family: 'Times New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston &lt;<a =
href=3D"mailto:alan.b.johnston@gmail.com" target=3D"_blank" style=3D"color:=
 purple; text-decoration: underline;"><span style=3D"color: purple;">alan.b=
.johnston@gmail.com</span></a>&gt; wrote:<o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">Personally, I am not sure how much QoS is actually in =
scope for TRAM. Have you been following RMCAT where congestion avoidance fo=
r RTP is being developed? &nbsp;I see some overlap in your goals and the go=
als of that work.<o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span class=3D"hoenzb"><span lang=3D"EN-US" style=3D"color: rgb(136, 136, 1=
36);">-</span></span><span lang=3D"EN-US"><o:p></o:p></span></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"color: rgb(136, 136, 136);">&nbsp;</span><spa=
n lang=3D"EN-US"><o:p></o:p></span></div>
</div>
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font=
-family: 'Times New Roman', serif;">
<span lang=3D"EN-US">I'd concentrate on the TURN application-level function=
ality, for now, and I'd leave QoS for the future discussions.<br>
<br>
<br>
<o:p></o:p></span></p>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif;">
<span lang=3D"EN-US" style=3D"font-size: 9pt; font-family: Helvetica, sans-=
serif;">_______________________________________________<br>
tram mailing list<br>
<a href=3D"mailto:tram@ietf.org" style=3D"color: purple; text-decoration: u=
nderline;">tram@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tram" style=3D"color: purp=
le; text-decoration: underline;">https://www.ietf.org/mailman/listinfo/tram=
</a></span></div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</body>
</html>

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From: Barry Dingle <btdingle@gmail.com>
Date: Tue, 25 Feb 2014 23:55:15 +1100
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On Tue, Feb 25, 2014 at 9:58 PM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-02-23 10:54, Barry Dingle wrote:
> > In reference to your first _Section 1 comments_, I agree that we need t=
o
> > clearly differentiate the SRTP media and the SCTP media. (Note - SRTP -
> > not RTP !)
>
> So SRTP is an extension of RTP, in fact it is a profile, or rather two
> (RTP/SAVP or RTP/SAVPF). I wrote RTP because I meant no specific RTP
> configuration here, it can be any of the profiles and any of the
> extensions. I know and fully support that RTCWEB uses RTP/SAVPF for RTP
> transported media, however in the wider context and possible other usage
> of the defined mechanism this may or may not be true. Therefore I see a
> need to be generic in that discussion. If one want to emphasize the SRTP
> part, then use (S)RTP.
>
> I see your point about being generic - and it is consistent with
draft-ietf-rtcweb-rtp-usage.


 >
> > I agree that there could be real-time media that can be sent over SCTP
> > Channels are still meet the criteria for 'real-time' as being 'received
> > within a few hundred milliseconds of being generated' - see WebRTC
> > Overview section 2.4.
> >
> > Maybe we need to differentiate the Channel types - and then refer to th=
e
> > data going over those Channels. For instance,if we define "SRTP
> > Channels" and "SCTP Channels", we could then refer to 'SRTP Channel
> > media' which would have a specific meaning.
>
> Well, around RTP we aren't discussing "channels" at all. Thus, this
> becomes very ambiguous to what it means, unless you have a definition to
> go along with it.
>
> In the context of a PeerConnection we can have one or more RTP session.
>
> Also, I don't know what an SCTP channel is. There is an SCTP
> association, which is something between two endpoints. On top of this we
> have the SCTP Data channels. So what are you missing or want from these
> definitions?
>
> I was trying to find a way to identify
(a) 'all the SCTP Streams' (or maybe "all the SCTP Channels") as well as
(b) "all the (S)RTP Streams".

As you suggested, the term "SCTP Association" seems to work for (a). Does
"RTP Session" work for (b) ??

Cheers,
/Barry Dingle

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Feb 25, 2014 at 9:58 PM, Magnus Westerlund <span dir=3D"ltr">&l=
t;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnu=
s.westerlund@ericsson.com</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">On 2014-02-23 10:54, Barr=
y Dingle wrote:<br>
&gt; In reference to your first _Section 1 comments_, I agree that we need =
to<br>
<div>&gt; clearly differentiate the SRTP media and the SCTP media. (Note - =
SRTP -<br>
&gt; not RTP !)<br>
<br>
</div>So SRTP is an extension of RTP, in fact it is a profile, or rather tw=
o<br>
(RTP/SAVP or RTP/SAVPF). I wrote RTP because I meant no specific RTP<br>
configuration here, it can be any of the profiles and any of the<br>
extensions. I know and fully support that RTCWEB uses RTP/SAVPF for RTP<br>
transported media, however in the wider context and possible other usage<br=
>
of the defined mechanism this may or may not be true. Therefore I see a<br>
need to be generic in that discussion. If one want to emphasize the SRTP<br=
>
part, then use (S)RTP.<br>
<div><br></div></blockquote><div>I see your point about being generic - and=
 it is consistent with draft-ietf-rtcweb-rtp-usage. <br><br><br></div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:=
1px solid rgb(204,204,204);padding-left:1ex">


<div>
&gt;<br>
&gt; I agree that there could be real-time media that can be sent over SCTP=
<br>
&gt; Channels are still meet the criteria for &#39;real-time&#39; as being =
&#39;received<br>
&gt; within a few hundred milliseconds of being generated&#39; - see WebRTC=
<br>
&gt; Overview section 2.4.<br>
&gt;<br>
&gt; Maybe we need to differentiate the Channel types - and then refer to t=
he<br>
&gt; data going over those Channels. For instance,if we define &quot;SRTP<b=
r>
&gt; Channels&quot; and &quot;SCTP Channels&quot;, we could then refer to &=
#39;SRTP Channel<br>
&gt; media&#39; which would have a specific meaning.<br>
<br>
</div>Well, around RTP we aren&#39;t discussing &quot;channels&quot; at all=
. Thus, this<br>
becomes very ambiguous to what it means, unless you have a definition to<br=
>
go along with it.<br>
<br>
In the context of a PeerConnection we can have one or more RTP session.<br>
<br>
Also, I don&#39;t know what an SCTP channel is. There is an SCTP<br>
association, which is something between two endpoints. On top of this we<br=
>
have the SCTP Data channels. So what are you missing or want from these<br>
definitions?<br>
<div><div><br></div></div></blockquote><div>I was trying to find a way to i=
dentify <br>(a) &#39;all the SCTP Streams&#39; (or maybe &quot;all the SCTP=
 Channels&quot;) as well as<br>(b) &quot;all the (S)RTP Streams&quot;. <br>


<br></div><div>As you suggested, the term &quot;SCTP Association&quot; seem=
s to work for (a). Does &quot;RTP Session&quot; work for (b) ??<br><br></di=
v><div>Cheers,<br></div><div>/Barry Dingle <br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287" target=
=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079" ta=
rget=3D"_blank">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

--089e0102e6daae89c704f33a9a98--


From nobody Tue Feb 25 06:17:20 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: "saag@ietf.org" <saag@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [irsg] Final agenda for the IRTF Open Meeting at IETF-89
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Date: Tue, 25 Feb 2014 14:17:11 +0000
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The two ANRP award talks may be of interest to folks on this list.

Begin forwarded message:

> From: "Eggert, Lars" <lars@netapp.com>
> Subject: [irsg] Final agenda for the IRTF Open Meeting at IETF-89
> Date: February 25, 2014 at 14:59:14 GMT+1
> To: "irtf-discuss@irtf.org" <irtf-discuss@irtf.org>, =
"irtf-announce@irtf.org" <irtf-announce@irtf.org>
> Reply-To: "irtf-discuss@irtf.org" <irtf-discuss@irtf.org>
>=20
> The preliminary agenda for the IRTF Open Meeting at IETF-88 is now =
online: http://www.ietf.org/proceedings/89/agenda/agenda-89-irtfopen
>=20
> Lars
>=20
> --
>=20
> Preliminary Agenda
> IRTF Open Meeting @ IETF-89
> London, UK
> WEDNESDAY, March 5, 2014
> 0900-1130  Morning Session I
>=20
> [Slot lengths below indicate presentation+discussion time.]
>=20
> State of the IRTF
>    Lars Eggert
>    5+5 min
>=20
> Applied Networking Prize (ANRP) Award Talks
>    30+10 min (x2)
>=20
>    *** Kenny Paterson *** on finding and documenting new attacks
>    against TLS and DTLS:
>=20
>       N. J. Al Fardan and K. G. Paterson. Lucky Thirteen: Breaking
>       the TLS and DTLS Record Protocols. Proc. IEEE Symposium on
>       Security and Privacy, pp. 526-540, San Francisco, CA, USA,
>       May 2013.
>=20
>    *** Keith Winstein *** on designing a transport protocol for
>    interactive applications that desire high throughput and low delay:
>=20
>       Keith Winstein, Anirudh Sivaraman, and Hari Balakrishnan
>       Stochastic Forecasts Achieve High Throughput and Low Delay
>       over Cellular Networks. Proc. 10th USENIX Symposium on
>       Networked Systems Design and Implementation (NSDI), Lombard,
>       IL, USA, April 2013.
>=20
>       Keith will talk about our work building Sprout, a transport
>       protocol for interactive traffic over cellular networks, and =
Remy, a
>       tool that generates congestion-control protocols automatically.
>=20
>=20


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From nobody Tue Feb 25 07:52:31 2014
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Karl Stahl <karl.stahl@intertex.se>
Thread-Topic: [tram] [rtcweb]  Payload Types assignments
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Date: Tue, 25 Feb 2014 15:52:19 +0000
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com> <5238446D.8050700@ericsson.com> <9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net> <07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se> <07e401ceb713$bef87a60$3ce96f20$@stahl@intertex.se> <524AB730.7040809@ericsson.com> <00b001cebfc1$ba8e89e0$2fab9da0$@stahl@intertex.se> <525272E8.5050300@ericsson.com> <050801cec3f6$6172aec0$24580c40$@stahl@intertex.se> <5253E5EB.8030608@alvestrand.no> <02bf01cecf34$35e174a0$a1a45de0$@stahl@intertex.se> <04dd01cf31ad$0fe62d00$2fb28700$@stahl@intertex.se> <580B467D-4679-4DE1-96DE-CA37DE755563@csperkins.org> <052e01cf31cb$5311a0a0$f934e1e0$@stahl@intertex.se> <530C56CD.3010003@ericsson.com>
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It may have been mentioned previously, but those interested in this may
well want to have a look at the following drafts being discussed on the
AEON list (aeon@ietf.org):

http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00
http://tools.ietf.org/html/draft-eckel-aeon-use-cases-00

While QoS is not the sole focus of this work, it is one of the important
work flows prevalent in networks today that is more challenging in the
face of emerging communication applications such as those enabled by
WebRTC. One of the goals of the AEON effort is to provide a framework such
as the one mentioned by Magnus.
Comments on the AEON list are welcome, as is your attendance at the AEON
side meeting at IETF 89:
http://www.ietf.org/mail-archive/web/aeon/current/msg00018.html

Cheers,
Charles


On 2/25/14, 12:39 AM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>Karl,
>(As individual)
>
>I also wish for more usable QoS mechanisms. However, I don't see that
>being achieved by your proposal. In addition I agree with Colin about
>the issues with putting the information in the RTP header extension. I
>would also note that this would be very RTP specific, and not at all
>help with the data channels multiple streams and their priorities. There
>might be data channel information that is more crucial than any of the
>RTP media stream packets.
>
>You are pushing for a small piece in the middle. A piece that will not
>help with the more general issue of QoS. How does the application and
>the multiple ISPs that carries the traffic reach an agreement on what
>properties that can be provided, that the application in this instance
>have the right to request those properties and that any cost is
>correctly associated with the user or the user's agent in regards to
>carrying the associated cost.
>
>When it comes to setting DSCP from user land in the OS, that is
>restricted due to the security implications. If those implications where
>resolved, then OS could open up those interfaces.  There are many
>interlinked reasons why things look like they do today. I don't believe
>in tugging on a single random thread in ball of yarn and hope that it
>comes out without any knots and ties on it.
>
>Show me the framework for the QoS functions you have in mind that at
>least has less issues than the currently deployed and take care of at
>least some of the bigger issues. If that requires information in the RTP
>streams for some reasons, then let us talk about how to best encoded it.
>But, we need that framework first, the architecture that makes this a
>better solution than the current Diffserv architecture or any other QoS
>architecture.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Services, Media and Network features, Ericsson Research EAB/TXM
>----------------------------------------------------------------------
>Ericsson AB                 | Phone  +46 10 7148287
>F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>tram mailing list
>tram@ietf.org
>https://www.ietf.org/mailman/listinfo/tram


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Hi Crister,

Thanks.
Please see my comments inline.

> Hi Raju,
>=20
> >> Anyway, can you give me an example of a case where you want to use
> >> SDP, and where you need to negotiate the re-transmission values etc?
> >
> > [Raju] A simple example would be 2 browsers talking thru an intermedia
> proxy, which wants to know what protocols are
> > being negitiated and used as part of the session.
>=20
> Sure. I don't question the use-case/need for using SDP to negotiate chann=
el
> usages.
>=20
> > 1. Calling client creates data channel (using
> http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel)
> with desired
> > attributes and sets negotiated=3Dtrue. So, calling browser saves the
> attributes for the data channel but won't use DCP.
> > 2. These attributes are sent via SDP to peer client (via proxy).
> > 3. Peer client sets these same attributes while creating data channel a=
nd
> sets negotiated=3Dtrue but won't do DCP.
> > 4. Peer client echo the same attributes back in SDP answer.
> >
> > Now, data channel stacks on both ends will use same attributes, similar=
 to
> DCP use case, with the only difference being DCP is not used here.
>=20
> My question was regarding the re-transmission values etc.
>=20
> Why do the peers need to negotiate those? Why would an intermediary need =
to
> know about those? Why not simply define those in the protocol description=
,
> and each endpoint supporting the protocol will then know what values to u=
se?

[Raju] Peers need to negotiate those so that both ends of the data channel =
can use same timers (similar to DCP).
Intermediate entities pass these values to the peer in some cases and need =
to know these in other cases so that they can be passed down to data channe=
l entity which is terminating the data channel (and interworking to other t=
ransports like TCP).
IMHO, these attributes are specific to data channel thus they should remain=
 associated to data channel (DCP or external negotiation; making them part =
of other protocols is not only wrong but also causes unenecessary modificat=
ions adding overhead to these protocols.

Best Regards
Raju


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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Christer - comment at end

On 2/25/14 4:08 AM, Christer Holmberg wrote:
> Hi Raju,
>
>>> Anyway, can you give me an example of a case where you want to use
>>> SDP, and where you need to negotiate the re-transmission values etc?
>>
>> [Raju] A simple example would be 2 browsers talking thru an intermedia proxy, which wants to know what protocols are
>> being negitiated and used as part of the session.
>
> Sure. I don't question the use-case/need for using SDP to negotiate channel usages.
>
>> 1. Calling client creates data channel (using http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel) with desired
>> attributes and sets negotiated=true. So, calling browser saves the attributes for the data channel but won't use DCP.
>> 2. These attributes are sent via SDP to peer client (via proxy).
>> 3. Peer client sets these same attributes while creating data channel and sets negotiated=true but won't do DCP.
>> 4. Peer client echo the same attributes back in SDP answer.
>>
>> Now, data channel stacks on both ends will use same attributes, similar to DCP use case, with the only difference being DCP is not used here.
>
> My question was regarding the re-transmission values etc.
>
> Why do the peers need to negotiate those? Why would an intermediary need to know about those? Why not simply define those in the protocol description, and each endpoint supporting the protocol will then know what values to use?

IIUC, the webrtc people are "humoring* us by making provision for 
registered subprotocol names, and using SDP to negotiate channels. I 
suspect that for the most part they have no intention of using those. 
(The subprotocol name is *optional*.)

So these parameters are needed to properly initialize a channel when the 
subprotocol is not specified.

Also, we still have an open question about what attributes must be 
present in a subprotocol specification. It is at least possible that 
some subprotocols may work with a variety of settings of these attributes.

This draft should probably say more about this. It could say that:

* the SDP MUST be consistent with the attributes specified in the 
subprotocol definition, OR

* the SDP MUST NOT specify attributes specified in the subprotocol 
definition, OR

* attributes specified in the SDP override those specified in the 
subprotocol definition.

	Thanks,
	Paul


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Inline

On 2/24/14 5:12 AM, Michael Tuexen wrote:
> On Feb 23, 2014, at 4:07 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> A few comments on this draft:
>>
>> * Section 5.1:
>>
>>       DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
>>          a partial reliable in-order bi-directional Communication
>>          channel.  User messages might not be transmitted or
>>          retransmitted after a specified life-time given in milli-
>>          seconds in the Reliability Parameter.  This life-time starts
>>          when providing the user message to the Javascript engine.
>>
>> The last sentence above is troubling. This protocol won't always be used via Javascript. Can we please have a definition that
> OK. What about replacing the last sentence with
>
> This life-time starts when providing the user
> message to the protocol stack.

*which* protocol stack?

This is part of the problem. As the documents are currently structured, 
there is a "data channel establishment protocol", but there is no "data 
channel protocol".

But in reality there *is* a (thin) data channel protocol. It just isn't 
described explicitly. In a browser it is perhaps just the code behind 
the data channel API. But in other environments it may need to be more 
explicit. It has some obligations. Presumably enforcing this "lifetime" 
parameter is one of them. Enforcing the use of sequenced messages 
between a DCEP open and receipt of ACK is another.

>> doesn't depend on that? Is the timing being done by SCTP, or are you assuming that a data channel layer on top of SCTP is doing this? Is it specific to SCEP, or is it still applicable when the channel has been established via external negotiation?
> It is actually the time when the message is provided to the SCTP stack, but I think
> there might be some code running between the SCTP code and the user code (let it be
> JS or anything else). I'm assuming that there is no substantial buffering happening
> there.
>>
>> Does use of this option imply that retransmission continues until this time limit is reached? Or might it stop after some implementation defined number of retransmissions?
> The only way the retransmissions don't happen until the time limit is reached, is that
> SCTP decides that the association is broken because of excessive retransmissions.
>>
>> The description of the reliability parameter says:
>>
>>       This field is ignored if a reliable channel is used.
>>
>> IMO folk wisdom says that some specific value (e.g., 0) should be prescribed to be used in such cases. That makes things more deterministic and provides more flexibility in extending the protocol in the future.
> What about:
>
> For reliable channels this field MUST be set to 0 on the sending side
> and MUST be ignored no the receiving side.

WFM

>> The name "Protocol" (and "Protocol Length") here is troublesome, because it is ambiguous with other sorts of uses. (See my comment about this point earlier today wrt draft-ietf-rtcweb-data-channel-07.) Others (including draft-ejzak-mmusic-data-channel-sdpneg-00 and websockets) call this "subprotocol". Using that would make it a little less confusing.
> In the W3C specification it is also called protocol, but the text talks about sub-protocol. Maybe we should do
> the same here.

I'm not following the W3C part.

I'm just interested in getting consistency, and avoiding confusion.
Right now there is a lot of confusion around the use of "protocol".

>> Also, ISTM that all of these things are applicable to externally negotiated channels, and so ought to be defined by draft-ietf-rtcweb-data-channel. (But of course their use in the DATA_CHANNEL_OPEN message belongs here.)
> Reliability is discussed there, but I agree, we need to discuss label and protocol there, too.

Good!

>> * Section 8.4:
>>
>> ISTM there ought to be a template of the minimum information that the specification for a registered (sub)protocol MUST (SHOULD?) include. E.g.,
>>
>> - what Channel Type(s) are valid for this (sub)protocol
> I assumed all...

ISTM that in *most* cases a if a (sub)protocol is designed for use with 
a reliable channel it won't work correctly with an unreliable channel.

And while things designed to work with an unreliable channel will 
probably continue to work with a reliable channel, they might not have 
the intended properties.

That's why its good to have some guidelines about what the specification 
must contain. It prevents people providing something sloppy and 
incomplete that isn't sufficient to use the (sub)protocol.

>> - A contact for more information about this protocol
> You need to provide a reference for the protocol. Isn't that enough?

Depends on the form of the reference. Is the reference required to be 
publicly available? Stable?

Note the definition of FCFS policy in 5226:

       First Come First Served - Assignments are made to anyone on a
             first come, first served basis.  There is no substantive
             review of the request, other than to ensure that it is
             well-formed and doesn't duplicate an existing assignment.
             However, requests must include a minimal amount of clerical
             information, such as a point of contact (including an email
             address) and a brief description of how the value will be
             used.  Additional information specific to the type of value
             requested may also need to be provided, as defined by the
             namespace.  For numbers, the exact value is generally
             assigned by IANA; with names, specific text strings can
             usually be requested.

Having a sufficient definition of what must be in the registry, and what 
must be in a reference from the registry is *more* important for FCFS 
than it is for some of the more restrictive policies. The policies that 
get careful review can perhaps depend on the reviewers stopping 
something that is incomplete.

IMO the registry ought to provide, directly or indirectly, enough 
information to implement/use the (sub)protocol.

	Thanks,
	Paul

> Best regards
> Michael
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi Paul,

>> My question was regarding the re-transmission values etc.
>>
>> Why do the peers need to negotiate those? Why would an intermediary need=
 to know about=20
>> those? Why not simply define those in the protocol description, and each=
 endpoint supporting the >> protocol will then know what values to use?
>
> IIUC, the webrtc people are "humoring* us by making provision for registe=
red subprotocol names, > and using SDP to negotiate channels. I suspect tha=
t for the most part they have no intention of=20
> using those.=20
> (The subprotocol name is *optional*.)
>
> So these parameters are needed to properly initialize a channel when the =
subprotocol is not=20
> specified.

Why would someone NOT specify the subprotocol?

Especially considering Richard's draft, I thought the whole idea was to be =
able to negotiate the subprotocol using SDP.

> Also, we still have an open question about what attributes must be presen=
t in a subprotocol=20
> specification. It is at least possible that some subprotocols may work wi=
th a variety of settings of=20
> these attributes.

Well, in that case the subprotocol specification should define the addition=
al SDP attributes needed.

> This draft should probably say more about this. It could say that:
>
> * the SDP MUST be consistent with the attributes specified in the subprot=
ocol definition, OR
>
> * the SDP MUST NOT specify attributes specified in the subprotocol defini=
tion, OR
>
> * attributes specified in the SDP override those specified in the subprot=
ocol definition.

If people see a need for these attributes, fine. But, I still have not seen=
 an example where it would be needed (e.g. the UDP RFC does not specify the=
 timers for SIP - RFC 3261 does :)

Regards,

Christer


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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I agree with Haralds comments on this.

I do not see any point in writing RTCWEB specific requirements on this.

Keith=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
> Magnus Westerlund
> Sent: 25 February 2014 10:24
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Number of samples (ptime) to be=20
> supported by required codecs (draft-ietf-rtcweb-audio-05)
>=20
> On 2014-02-19 19:11, Harald Alvestrand wrote:
> > On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
> >> On 2014-02-18 23:58, Harald Alvestrand wrote:
> >>> I may be a little simple-minded, but if we have a recommendation=20
> >>> that receivers MUST be able to receive all packetizations=20
> of G.711=20
> >>> and OPUS up to 200 ms per packet, and that receivers=20
> should signal=20
> >>> this by sending "a=3Dmaxptime:200" in their SDP, what=20
> situations exist=20
> >>> where interoperability is not going to be achieved?
> >> Interoperability is going to be achieve in the direction=20
> towards this=20
> >> endpoint. The question is if we achieve interoperability in the=20
> >> direction from that endpoint.
> >=20
> > So, given that there is nothing in the G.711 specification=20
> about which=20
> > sample sizes a G.711 receiver has to support - the right approach=20
> > seems to be to amend the G.711 MIME registration with this=20
> information.
>=20
> Yes, it would for the future. But considering the wide-spread=20
> deployment of this payload format I don't see that having any=20
> short term effect on the interoperability.
>=20
> Also, the recommendations are actually context dependent.=20
> Thus, it is not obvious that a single recommendation is suitable.
>=20
> >=20
> > This is not a WebRTC issue; it is an absence of=20
> specification for the=20
> > non-WebRTC devices that use G.711. The RTCWEB specifications can=20
> > reasonably be expected to point to existing information about this=20
> > issue; it is not reasonable (in my mind) that the RTCWEB WG decide=20
> > what these values should be.
> >=20
>=20
> I would argue for this consideration based on that this is=20
> done in the WebRTC context, not any other context.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =


From nobody Tue Feb 25 09:55:09 2014
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi Paul,

Please see comments inline.

> Christer - comment at end
>=20
> On 2/25/14 4:08 AM, Christer Holmberg wrote:
> > Hi Raju,
> >
> >>> Anyway, can you give me an example of a case where you want to use
> >>> SDP, and where you need to negotiate the re-transmission values etc?
> >>
> >> [Raju] A simple example would be 2 browsers talking thru an intermedia
> proxy, which wants to know what protocols are
> >> being negitiated and used as part of the session.
> >
> > Sure. I don't question the use-case/need for using SDP to negotiate
> channel usages.
> >
> >> 1. Calling client creates data channel (using
> http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel)
> with desired
> >> attributes and sets negotiated=3Dtrue. So, calling browser saves the
> attributes for the data channel but won't use DCP.
> >> 2. These attributes are sent via SDP to peer client (via proxy).
> >> 3. Peer client sets these same attributes while creating data channel =
and
> sets negotiated=3Dtrue but won't do DCP.
> >> 4. Peer client echo the same attributes back in SDP answer.
> >>
> >> Now, data channel stacks on both ends will use same attributes, simila=
r
> to DCP use case, with the only difference being DCP is not used here.
> >
> > My question was regarding the re-transmission values etc.
> >
> > Why do the peers need to negotiate those? Why would an intermediary nee=
d
> to know about those? Why not simply define those in the protocol
> description, and each endpoint supporting the protocol will then know wha=
t
> values to use?
>=20
> IIUC, the webrtc people are "humoring* us by making provision for
> registered subprotocol names, and using SDP to negotiate channels. I
> suspect that for the most part they have no intention of using those.
> (The subprotocol name is *optional*.)
>=20
> So these parameters are needed to properly initialize a channel when the
> subprotocol is not specified.

[Raju] If I understand correctly, these parameters simply define the=20
data channel transport properties. Yes, they may have some correletion with=
=20
subprotocol, but that's not a one-to-one mapping.
For example, a given subprotocol could run on a reliable or=20
unreliable transport; depending on app needs it may use unreliable=20
transport and deal with reliablity at subprotocol stack (or ignore it).

>=20
> Also, we still have an open question about what attributes must be
> present in a subprotocol specification. It is at least possible that
> some subprotocols may work with a variety of settings of these attributes=
.

[Raju] Right, this fits into my explanation above.

>=20
> This draft should probably say more about this. It could say that:
>=20
> * the SDP MUST be consistent with the attributes specified in the
> subprotocol definition, OR
>=20
> * the SDP MUST NOT specify attributes specified in the subprotocol
> definition, OR
>=20
> * attributes specified in the SDP override those specified in the
> subprotocol definition.

[Raju] Good point. I like item 1 above; sure we can add it to this=20
draft. Then individual subprotocol drafts=20
(e.g. draft-ejzak-dispatch-msrp-data-channel-00) could get into the
details of what that consistency is.

Best Regards
Raju


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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi Christer,

Please find my comments inline.

>=20
> Especially considering Richard's draft, I thought the whole idea was to b=
e
> able to negotiate the subprotocol using SDP.
>=20
> > Also, we still have an open question about what attributes must be pres=
ent
> in a subprotocol
> > specification. It is at least possible that some subprotocols may work
> with a variety of settings of
> > these attributes.
>=20
> Well, in that case the subprotocol specification should define the
> additional SDP attributes needed.
>=20
> > This draft should probably say more about this. It could say that:
> >
> > * the SDP MUST be consistent with the attributes specified in the
> subprotocol definition, OR
> >
> > * the SDP MUST NOT specify attributes specified in the subprotocol
> definition, OR
> >
> > * attributes specified in the SDP override those specified in the
> subprotocol definition.
>=20
> If people see a need for these attributes, fine. But, I still have not se=
en
> an example where it would be needed (e.g. the UDP RFC does not specify th=
e
> timers for SIP - RFC 3261 does :)

[Raju] Aren't the attributes we are talking here are data channel propertie=
s?
DCP explains the following type of data channel as:
      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
         a partial reliable in-order bi-directional Communication
         channel.  User messages might not be transmitted or
         retransmitted after a specified life-time given in milli-
         seconds in the Reliability Parameter.  This life-time starts
         when providing the user message to the Javascript engine.

Fo example,'maxRetransmitTime' value specified by createDataChannel API (SD=
P's maxtime) here is realized by data channel stack, NOT by the application=
. The timer value given by the client may be a generic value rather than sp=
ecific to a subprotocol. To draw comparision to TCP, Linux allows TCP_USER_=
TIMEOUT sockopt which allows application to specify max timer value for any=
 unacknowledged data. 'maxRetransmitTime' is similar.
The SIP analogy mentioned above is for subprotocol level timers but not for=
 transport level retransmissions timers. A subprotocol could have it's own =
retransmission timer for lack of response for a req. Lack of subprotocol re=
sponse could still happen, if the peer application drops the request, even =
when data channel successfully sent the request.=20

So, I think timers (parameters) at both data channel stack and subprotocol =
have clear semantics and mostly independent of each other.

Best Regards
Raju


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Date: Tue, 25 Feb 2014 13:26:57 -0600
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Why are we having this discussion on the RTCWEB mailing list rather than
the MMUSIC mailing list?   This document was dispatched to MMUSIC at
IETF-88: http://trac.tools.ietf.org/wg/dispatch/trac/wiki
And, the nifty datatracker tool tells us that this (expired) document has
been replaced by an individual draft that has been submitted to the MMUSIC
WG:
http://datatracker.ietf.org/doc/draft-ejzak-mmusic-data-channel-sdpneg/

It seems there's a lot of interest in this document, so if folks want it to
be considered for adoption in the MMUSIC WG, PLEASE move the discussion to
the MMUSIC WG list.  The chairs need the discussion there in order to get a
sense of WG interest, so you might want to include a link to this thread of
discussion in a NEW email to the MMUSIC WG mailing list and point this
discussion to that thread. But, please do not crosspost.

Thanks,
Mary.


On Mon, Feb 24, 2014 at 7:18 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> A few questions on  draft-ejzak-dispatch-webrtc-data-channel-sdpneg:
>
>
>
> *Q1:*
>
>
>
> Instead of defining a new SDP webrtc-DataChannel attribute, would it be
> possible to define the parameters as extension parameters to the SDP
> sctpmap attribute (yes, I know the ABNF currently does not allow that)?
>
>
>
> Example:
>
>
>
> *a=sctpmap:1000 webrtc-datachannel 1;stream=6;subprotocol="CLUE"*
>
>
>
> I am not saying this would be good or bad - at this point I just want to
> understand whether it would work.
>
>
>
>
>
> *Q2:*
>
>
>
> Would it be possible, for subprotocol values, use the same IANA
> registry/values as for the WebRTC Data Channel Protocol?
>
>
>
>
>
> *Q3:*
>
>
>
> It would be good to clarify, if only one data channel is requested (or,
> even required), that the stream value must be the same in the Offer and
> Answer.
>
>
>
>
>
> *Q4:*
>
>
>
> I have issues with the max_retr, max_time and unordered parameters.
>
>
>
> First, they seem to specify SENDING capabilities, rather than RECEIVING
> capabilities.
>
>
>
> Second, they seem to describe characteristics associated with the
> subprotocol, in which case they could be specified in the associated
> subprotocol specification.
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bb047944eefc504f340125e
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<div dir=3D"ltr">Why are we having this discussion on the RTCWEB mailing li=
st rather than the MMUSIC mailing list? &nbsp; This document was dispatched=
 to MMUSIC at IETF-88: <a href=3D"http://trac.tools.ietf.org/wg/dispatch/tr=
ac/wiki">http://trac.tools.ietf.org/wg/dispatch/trac/wiki</a><div>
And, the nifty datatracker tool tells us that this (expired) document has b=
een replaced by an individual draft that has been submitted to the MMUSIC W=
G:&nbsp;<br><div><a href=3D"http://datatracker.ietf.org/doc/draft-ejzak-mmu=
sic-data-channel-sdpneg/">http://datatracker.ietf.org/doc/draft-ejzak-mmusi=
c-data-channel-sdpneg/</a></div>
<div><br></div><div>It seems there&#39;s a lot of interest in this document=
, so if folks want it to be considered for adoption in the MMUSIC WG, PLEAS=
E move the discussion to the MMUSIC WG list. &nbsp;The chairs need the disc=
ussion there in order to get a sense of WG interest, so you might want to i=
nclude a link to this thread of discussion in a NEW email to the MMUSIC WG =
mailing list and point this discussion to that thread. But, please do not c=
rosspost. &nbsp;</div>
<div><br></div><div>Thanks,</div><div><div>Mary.&nbsp;</div></div></div></d=
iv><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Fe=
b 24, 2014 at 7:18 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@e=
ricsson.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"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">A few questions on&nbsp; draft-=
ejzak-dispatch-webrtc-data-channel-sdpneg:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q1:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Instead of defining a new SDP w=
ebrtc-DataChannel attribute, would it be possible to define the parameters =
as extension parameters to the SDP sctpmap attribute (yes, I know the ABNF =
currently does not allow that)?<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Example:<u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:65.2pt"><b><span lang=3D"EN-US"=
>a=3Dsctpmap:1000 webrtc-datachannel 1;stream=3D6;subprotocol=3D&rdquo;CLUE=
&quot;<u></u><u></u></span></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I am not saying this would be g=
ood or bad &ndash; at this point I just want to understand whether it would=
 work.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q2:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Would it be possible, for subpr=
otocol values, use the same IANA registry/values as for the WebRTC Data Cha=
nnel Protocol?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q3:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">It would be good to clarify, if=
 only one data channel is requested (or, even required), that the stream va=
lue must be the same in the Offer and Answer.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q4:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have issues with the max_retr=
, max_time and unordered parameters.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">First, they seem to specify SEN=
DING capabilities, rather than RECEIVING capabilities.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Second, they seem to describe c=
haracteristics associated with the subprotocol, in which case they could be=
 specified in the associated subprotocol specification.<u></u><u></u></span=
></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<span class=3D"HOEnZb">=
<font color=3D"#888888"><u></u><u></u></font></span></span></p><span class=
=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<u></u><u></u></span></=
p>
</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>

--047d7bb047944eefc504f340125e--


From nobody Tue Feb 25 14:00:57 2014
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Date: Tue, 25 Feb 2014 16:59:32 -0500
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On 2/21/2014 11:27 AM, Magnus Westerlund wrote:
> 5. Section 3.2:
>     U-C 7:  Proxy browsing, where a browser uses data channels of a
>             PeerConnection to send and receive HTTP/HTTPS requests and
>             data, for example to avoid local Internet filtering or
>             monitoring.
>
> >From a data channel perspective this doesn't look strange at all,
> however I get the shivers from the security implications of this. To my
> understanding Peer A using Peer B to proxy browse must have B fetch all
> elements that A needs into its sandbox and then transfer it to A over
> the PeerConnections Data Channel. Thus, all you do are revealed to B,
> massive privacy consideration. Or am I missing someway that A can tunnel
> its HTTP request through B without having B see the content of those
> requests, I am especially concerned with HTTPS.
>
> If this use case is going to stay, I do want some security consideration
> discussion around it.

I'll note that there are active projects looking to use this.  Yes, it 
will likely expose non-https data to the proxy.
https traffic should in theory be as protected as it normally is against 
proxies.

This usage would support various forms of monitoring-avoidance (and 
avoidance of relying on local network resources such as in a WiFi 
hotspot).  It's not a direct replacement for Tor, but it's also much 
less intrusive than Tor.  It inherently requires some level of trust of 
the person proxying for you.

We can expose some of the security considerations, but the primary ones 
would be dependent on how this is used, not that this protocol has the 
characteristics that would support such a use.  This is mostly saying 
"we want it to be low overhead for stream creation and initial messages 
and support a bunch of them at once".


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/25/14 12:51 PM, Christer Holmberg wrote:
> Hi Paul,
>
>>> My question was regarding the re-transmission values etc.
>>>
>>> Why do the peers need to negotiate those? Why would an intermediary need to know about
>>> those? Why not simply define those in the protocol description, and each endpoint supporting the >> protocol will then know what values to use?
>>
>> IIUC, the webrtc people are "humoring* us by making provision for registered subprotocol names, > and using SDP to negotiate channels. I suspect that for the most part they have no intention of
>> using those.
>> (The subprotocol name is *optional*.)
>>
>> So these parameters are needed to properly initialize a channel when the subprotocol is not
>> specified.
>
> Why would someone NOT specify the subprotocol?

Because there is no defined subprotocol for what they are doing?

In the common webrtc use case with two browsers talking to the same web 
service, and the web service feeding the needed javascript to the 
browsers, they just "know" the protocol they are using. So they may just 
"make it up", and have nothing registered.

It is when they need to interoperate with something else that it becomes 
important to name and register the protocol.

> Especially considering Richard's draft, I thought the whole idea was to be able to negotiate the subprotocol using SDP.

That's for us, not them. :-)

They will be using DCEP.

>> Also, we still have an open question about what attributes must be present in a subprotocol
>> specification. It is at least possible that some subprotocols may work with a variety of settings of
>> these attributes.
>
> Well, in that case the subprotocol specification should define the additional SDP attributes needed.
>
>> This draft should probably say more about this. It could say that:
>>
>> * the SDP MUST be consistent with the attributes specified in the subprotocol definition, OR
>>
>> * the SDP MUST NOT specify attributes specified in the subprotocol definition, OR
>>
>> * attributes specified in the SDP override those specified in the subprotocol definition.
>
> If people see a need for these attributes, fine. But, I still have not seen an example where it would be needed (e.g. the UDP RFC does not specify the timers for SIP - RFC 3261 does :)

I was talking in general. The answers may be different for each attribute.

	Thanks,
	Paul


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On 2/24/2014 5:05 AM, Magnus Westerlund wrote:
> On 2014-02-22 02:14, Martin Thomson wrote:
>
>>
>> Maybe Magnus is looking for the mechanism described in
>> http://tools.ietf.org/html/draft-thomson-mmusic-rtcweb-bw-consent-00,
>> which might allow for control below the limit that the congestion
>> control permits.  I basically abandoned this, because it's small
>> potatoes. It's trivial to revoke consent if you notice a misbehaving
>> peer and then you only have to wait until the consent timer runs out
>> on the sender.  30s.
>>
>>
> My position is that the above mitigation is unlikely to have significant
> impact on this attack. This as it is difficult to know what values to
> set. They vary over time and depending on network attachment, what
> services the user tries to use etc.
>
> There are two reasons I wanted this attack to be considered. First, it
> provides a clear requirement on having congestion control as first level
> mitigation.
>
> Secondly, I think this could become a significant issue as data channel
> PeerConnections can be opened without user consent. A malicious JS that
> is sufficient well spread with a well working find and connect could
> establish a large mesh of peer connections that could come close to
> saturate each endpoints local access link, resulting in very heavy loads
> on the network, even with congestion control. With congestion control
> you can likely prevent congestion collapse, but you should be fully
> capable of driving a network into a state of "mostly useless",
> especially a network suffering from buffer bloat inside of the ingress
> nodes.

Of course, any smart attacker could do the same today by opening large 
numbers of http or WebSocket connections that cross the same networks, 
especially if the target isn't the remote endpoint but intermediary 
networks as you indicated.  A difference is that it may be harder to 
find destination nodes that will accept such connections and return or 
accept data across the link you want to attack, though the open requests 
themselves can be an attack. However, in real terms, there are usually 
nodes in an access network that return some data to requests (such as 
routers with remote admin turned on, servers (or rogue servers), 
especially on non-standard ports (not 80), etc.  Simply finding routers 
that respond on 80 or 8080, etc and you can send large bogus HTTP 
request to or get to return static images repeatedly might be enough to 
construct an equivalent attack.  (Even more-so perhaps with IW10).

So: congestion control: of course.  Perhaps this makes it a little 
simpler to set up certain types of indirect attacks - but I don't think 
it adds a fundamental class of attacks.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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On Feb 24, 2014, at 5:32 PM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> Hi,
> (as individual)
Hi Magnus,

thank you very much for the detailed review. Please see my comments =
in-line.

Best regards
Michael
>=20
> I have reviewed the WebRTC Data Channel Establishment protocol
> (draft-ietf-rtcweb-data-protocol-03) and have some comments:
>=20
> 1. Section 4:
> Shouldn't this section discuss the priority field?
I added in the list of consistent properties:

<t>the priority of the data Channel.</t>

and in the text below that enumeration:

??????
>=20
> 2. Section 4:
>=20
> The method
>   used to determine which side uses odd or even is based on the
>   underlying DTLS connection role when used in WebRTC, with the side
>   acting as the DTLS client using even stream identifiers.
>=20
> Isn't this unnecessary using the vague word of WebRTC instead of =
simply
> pointing to the DTLS roles of the established data channel?
The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
In that case DTLS is not used and you can not refer to the DTLS role.
That is why the restriction is used.
>=20
> 3. Section 5.1:
>      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
>         a partial reliable in-order bi-directional Communication
>         channel.  User messages might not be transmitted or
>         retransmitted after a specified life-time given in milli-
>         seconds in the Reliability Parameter.  This life-time starts
>         when providing the user message to the Javascript engine.
>=20
> I think the use of "Javascript engine" in the last sentence is =
unclear.
> Can we provide a implementation neutral definition, like "This =
life-time
> starts when requesting transmission of the user message."?
This comment was already made on the list and I changed the text to

This life-time starts when providing the user message to the protocol =
stack.

OK?
>=20
> 4. Section 5.1:
>   Label: Variable Length (sequence of characters)
>      The name of the channel.  This may be an empty string.
>=20
>   Protocol: Variable Length (sequence of characters)
>      The protocol for the channel.  If this is an empty string the
>      protocol us unspecified.  If it is an non-empty string, it
>      specifies an IANA-registered protocol (see Section 8.4).
>=20
> Both of these fields are strings, shouldn't a particular encoding be
> specified here? Like UTF-8. Secondly, what values are allowed, the =
full
> set of Unicode?
You are right. Any need for restrictions? We only need to be able to
transform it to a DomString.
So I changed it to:

<t hangText=3D'Label: Variable Length (sequence of characters)'>
<vspace blankLines=3D'0'/>
The name of the channel as a UTF-8 encoded string.
This may be an empty string.</t>

<t hangText=3D'Protocol: Variable Length (sequence of characters)'>
<vspace blankLines=3D'0'/>
The protocol for the channel as a UTF-8 encoded string.
If this is an empty string the protocol us unspecified.
If it is an non-empty string, it specifies an IANA-registered protocol
(see <xref target=3D'iana_protocol'/>).</t>
>=20
> 5. Section 6:
> All Data Channel Establishment Protocol messages MUST be sent
>   requesting ordered delivery and using reliable transmission.
>=20
> I wonder of the use of requesting ordered delivery and using reliable
> transmission, from an SCTP stream perspective, wouldn't using in both
> places be appropriate? Or is how object which has been requested to be
> transmitted unordered interact in SCTP with the ordered ones?
I'm sorry, I don't understand what you are asking...

The reason of this restriction is that this restriction combined with

However, before the DATA_CHANNEL_ACK message or any other message has =
been
received on a data channel, all other messages containing user data and
belonging to this data channel MUST be sent ordered, no matter whether =
the
data channel is ordered or not.

makes sure that the SCTP at the peer delivers the DATA_CHANNEL_OPEN =
message
first on a stream.


> 6. Section 7:
>=20
> I think this section can be beefed up a bit. First make clear that the
> Data Channel's required usage of DTLS ensures that the message =
integrity
> and possible source authentication as well as confidentiality. Then
> going over any security risks with a malicous peer using this =
protocol.
> Can a malicous side screw up the peer using this protocol? What are =
the
> implications?
Just to double check: Aren't the referenced documents the ones which
discuss all security stuff in one place?
>=20
> 7. Section 8.2:
>=20
> This registry reserves 0xff without any motivation, please add one.
I added:

The value 0xff has been reserved for future extensibility.

>=20
> 8. Section 8.3:
> If I define a channel behavior that allows mixed ordered and =
unordered,
> how would I register this?
You can't. For SCTP properties like ordered/unordered delivery are
on a per message base. In RTCWeb it was decided to make per channel
base. So you can't have a data channel supporting a mixtures of
ordered and unordered messages.
>=20
> 9. Section 8.4:
>=20
> I have a suggestion for this registries name, instead of "Protocol
>   Registry" I propose  "DCEP Protocol Identification Registry"
The problem I have with this definition is the following:
Your name suggests that the use of this entity is restricted to
DCEP. But isn't the protocol a property of a data channel, no matter
whether negotiated by used DCEP or not?
>=20
> 10. Section 8.4:
>=20
> There is missing any specification of what type of strings I may
> register. Also, no length limitation, although the protocol allows at
> maximum of 255 octets of encoded characters.
I don't understand the limitation to 255 octets. The length field
is 16 bit, so you can have up to 65535 octets. Therefore I changed:
<t>A name for the protocol;</t>
to
<t>A name for the protocol which length is smaller than 65536 when using
an UTF-8 encoding;</t>
>=20
> 11. This comments concerns the relation to the Data Channel
> specification. So my interpretation is that the WebRTC Data channel
> draft has become both the base for this specification as well as the =
one
> that specifies that the DCEP shall be implemented and supported. For
> WebRTC I don't think this matters much, but if someone likes to re-use
> the basic Data Channel specification, this structure makes it more
> difficult and have no need for the bi-directional negotiated parameter
> settings that the DCEP provides. In that case one have to sub-set the
I think the data channel draft discusses the data channels independent
how they are opened. This covers data parameters, user data transmission
and closing of data channels. The data channel protocol draft discusses
an in-band protocol for setting up data channels.
> data channel specification. I wonder if it would be better to move =
some
> normative statements around, so that the chain of implementation
> requirement goes draft-ietf-rtcweb-transports ->
> draft-ietf-rtcweb-data-protocol -> draft-ietf-rtcweb-data-channel thus
> providing a cleaner and more modular build up of the protocols.
I think you can implement draft-ietf-rtcweb-data-channel without
draft-ietf-rtcweb-data-protocol, but not vice versa.

So you don't like the statement:

   A simple protocol for internal negotiation is specified in
   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.

from draft-ietf-rtcweb-data-channel. What would you suggest?
>=20
> This a suggestion based on that it looks like others, like CLUE is =
going
> to use the Data Channel specification, but I am uncertain if they want
> DCEP or not.
Understood.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Date: Tue, 25 Feb 2014 17:26:39 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/25/14 12:54 PM, Makaraju, Maridi Raju (Raju) wrote:
> Hi Paul,
>
> Please see comments inline.
>
>> Christer - comment at end
>>
>> On 2/25/14 4:08 AM, Christer Holmberg wrote:
>>> Hi Raju,
>>>
>>>>> Anyway, can you give me an example of a case where you want to use
>>>>> SDP, and where you need to negotiate the re-transmission values etc?
>>>>
>>>> [Raju] A simple example would be 2 browsers talking thru an intermedia
>> proxy, which wants to know what protocols are
>>>> being negitiated and used as part of the session.
>>>
>>> Sure. I don't question the use-case/need for using SDP to negotiate
>> channel usages.
>>>
>>>> 1. Calling client creates data channel (using
>> http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannel)
>> with desired
>>>> attributes and sets negotiated=true. So, calling browser saves the
>> attributes for the data channel but won't use DCP.
>>>> 2. These attributes are sent via SDP to peer client (via proxy).
>>>> 3. Peer client sets these same attributes while creating data channel and
>> sets negotiated=true but won't do DCP.
>>>> 4. Peer client echo the same attributes back in SDP answer.
>>>>
>>>> Now, data channel stacks on both ends will use same attributes, similar
>> to DCP use case, with the only difference being DCP is not used here.
>>>
>>> My question was regarding the re-transmission values etc.
>>>
>>> Why do the peers need to negotiate those? Why would an intermediary need
>> to know about those? Why not simply define those in the protocol
>> description, and each endpoint supporting the protocol will then know what
>> values to use?
>>
>> IIUC, the webrtc people are "humoring* us by making provision for
>> registered subprotocol names, and using SDP to negotiate channels. I
>> suspect that for the most part they have no intention of using those.
>> (The subprotocol name is *optional*.)
>>
>> So these parameters are needed to properly initialize a channel when the
>> subprotocol is not specified.
>
> [Raju] If I understand correctly, these parameters simply define the
> data channel transport properties. Yes, they may have some correletion with
> subprotocol, but that's not a one-to-one mapping.
> For example, a given subprotocol could run on a reliable or
> unreliable transport; depending on app needs it may use unreliable
> transport and deal with reliablity at subprotocol stack (or ignore it).

In some cases. But in general it is easy and common to define a protocol 
that needs a reliable transport, and it won't work without one.

What if every application in the world that uses TCP had the option to 
specify if it was reliable or not? How many would screw it up?

>> Also, we still have an open question about what attributes must be
>> present in a subprotocol specification. It is at least possible that
>> some subprotocols may work with a variety of settings of these attributes.
>
> [Raju] Right, this fits into my explanation above.
>
>>
>> This draft should probably say more about this. It could say that:
>>
>> * the SDP MUST be consistent with the attributes specified in the
>> subprotocol definition, OR
>>
>> * the SDP MUST NOT specify attributes specified in the subprotocol
>> definition, OR
>>
>> * attributes specified in the SDP override those specified in the
>> subprotocol definition.
>
> [Raju] Good point. I like item 1 above; sure we can add it to this
> draft. Then individual subprotocol drafts
> (e.g. draft-ejzak-dispatch-msrp-data-channel-00) could get into the
> details of what that consistency is.

	Thanks,
	Paul


From nobody Tue Feb 25 14:27:12 2014
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Hi Magnus,

thank you very much for the detailed review.
Please see my comments below.

Best regards
Michael


On Feb 21, 2014, at 5:27 PM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> Hi,
> (as individual)
>=20
> I have reviewed the WebRTC Data Channels draft
> (draft-ietf-rtcweb-data-channel-07) and have some comments.
>=20
> 1. Section 1:
> Non-media data types in the context of WebRTC are handled by using
>   SCTP [RFC4960] encapsulated in DTLS [RFC6347].
>=20
> There are many places in this draft where "media data" is a reference =
to
> the RTP transport media. I would suggest to review this and consider =
to
> change this to "RTP Transport media", or "RTP media". The reason is =
that
> I expect also real-time media to flow over the data channel.
OK. Based on another suggestion on the mailing list, I use
SRTP media. Although 'non SRTP media' looks strange...
>=20
> 2. Section 1:
> As the other SCTP extensions are discussed in the introduction, I =
wonder
> why NADA and PR-Policy isn't.
most likely they were not required when this text was written...

So I changed it to:

<t>SCTP as specified in <xref target=3D'RFC4960'/> with the partial =
reliability
extension defined in <xref target=3D'RFC3758'/> and the additional =
policies
defined in <xref target=3D'I-D.ietf-tsvwg-sctp-prpolicies'/>
provides multiple streams natively with reliable, and the relevant=20
partially-reliable delivery modes for user messages.
[...]
Using <xref target=3D'I-D.ietf-tsvwg-sctp-ndata'/> allows to interleave
large messages to avoid the monopolization and adds the support of
prioritising of SCTP streams.</t>

>=20
> 3. Section 1:
> Section 5 arguments SCTP over DTLS over UDP;
>=20
> "arguments" looks strange in the above sentence fragment.
Replaced by "discusses".
>=20
> 4. Section 3.1:
>           Note that
>           at any time there may be no media channels, or all media
>           channels may be inactive, and that there may also be =
reliable
>           data channels in use.
>=20
> "no media channels" is a bit strange here as we not really call =
anything
> regarding RTP channels. I would suggest to change this to "RTP Packet
> Streams" or "RTP Stream".
According to the above change it reads now:

Note that at any time there may be no SRTP media channels, or all SRTP =
media
channels may be inactive, and that there may also be reliable data =
channels
in use.</t>


>=20
> 5. Section 3.2:
>   U-C 7:  Proxy browsing, where a browser uses data channels of a
>           PeerConnection to send and receive HTTP/HTTPS requests and
>           data, for example to avoid local Internet filtering or
>           monitoring.
>=20
>> =46rom a data channel perspective this doesn't look strange at all,
> however I get the shivers from the security implications of this. To =
my
> understanding Peer A using Peer B to proxy browse must have B fetch =
all
> elements that A needs into its sandbox and then transfer it to A over
> the PeerConnections Data Channel. Thus, all you do are revealed to B,
> massive privacy consideration. Or am I missing someway that A can =
tunnel
> its HTTP request through B without having B see the content of those
> requests, I am especially concerned with HTTPS.
>=20
> If this use case is going to stay, I do want some security =
consideration
> discussion around it.
Lets discuss whether this will stay or not... I find it an interesting
use case, would prefer to keep it, and would therefore prefer to
deal with it appropriately in the security considerations.
>=20
> 6. Section 5:
> "   o  The congestion control is modifiable for integration with media
>      stream congestion control."
>=20
> As this statement is only partial correct. There is no current =
mechanism
> to negotiate congestion control, nor does it exist any alternative
> defined SCTP congestion control. Thus, I would like to have this
> sentence modified. I am however not certain to what the authors =
consider
> important. It is the fact that it is possible in the future to define =
a
> new CC algorithm and use that?
Correct.
>=20
> 7. Section 5:
>   o  provides privacy for the SCTP control information.
>=20
> Is "privacy" really the right word here? Either being explicit, =
"source
> authentication, integrity and confidentiality" or something like =
"security"?
The bullet list focuses on the benefits of using SCTP/DTLS/UDP instead
of DTLS/SCTP/UDP as indicated by the sentence before the bullet list.
Source authentication and integrity for SCTP control information is
also provided by DTLS/SCTP/UDP, since you can protect it by the AUTH =
chunk.
Privacy, however, for this information can't be provided by =
DTLS/SCTP/UDP,
since only the user data is encrypted. So I guess the bullet point as
cited above is correct.
>=20
> 8. section 5:
>=20
>   DTLS implementations used for this stack SHOULD support controlling
>   fields of the IP layer like the Don't Fragment (DF)-bit in case of
>   IPv4 and the Differentiated Services Code Point (DSCP) field =
required
>   for supporting [I-D.ietf-rtcweb-qos].
>=20
> First of all this reference must be changed. What I am currently
> uncertain of is what it should point on, most likely
> draft-dhesikan-tsvwg-rtcweb-qos-04. But, I am not yet certain how the
> scope is going to be divided between that document and
> draft-ietf-rtcweb-transports.
I think it is much better to move the DSCP text to
I-D.ietf-tsvwg-sctp-dtls-encaps, where corresponding text regarding the =
DF bit
already exists and just get rid of the above paragraph.
If OK, this might be a WG LC comment on I-D.ietf-tsvwg-sctp-dtls-encaps
>=20
> 9. Section 5:
>=20
> The initial Path MTU
>   at the IP layer MUST NOT exceed 1200 bytes for IPv4 and 1280 for
>   IPv6.
>=20
> I have to say that I think MUST NOT in the above is to strict. I think
> SHOULD NOT is appropriate. The reason is that a deployment may =
actually
> know that a larger initial MTU is better. This value may also change =
due
> to how the networks develop in the future.
Changed.
>=20
> 10. Section 5:
>=20
> Since SCTP does not support the
>   negotiation of a congestion control algorithm yet, alternate
>   congestion controls SHOULD either only require a different sender
>   side behavior using existing information carried in the association
>   or need also specify a negotiation of of a congestion control
>   algorithm.
>=20
> First of all I don't like the SHOULD in this sentence.
>=20
> Secondly, I think we need to get a good agreement and statement what =
to
> do with SCTP's congestion control algorithm in the future. I don't =
like
> this statement saying basically agree on anything and use that. I =
think
> the IETF specification should specify something to use that we know is
> safe to deploy. Thus, I would strongly prefer this to simply to be
> changed to say that: We know we want a new congestion control that
> better can handle interaction the the RTP packet streams and its
> congestion control, but that will not be initially available. Thus use
> the specified congestion control and consider these issues. And if a =
new
> CC algo becomes available that can be used and should be negotiated
> between the endpoints somehow.
The intention of the above sentence to make clear that such a CC can =
either
be a sender side only mechanism or needs to add negotiation within the
association setup. Not more. Not less. This goes back to a discussion
at an IETF session. For me the paragraph basically covers what you want.
Could you be more specific, where you disagree?
>=20
> 11. Section 6.1:
>   Once support for message interleaving as currently being discussed =
in
>   [I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.
>=20
> I don't know why you have formulated this, so awkward? NDATA is
> currently listed as a normative reference. This document is going to
... put a consensus from an RTCWeb meeting into my words...
> hand in the RFC-editor queue until it becomes available or an AD =
orders
> some rewriting. Thus, why not simply  write a simpler statement making
> clear that one SHOULD support it.
OK. Makes sense to me. What about:

<t>The support for message interleaving as defined in
<xref target=3D'I-D.ietf-tsvwg-sctp-ndata'/> SHOULD be used.</t>

>=20
> I would also note that there are disagreement between this and the =
text
> in Section 6.6.:
>=20
> To overcome this
>   limitation, [I-D.ietf-tsvwg-sctp-ndata]  defines an extension to
>   support message interleaving.  Once this extension is available, it
>   MUST be used.
>=20
> Thus pick the level of requirement and then simply state that with the
> motivation why. Personally I think SHOULD would be acceptable. There =
are
> clear motivation why (in 6.6) why one SHOULD implelement it, and the
> basic functionality is still there if it is not supported and its =
usage
> will be negotiated as SCTP association creation.
OK. Changed to:

To overcome this limitation, <xref target=3D'I-D.ietf-tsvwg-sctp-ndata'/>
defines an extension to support message interleaving, which SHOULD be =
used.

So it is consistently SHOULD. The question is why SHOULD. So when not to
use it? Currently the problem is the availability of stable code, but
that is not an argument for a SHOULD vs. a MUST, right?
>=20
> 12. Section 6.2:
> Additionally,
>   the negotiation SHOULD include some type of congestion control
>   selection.
>=20
> Another congestion control statement that I am not happy with. See =
issue
> 10. There is no default and it is also not clear on what to really
> negotiate.
The default is the standard one. Currently there is no other and there
is no way for negotiation. So how to deal with it? Add a statement about
the default? Or remove the negotiation now and add it to a document
defining an alternate CC?
>=20
> 13. Section 6.3:
>=20
>   SCTP defines a stream as a unidirectional logical channel existing
>   within an SCTP association one to another SCTP endpoint.
>=20
> The "one" looks spurious.
Removed.
>=20
> 14. Section 6.4:
> These priorities MUST NOT be strict priorities.
>=20
> Ok, this is one part of the agreement from the earlier meeting. There
> was also agreement on using weighted fair queuing to solve the
> priorities. The inter stream priority handling needs much firmer
> specification.
My question: What are the requirements from user JS application.
I would like to leave as much room as possible for implementing
these requirements...So up to know I have heard non-strict priorities...

This needs to be in tune with the W3C...
>=20
> 15. Section 6.5:
> "This can be based on the DTLS role (the client picks
>   even stream identifiers, the server odd stream identifiers) or done
>   in a different way."
>=20
> This is very unclear and you can't know what will happen. I think it
> would be better to say: Unless otherwise defined or negotiated, the
> streams are picked based on th DTLS role.  ...
OK. It reads now:

Unless otherwise defined or negotiated, the streams are picked based on
the DTLS role (the client picks even stream identifiers,
the server odd stream identifiers).

>=20
> 16. Section 6.5:
> "In addition to choosing a stream, the application SHOULD also
>   inform the protocol of the options to use for sending messages."
>=20
> So what is the application, and what is the protocol in this sentence?
Good question... The whole paragraph reads strange... So I changed its
end to:
[...]
In addition to choosing a stream, the application SHOULD also determine =
the
the options to use for sending messages.
The application MUST ensure in an application-specific manner that
the application at the peer will also know selected stream to
be used, and the options for sending data from that side.</t>

>=20
> 17. Section 7.
>=20
> I find this insufficient. I would this section to answer the following
> questions:
>=20
> 1. Any security issues that arise from the combination of mechanism.
> 2. Any significant issue that one needs to know about the behavior of
> the mechanisms, any of the normative specs that should be highlighted?
> 3. Clarification on what security properties one achieve with the
> defined mechanism.
Isn't/shouldn't all this be handled in the referenced documents?
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Tue Feb 25 14:27:13 2014
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-data-channel-07
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On Feb 23, 2014, at 10:54 AM, Barry Dingle <btdingle@gmail.com> wrote:

> In reference to your first Section 1 comments, I agree that we need to =
clearly differentiate the SRTP media and the SCTP media. (Note - SRTP - =
not RTP !)
I changed it to SRTP media...
>=20
> I agree that there could be real-time media that can be sent over SCTP =
Channels are still meet the criteria for 'real-time' as being 'received =
within a few hundred milliseconds of being generated' - see WebRTC =
Overview section 2.4.=20
>=20
> Maybe we need to differentiate the Channel types - and then refer to =
the data going over those Channels. For instance,if we define "SRTP =
Channels" and "SCTP Channels", we could then refer to 'SRTP Channel =
media' which would have a specific meaning.=20
>=20
> The best place for the definition of SRTP Channel and SCTP Channel is =
probably in the WebRTC Overview document.=20
I agree...

Best regards
Michael
>=20
> Cheers,
> /Barry Dingle
>=20
>=20
>=20
> On Sat, Feb 22, 2014 at 3:27 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> Hi,
> (as individual)
>=20
> I have reviewed the WebRTC Data Channels draft
> (draft-ietf-rtcweb-data-channel-07) and have some comments.
>=20
> 1. Section 1:
> Non-media data types in the context of WebRTC are handled by using
>    SCTP [RFC4960] encapsulated in DTLS [RFC6347].
>=20
> There are many places in this draft where "media data" is a reference =
to
> the RTP transport media. I would suggest to review this and consider =
to
> change this to "RTP Transport media", or "RTP media". The reason is =
that
> I expect also real-time media to flow over the data channel.
>=20
> 2. Section 1:
> As the other SCTP extensions are discussed in the introduction, I =
wonder
> why NADA and PR-Policy isn't.
>=20
> 3. Section 1:
> Section 5 arguments SCTP over DTLS over UDP;
>=20
> "arguments" looks strange in the above sentence fragment.
>=20
> 4. Section 3.1:
>            Note that
>            at any time there may be no media channels, or all media
>            channels may be inactive, and that there may also be =
reliable
>            data channels in use.
>=20
> "no media channels" is a bit strange here as we not really call =
anything
> regarding RTP channels. I would suggest to change this to "RTP Packet
> Streams" or "RTP Stream".
>=20
> 5. Section 3.2:
>    U-C 7:  Proxy browsing, where a browser uses data channels of a
>            PeerConnection to send and receive HTTP/HTTPS requests and
>            data, for example to avoid local Internet filtering or
>            monitoring.
>=20
> >=46rom a data channel perspective this doesn't look strange at all,
> however I get the shivers from the security implications of this. To =
my
> understanding Peer A using Peer B to proxy browse must have B fetch =
all
> elements that A needs into its sandbox and then transfer it to A over
> the PeerConnections Data Channel. Thus, all you do are revealed to B,
> massive privacy consideration. Or am I missing someway that A can =
tunnel
> its HTTP request through B without having B see the content of those
> requests, I am especially concerned with HTTPS.
>=20
> If this use case is going to stay, I do want some security =
consideration
> discussion around it.
>=20
> 6. Section 5:
> "   o  The congestion control is modifiable for integration with media
>       stream congestion control."
>=20
> As this statement is only partial correct. There is no current =
mechanism
> to negotiate congestion control, nor does it exist any alternative
> defined SCTP congestion control. Thus, I would like to have this
> sentence modified. I am however not certain to what the authors =
consider
> important. It is the fact that it is possible in the future to define =
a
> new CC algorithm and use that?
>=20
> 7. Section 5:
>    o  provides privacy for the SCTP control information.
>=20
> Is "privacy" really the right word here? Either being explicit, =
"source
> authentication, integrity and confidentiality" or something like =
"security"?
>=20
> 8. section 5:
>=20
>    DTLS implementations used for this stack SHOULD support controlling
>    fields of the IP layer like the Don't Fragment (DF)-bit in case of
>    IPv4 and the Differentiated Services Code Point (DSCP) field =
required
>    for supporting [I-D.ietf-rtcweb-qos].
>=20
> First of all this reference must be changed. What I am currently
> uncertain of is what it should point on, most likely
> draft-dhesikan-tsvwg-rtcweb-qos-04. But, I am not yet certain how the
> scope is going to be divided between that document and
> draft-ietf-rtcweb-transports.
>=20
> 9. Section 5:
>=20
> The initial Path MTU
>    at the IP layer MUST NOT exceed 1200 bytes for IPv4 and 1280 for
>    IPv6.
>=20
> I have to say that I think MUST NOT in the above is to strict. I think
> SHOULD NOT is appropriate. The reason is that a deployment may =
actually
> know that a larger initial MTU is better. This value may also change =
due
> to how the networks develop in the future.
>=20
> 10. Section 5:
>=20
> Since SCTP does not support the
>    negotiation of a congestion control algorithm yet, alternate
>    congestion controls SHOULD either only require a different sender
>    side behavior using existing information carried in the association
>    or need also specify a negotiation of of a congestion control
>    algorithm.
>=20
> First of all I don't like the SHOULD in this sentence.
>=20
> Secondly, I think we need to get a good agreement and statement what =
to
> do with SCTP's congestion control algorithm in the future. I don't =
like
> this statement saying basically agree on anything and use that. I =
think
> the IETF specification should specify something to use that we know is
> safe to deploy. Thus, I would strongly prefer this to simply to be
> changed to say that: We know we want a new congestion control that
> better can handle interaction the the RTP packet streams and its
> congestion control, but that will not be initially available. Thus use
> the specified congestion control and consider these issues. And if a =
new
> CC algo becomes available that can be used and should be negotiated
> between the endpoints somehow.
>=20
> 11. Section 6.1:
>    Once support for message interleaving as currently being discussed =
in
>    [I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.
>=20
> I don't know why you have formulated this, so awkward? NDATA is
> currently listed as a normative reference. This document is going to
> hand in the RFC-editor queue until it becomes available or an AD =
orders
> some rewriting. Thus, why not simply  write a simpler statement making
> clear that one SHOULD support it.
>=20
> I would also note that there are disagreement between this and the =
text
> in Section 6.6.:
>=20
> To overcome this
>    limitation, [I-D.ietf-tsvwg-sctp-ndata]  defines an extension to
>    support message interleaving.  Once this extension is available, it
>    MUST be used.
>=20
> Thus pick the level of requirement and then simply state that with the
> motivation why. Personally I think SHOULD would be acceptable. There =
are
> clear motivation why (in 6.6) why one SHOULD implelement it, and the
> basic functionality is still there if it is not supported and its =
usage
> will be negotiated as SCTP association creation.
>=20
> 12. Section 6.2:
>  Additionally,
>    the negotiation SHOULD include some type of congestion control
>    selection.
>=20
> Another congestion control statement that I am not happy with. See =
issue
> 10. There is no default and it is also not clear on what to really
> negotiate.
>=20
> 13. Section 6.3:
>=20
>    SCTP defines a stream as a unidirectional logical channel existing
>    within an SCTP association one to another SCTP endpoint.
>=20
> The "one" looks spurious.
>=20
> 14. Section 6.4:
> These priorities MUST NOT be strict priorities.
>=20
> Ok, this is one part of the agreement from the earlier meeting. There
> was also agreement on using weighted fair queuing to solve the
> priorities. The inter stream priority handling needs much firmer
> specification.
>=20
> 15. Section 6.5:
> "This can be based on the DTLS role (the client picks
>    even stream identifiers, the server odd stream identifiers) or done
>    in a different way."
>=20
> This is very unclear and you can't know what will happen. I think it
> would be better to say: Unless otherwise defined or negotiated, the
> streams are picked based on th DTLS role.  ...
>=20
> 16. Section 6.5:
> "In addition to choosing a stream, the application SHOULD also
>    inform the protocol of the options to use for sending messages."
>=20
> So what is the application, and what is the protocol in this sentence?
>=20
> 17. Section 7.
>=20
> I find this insufficient. I would this section to answer the following
> questions:
>=20
> 1. Any security issues that arise from the combination of mechanism.
> 2. Any significant issue that one needs to know about the behavior of
> the mechanisms, any of the normative specs that should be highlighted?
> 3. Clarification on what security properties one achieve with the
> defined mechanism.
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: VS: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

>>> So these parameters are needed to properly initialize a channel when=20
>>> the subprotocol is not specified.
>>
>> Why would someone NOT specify the subprotocol?
>
> Because there is no defined subprotocol for what they are doing?
>
> In the common webrtc use case with two browsers talking to the same web s=
ervice, and the web=20
> service feeding the needed javascript to the browsers, they just "know" t=
he protocol they are using. > So they may just "make it up", and have nothi=
ng registered.
>
> It is when they need to interoperate with something else that it becomes =
important to name and=20
> register the protocol.
>
>> Especially considering Richard's draft, I thought the whole idea was to =
be able to negotiate the=20
>> subprotocol using SDP.
>
> That's for us, not them. :-)
>
> They will be using DCEP.

Sure, but my question is about the usage of the attributes in SDP (Richard'=
s draft).

Regards,

Christer


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On Feb 25, 2014, at 11:58 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> On 2014-02-23 10:54, Barry Dingle wrote:
>> In reference to your first _Section 1 comments_, I agree that we need =
to
>> clearly differentiate the SRTP media and the SCTP media. (Note - SRTP =
-
>> not RTP !)
>=20
> So SRTP is an extension of RTP, in fact it is a profile, or rather two
> (RTP/SAVP or RTP/SAVPF). I wrote RTP because I meant no specific RTP
> configuration here, it can be any of the profiles and any of the
> extensions. I know and fully support that RTCWEB uses RTP/SAVPF for =
RTP
> transported media, however in the wider context and possible other =
usage
> of the defined mechanism this may or may not be true. Therefore I see =
a
> need to be generic in that discussion. If one want to emphasize the =
SRTP
> part, then use (S)RTP.
OK, I'll use (S)RTP media...

Best regards
Michael
>=20
>>=20
>> I agree that there could be real-time media that can be sent over =
SCTP
>> Channels are still meet the criteria for 'real-time' as being =
'received
>> within a few hundred milliseconds of being generated' - see WebRTC
>> Overview section 2.4.
>>=20
>> Maybe we need to differentiate the Channel types - and then refer to =
the
>> data going over those Channels. For instance,if we define "SRTP
>> Channels" and "SCTP Channels", we could then refer to 'SRTP Channel
>> media' which would have a specific meaning.
>=20
> Well, around RTP we aren't discussing "channels" at all. Thus, this
> becomes very ambiguous to what it means, unless you have a definition =
to
> go along with it.
>=20
> In the context of a PeerConnection we can have one or more RTP =
session.
>=20
> Also, I don't know what an SCTP channel is. There is an SCTP
> association, which is something between two endpoints. On top of this =
we
> have the SCTP Data channels. So what are you missing or want from =
these
> definitions?
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/25/14 5:29 PM, Christer Holmberg wrote:
> Hi,
>
>>>> So these parameters are needed to properly initialize a channel when
>>>> the subprotocol is not specified.
>>>
>>> Why would someone NOT specify the subprotocol?
>>
>> Because there is no defined subprotocol for what they are doing?
>>
>> In the common webrtc use case with two browsers talking to the same web service, and the web
>> service feeding the needed javascript to the browsers, they just "know" the protocol they are using. > So they may just "make it up", and have nothing registered.
>>
>> It is when they need to interoperate with something else that it becomes important to name and
>> register the protocol.
>>
>>> Especially considering Richard's draft, I thought the whole idea was to be able to negotiate the
>>> subprotocol using SDP.
>>
>> That's for us, not them. :-)
>>
>> They will be using DCEP.
>
> Sure, but my question is about the usage of the attributes in SDP (Richard's draft).

Are you then suggesting that the subprotocol MUST be supplied in the SDP?

I think we need some careful consideration of what rules are common to 
all data channels, and which ones can be different when using SDP rather 
than DCEP. That should consider how something first built using DCEP can 
be migrated to using SDP when it is discovered that it needs to 
interoperate with a sip server.

Unclear if that discussion should take place on rtcweb or mmusic.

	Thanks,
	Paul


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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>
> I still don't think the information belong to SDP - they define protocol characteristics. The protocol specification should define re-transmission timers etc.
>
> Anyway, we can argue about it later. At this moment I think the focus should be on whether to move forward with the mechanism to begin with, and the main feature of the mechanism - which is about negotiating usage of channels :)

This (whether to move forward) is my concern (in addition to Mary's 
about this being the wrong group - MMUSIC is where the SDP should be and 
has been).

The apparent justification for this document is "This document defines 
SDP-based out-of-band negotiation procedures to establish WebRTC data 
channels for transport of well-defined sub- protocols." This seems a 
pretty sparse justification.

We very purposely avoided using SDP for channel setup (after MUCH 
discussion at Boston and elsewhere).  That isn't to say that for 
non-webrtc usage, one could define an SDP used to externally-negotiate 
the settings for DataChannels, which this appears to be.  But I think 
there should a clearer statement as to "why" we should invest the time 
to standardize this now, not just that we "can".  For example, if CLUE 
really wants to use this instead of in-band opens, then maybe there's a 
reason to do so - but don't expect webrtc endpoints to implement this.  
And if so, it should be done in CLUE by request to MMUSIC.

At this point I don't see need to move forward with this, and certainly 
not here.  And I certainly don't want this to delay finalizing the stuff 
needed in MMUSIC for WebRTC use of DataChannels

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi Randell,

I am not sure whether an WebRTC endpoint (e.g. a browser) would have to imp=
lement this. The JS Application can insert the information into the SDP bef=
ore sending it out on the wire (in cases where SDP is used on the wire).

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
Sent: 26 February 2014 00:41
To: rtcweb@ietf.org
Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-c=
hannel-sdpneg-00

On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>
> I still don't think the information belong to SDP - they define protocol =
characteristics. The protocol specification should define re-transmission t=
imers etc.
>
> Anyway, we can argue about it later. At this moment I think the focus=20
> should be on whether to move forward with the mechanism to begin with,=20
> and the main feature of the mechanism - which is about negotiating=20
> usage of channels :)

This (whether to move forward) is my concern (in addition to Mary's about t=
his being the wrong group - MMUSIC is where the SDP should be and has been)=
.

The apparent justification for this document is "This document defines SDP-=
based out-of-band negotiation procedures to establish WebRTC data channels =
for transport of well-defined sub- protocols." This seems a pretty sparse j=
ustification.

We very purposely avoided using SDP for channel setup (after MUCH discussio=
n at Boston and elsewhere).  That isn't to say that for non-webrtc usage, o=
ne could define an SDP used to externally-negotiate the settings for DataCh=
annels, which this appears to be.  But I think there should a clearer state=
ment as to "why" we should invest the time to standardize this now, not jus=
t that we "can".  For example, if CLUE really wants to use this instead of =
in-band opens, then maybe there's a reason to do so - but don't expect webr=
tc endpoints to implement this. =20
And if so, it should be done in CLUE by request to MMUSIC.

At this point I don't see need to move forward with this, and certainly not=
 here.  And I certainly don't want this to delay finalizing the stuff neede=
d in MMUSIC for WebRTC use of DataChannels

--
Randell Jesup -- rjesup a t mozilla d o t com

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Anyhow, I agree we should move this discussion elsewhere.

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Christer Holmberg <christer.holmberg@ericsson.com> wrote:


Hi Randell,

I am not sure whether an WebRTC endpoint (e.g. a browser) would have to imp=
lement this. The JS Application can insert the information into the SDP bef=
ore sending it out on the wire (in cases where SDP is used on the wire).

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
Sent: 26 February 2014 00:41
To: rtcweb@ietf.org
Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-c=
hannel-sdpneg-00

On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>
> I still don't think the information belong to SDP - they define protocol =
characteristics. The protocol specification should define re-transmission t=
imers etc.
>
> Anyway, we can argue about it later. At this moment I think the focus
> should be on whether to move forward with the mechanism to begin with,
> and the main feature of the mechanism - which is about negotiating
> usage of channels :)

This (whether to move forward) is my concern (in addition to Mary's about t=
his being the wrong group - MMUSIC is where the SDP should be and has been)=
.

The apparent justification for this document is "This document defines SDP-=
based out-of-band negotiation procedures to establish WebRTC data channels =
for transport of well-defined sub- protocols." This seems a pretty sparse j=
ustification.

We very purposely avoided using SDP for channel setup (after MUCH discussio=
n at Boston and elsewhere).  That isn't to say that for non-webrtc usage, o=
ne could define an SDP used to externally-negotiate the settings for DataCh=
annels, which this appears to be.  But I think there should a clearer state=
ment as to "why" we should invest the time to standardize this now, not jus=
t that we "can".  For example, if CLUE really wants to use this instead of =
in-band opens, then maybe there's a reason to do so - but don't expect webr=
tc endpoints to implement this.
And if so, it should be done in CLUE by request to MMUSIC.

At this point I don't see need to move forward with this, and certainly not=
 here.  And I certainly don't want this to delay finalizing the stuff neede=
d in MMUSIC for WebRTC use of DataChannels

--
Randell Jesup -- rjesup a t mozilla d o t 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


From nobody Tue Feb 25 15:45:02 2014
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Dear all,

Magnus asked me to send a list of open issues regarding data channels
to the list. Here is my current list:


* Priority
  The W3C hasn't defined it yet. Neither for the (S)RTP media nor for =
the
  data channels. We agreed on using a non strict policy for the data =
channels
  (some sort of wighted fair queueing). That is all.

* Protocol
  It seems not to be clear what needs to be provided when registering a
  (sub)-protocol at IANA. And the name of the registry is unclear...

* SCTP parameters.
  There was discussed the issue how to set SCTP parameters, especially =
path.max.retrans
  and association.max.retrans. Also HB.Interval might be of interest.
  RFC 4060 recommends path.max.retrans=3D5, association.max.retrans=3D10, =
but has multihoming
  in mind. To avoid the dormant state, path.max.retrans =3D =
association.max.retrans should be used.
  I would suggest 10 for this value. Should HEARTBEATs be disabled?

* U-C 7: Proxy browsing

* Alternate CC for SCTP
  Currently there is only the standard CC. However, in some places =
negotiation of CC is
  mentioned.

I'm currently going through the backlog of comments regarding the data =
channels
ID and I'll try to address the issues. If I find other issues, I send an =
update
to the above list.

Best regards
Michael



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Karl,

I sympathise with your goal, but I really do not think RTP header =
extensions are appropriate for this purpose.

Ignoring the semantic mismatch, in order to identify an RTP header =
extension, you need access to the signalling data, and if you have =
access to the signalling, you can signal the QoS parameters without =
using RTP.=20

You state that only the payload is encrypted in SRTP. That is not =
necessarily true. RTP header extensions can be encrypted when RFC 6904 =
is in use, and rtcweb-rtp-usage draft recommends this be done in some =
cases. The signalling channel is also likely encrypted end-to-end in new =
applications, so making it difficult to extract the information you need =
to parse the RTP header extension, even if it is unencrypted.=20

Besides these focussed issues, I would also urge you to consider the =
much broader comments Magnus made. The QoS problem is broad, and a point =
solution based on RTP header extensions - even if it were workable, =
which I doubt - would address only a small part of the problem space.

Colin




On 25 Feb 2014, at 01:45, Karl Stahl <karl.stahl@intertex.se> wrote:
> Colin,
> =20
> If the below were the case, it would be =93DPI guesswork=94 that I =
also advice against. RTP doesn=92t even have unique protocol header =
within UDP =96 it can even be confused with other UDP payload.
> =20
> However, if the RTP is captured in a TURN-flow, as in TRAM Milestone =
3, the network point that this flow is directed to and can apply QoS =
methods relevant to the network (which is not diffserve in Mobile OTT =
and Cable Networks) has not a too difficult tasks. Linking each RTP-flow =
by its ID and sequence number, and picking exactly the right traffic =
type and bandwidth parameters is doable (see inline below, we at Ingate =
do it already)!
> =20
> Also DiffServ DSCP-bits are seldom maintained crossing network =
boundaries, thus carrying no relevant information at the receiving end, =
while the RTP extension header remains unchanged end-to-end. In =
DSCP-bits, there is no bandwidth requirement information when entering =
networks requiring reservation. That is always (and dynamically set) =
available in the RTP extension header for each packet.
> =20
> And, I am sure you are aware of the difficulties of getting DSCP-bits =
through OS sockets, which is even worse with multiple streams over the =
same UDP-port
> =20
> Further, defining the QoS RTP extension header as in RFC5285, does not =
in anyway conflict with other RTP extension headers or DSCP transfer or =
settings.
> =20
> /Karl
> =20
> Fr=E5n: Colin Perkins [mailto:csp@csperkins.org]=20
> Skickat: den 24 februari 2014 23:52
> Till: Karl Stahl
> Kopia: rtcweb@ietf.org; Magnus Westerlund; tram@ietf.org; Harald =
Alvestrand
> =C4mne: Re: [rtcweb] [tram] Payload Types assignments
> =20
> Karl,
> =20
> I strongly disagree with this suggestion. An RTP header extension, =
located at an unknown and variable offset into a packet that does not =
have a well-defined magic number in the header,
> This is how to find the traffic info:
>    The payload of the classifier header extension element can be =
encoded
>    using either the one-byte or two-byte header defined in [rfc5285] =
and
>    shown below in Figure 1 and 2 below.
> =20
>       0                   1                   2                   3
>       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>      |  ID   | len=3D1 |   Namespace   |    Value      |    0 (pad)    =
|
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> =20
>              Figure 1: Classifier Using the One-Byte Header
> =20
>       0                   1                   2                   3
>       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>      |      ID       |    len=3D2      |   Namespace   |    Value      =
|
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> =20
> =20
>              Figure 2: Classifier Using the Two-Byte Header
> =20
> indicated using a dynamically assigned identifier that is conveyed in =
an out-of-band
> --- It is the TURN flow!!! Requested by the ICE protocol for the =
specific media to come!
> and encrypted signalling channel,
> --- Only the payload is encrypted in SRTP =96 leaving id, seq no and =
header ext to be used as intended!
> --- Thus being the appropriate place=85
> is not an appropriate place to put QoS information that has to be =
processed on a per-packet basis.
> If you want DiffServ, you know where to find it.
> --- True, but if it cannot be used=85
> Colin
> =20
> =20
> On 24 Feb 2014, at 22:09, Karl Stahl <karl.stahl@intertex.se> wrote:
> I suggest to the RTCWEB WG that the below from the September and =
October discussions on the relevant [rtcweb] [avtext] [mmusic] lists =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html is =
introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC 5285, to =
allow:
> =20
> (1) WebRTC applications to directly convey QoS related real-time =
traffic info to the network at points where RTP flow is directed to by =
TRAM Milstone 3, to be used by *any network element implementing any =
suitable QoS methods for the particular network* for
> (2) *all* WebRTC browsers *and* clients, under *all* OSs, and *all* =
current and future IP network, to achieve best QoE
> (3) *without* having to force WebRTC into application specific =
networks (such as IMS) instead of using the Internet (including OTT).
> =20
> The only further activity required, is to call for ISPs=92 to review =
whether the traffic information transferred by RFC 5285 is sufficient =
for current and future needs in their network as suggested in below =
repeated =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
> <snip>
> =85two parameters (e.g. two bytes each) are encoded into the RTP =
header extension:
> =20
> A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic scale.
> =20
> B) The quality characteristics for the stream, with the highest bit =
set to 1, we could allocate a bit each for quality type e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.
> =20
> And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.
> </snip>
> =20
> Then this could be assigned numbers to have an RFC in place.
> =20
> With TRAM milestone 3 also place,
> market forces will drive ISPs and browser makers to implement just =
this, without even having it MUST-established.
> =20
> =93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and
> =93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?=94 and vice versa.
> =20
> Please see further emails soon following this one, for details and =
history.
> =20
> /Karl
> =20
> =20
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl Stahl
> Skickat: den 22 oktober 2013 16:37
> Till: 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus Westerlund'
> Kopia: 'Colin Perkins'
> =C4mne: [rtcweb] [avtext] Payload Types assignments was Re: SV: =
[mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
> =20
> Harald, I mostly agree with the quality requirements of different =
real-time traffic that the WebRTC browser/application may use. But =
rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. =20
> =20
> There is actually a good way to show these needs to the network =
(without using the PT, or diffserv bits, which aren=92t sufficient =
anyway).
> =20
> Let's use the RTP header extension field that also is visible outside =
the encrypted payload. A week ago came =
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that =
outlines the usage of the extension field for classification of traffic! =
This document does not yet outline what to put in there and how to =
encode it though.
> =20
> Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 =
discusses other webrtc usages of the RTP header extension in 5.2 (there =
can be many header extensions according to RFC 5285) and in 9 there is =
"WebRTC Use of RTP: Future Extensions".
> =20
> So, it looks obvious to use the RTP header extension to show the =
characteristics and bandwidth requirements to the network. It should not =
introduce any backward incompatibilities either.
> =20
> Such marking is done in every RTP packet so it can be set individually =
for each stream and could even be changed during a session (e.g. when =
limiting the bandwidth based on RTCP feedback). RFC 5286 also specifies =
how RTP extension header usage can be negotiated in SDP. I think this =
could be easily done by the WebRTC browser for "all current and future =
needs" if properly specified now.
> =20
> I suggest that two parameters (e.g. two bytes each) are encoded into =
the RTP header extension:
> =20
> A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic scale.
> =20
> B) The quality characteristics for the stream, with the highest bit =
set to 1, we could allocate a bit each quality e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.
> =20
> And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.
> =20
> Please note the totally different requirements a diffserv and an RSVP =
network have to know, so let=92s put all into these bytes. (E.g. a =
diffserv network don't need the bandwidth usage, but RSVP reservation =
networks (e.g. cable and 3G/4G OTT) do. There one should initially =
reserve the maximum bandwidth indicated, but can later re-reserve.)
> =20
> /Karl
> =20
> PS Microsoft seems to have done work in this field, defining a =
proprietary attribute =93MS Service Quality=94;
> However that seems to apply to the TURN server allocation request and =
would therefore:
> --- Apply to the whole UDP flow, and could not be set for each stream =
individually (with different requirements), and
> --- Does not handle the bandwidth requirement for incoming real-time =
traffic (required to reserve in RSVP type of networks)
> However the quality attributes conveyed and their encoding may  be =
considered.
> =20
> This is 2.2.2.19 MS-Service Quality Attribute from
> http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20=

> =20
> MS-Service Quality Attribute
> The MS-Service Quality attribute is used to convey information about =
the data stream that the protocol client is intending to transfer over =
an allocated port. The protocol client SHOULD<21> include this attribute =
as part of an Allocate request message. A TURN server SHOULD use the =
information in this attribute to make decisions about resource =
allocation, bandwidth prioritization, and data delivery methods. If the =
attribute is not present in the Allocate request message, the TURN =
server SHOULD assume that the data stream is audio with best effort =
delivery. The format of this attribute is as follows...
> ...
> The following stream types are supported in this extension. All other =
stream types are reserved for future use.
> =A7 "0x0001": Audio
> =A7 "0x0002": Video
> =A7 "0x0003": Supplemental Video
> =A7 "0x0004": Data
> Service Quality (2 bytes): The service quality level required by the =
protocol client for the stream.
> The following service quality levels are supported in this extension. =
All other service quality levels are reserved for future use.
> =A7 "0x0000": Best effort delivery.
> =A7 "0x0001": Reliable delivery.
> =20
> =20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Harald Alvestrand
> Skickat: den 8 oktober 2013 13:01
> Till: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] =
WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
> =20
> On 10/08/2013 09:17 AM, Karl Stahl wrote:
> > Hej Magnus,
> >=20
> >> Also, are you really interested in knowing that it is VP9 vs H.264,=20=

> >> isn't
> > the questions this is video of this priority that is important?
> >> I think you need to more carefully consider what are the goals you
> >> try to
> > achieve them.
> >=20
> > Actually, my concern is to get an idea of the maximum bandwidth that
> > could be required for a WebRTC (ICE) setup media flow. Both voice =
and
> > video should be prioritized over data (their individual priority is =
of
> > less importance as long as there is sufficient bandwidth for both).
> =20
> You don't know that without knowing what the application is for.
> In, for instance, a shooter game with voice backchannels, the movement =
and event information (data) is MORE time sensitive than the voice data.
> =20
> >=20
> > With diffserv you don=92t need to know the bandwidth requirement, =
but=20
> > with RSVP reservation (like in cable and mobile networks) you need =
to
> > know how much to reserve. Voice is like 100's kbit/s, video VP8 or
> > H.264 is like 3,5 mbps.
> =20
> Again, without knowing the application, you don't know that.
> The application could decide to use QCIF or HD, and the bandwidth =
variation of screencast (semi-static with sudden, large changes) is =
completely different from that of a talking head, which is again =
completely different from a high-movement scene.
> =20
> >=20
> > To add to the complication of codec variants, the video codecs in=20
> > question for WebRTC have variable bandwidth, and when there is a =
poor=20
> > connection we see Chrome reducing the video window size to reduce =
the bandwidth used...
> >=20
> > I think the payload type field at best can reflect a maximum =
bandwidth
> > to initially reserve bandwidth for, and thereafter make new
> > reservations if the bandwidth changes during the call. So could we
> > change RTP to show maximum bandwidth instead of payload type in that
> > field outside the encrypted payload :) ... Or maybe that is not a =
joke?
> =20
> I think these ruminations only lead to one conclusion:
> =20
> You can't tell what the needed bandwidth is up front without asking =
the application.
> You can't tell what the right priority ranking is without asking the =
application.
> =20
> If you need to know the bandwidth or the priority up front, the =
application has to tell you. Anything else is pure heuristics.
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =20
> =20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
> =20
> =20
> =20



--=20
Colin Perkins
http://csperkins.org/




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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;">Karl,<div><br></div><div>I sympathise with your =
goal, but I really do not think RTP header extensions are appropriate =
for this purpose.</div><div><br></div><div>Ignoring the semantic =
mismatch, in order to identify an RTP header extension, you need access =
to the signalling data, and if you have access to the signalling, you =
can signal the QoS parameters without using =
RTP.&nbsp;</div><div><br></div><div>You state that only the payload is =
encrypted in SRTP. That is not necessarily true. RTP header extensions =
can be encrypted when RFC 6904 is in use, and rtcweb-rtp-usage draft =
recommends this be done in some cases. The signalling channel is also =
likely encrypted end-to-end in new applications, so making it difficult =
to extract the information you need to parse the RTP header extension, =
even if it is unencrypted.&nbsp;</div><div><br></div><div>Besides these =
focussed issues, I would also urge you to consider the much broader =
comments Magnus made. The QoS problem is broad, and a point solution =
based on RTP header extensions - even if it were workable, which I doubt =
- would address only a small part of the problem =
space.</div><div><br></div><div>Colin</div><div><br></div><div><br></div><=
div><br></div><div><br><div><div>On 25 Feb 2014, at 01:45, Karl Stahl =
&lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:</div><blockquote type=3D"cite"><meta http-equiv=3D"Content-Type" =
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--><div lang=3D"SV" link=3D"blue" =
vlink=3D"purple"><div class=3D"WordSection1"><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Colin,<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">If the below were the case, it would be =93DPI =
guesswork=94 that I also advice against. RTP doesn=92t even have unique =
protocol header within UDP =96 it can even be confused with other UDP =
payload. <o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US"=
 =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">However, if the RTP is captured in a TURN-flow, as in =
TRAM Milestone 3, the network point that this flow is directed to and =
can apply QoS methods relevant to the network (which is not diffserve in =
Mobile OTT and Cable Networks) has not a too difficult tasks. Linking =
each RTP-flow by its ID and sequence number, and picking exactly the =
right traffic type and bandwidth parameters is doable (see inline below, =
we at Ingate do it already)!<o:p></o:p></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Also DiffServ DSCP-bits are seldom maintained crossing =
network boundaries, thus carrying no relevant information at the =
receiving end, while the RTP extension header remains unchanged =
end-to-end. In DSCP-bits, there is no bandwidth requirement information =
when entering networks requiring reservation. That is always (and =
dynamically set) available in the RTP extension header for each =
packet.<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">And, I am sure you are aware of the difficulties of =
getting DSCP-bits through OS sockets, which is even worse with multiple =
streams over the same UDP-port<o:p></o:p></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Further, defining the QoS RTP extension header as in =
RFC5285, does not in anyway conflict with other RTP extension headers or =
DSCP transfer or settings.<o:p></o:p></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">/Karl<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><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;">Fr=E5n:</span></b><span =
style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;"> Colin Perkins [<a =
href=3D"mailto:csp@csperkins.org">mailto:csp@csperkins.org</a>] =
<br><b>Skickat:</b> den 24 februari 2014 23:52<br><b>Till:</b> Karl =
Stahl<br><b>Kopia:</b> <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; Magnus Westerlund; =
<a href=3D"mailto:tram@ietf.org">tram@ietf.org</a>; Harald =
Alvestrand<br><b>=C4mne:</b> Re: [rtcweb] [tram] Payload Types =
assignments<o:p></o:p></span></p></div></div><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal">Karl,<o:p></o:p></p><div><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p></div><div><p =
class=3D"MsoNormal">I strongly disagree with this suggestion. An RTP =
header extension, located at an unknown and variable offset into a =
packet that does not have a well-defined magic number in the header, =
<span style=3D"color:blue"><o:p></o:p></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">This is how to find the traffic info:</span><span =
lang=3D"EN" style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;"><o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp; The payload of the classifier header extension =
element can be encoded<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp; using either the one-byte or two-byte header =
defined in [<a href=3D"http://tools.ietf.org/html/rfc5285" =
title=3D"&quot;A General Mechanism for RTP Header =
Extensions&quot;">rfc5285</a>] and<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp; shown below in Figure 1 and 2 =
below.<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 =
5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:=
p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; ID&nbsp;&nbsp; | len=3D1 =
|&nbsp;&nbsp; Namespace&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Value&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; 0 =
(pad)&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:=
p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; Figure 1: Classifier Using the One-Byte =
Header<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 =
5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:=
p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
ID&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
len=3D2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; =
Namespace&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Value&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:=
p></span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"page-break-before:always"><span lang=3D"EN" =
style=3D"font-size:12.0pt;font-family:&quot;Courier =
New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; Figure 2: Classifier Using the Two-Byte =
Header<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US">indicated using a dynamically assigned identifier that is =
conveyed in an out-of-band <span =
style=3D"color:blue"><o:p></o:p></span></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">--- It is the TURN flow!!! Requested by the ICE =
protocol for the specific media to come!<o:p></o:p></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US">and encrypted signalling =
channel, <span style=3D"color:blue"><o:p></o:p></span></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">--- Only the payload is encrypted in SRTP =96 leaving =
id, seq no and header ext to be used as =
intended!<o:p></o:p></span></p><p class=3D"MsoNormal"><span lang=3D"EN-US"=
 =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">--- Thus being the appropriate =
place=85<o:p></o:p></span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US">is not an appropriate place to put QoS information that =
has to be processed on a per-packet basis. <span =
style=3D"color:blue"><o:p></o:p></span></span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US">If you want DiffServ, you know =
where to find it.<o:p></o:p></span></p></div><div><p =
class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:blue">--- True, =
but if it cannot be used=85</span><span =
lang=3D"EN-US"><o:p></o:p></span></p></div><div><p =
class=3D"MsoNormal">Colin<o:p></o:p></p></div><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">On 24 Feb 2014, at 22:09, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<o:p></o:p></p></div><blockquote =
style=3D"margin-top:5.0pt;margin-bottom:5.0pt"><p =
class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">I suggest to the RTCWEB WG that the below from the =
September and October discussions on the relevant </span><span =
lang=3D"EN-US">[rtcweb] [avtext] [mmusic]</span><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue"> lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html"=
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
is introduced into </span><span lang=3D"EN-US">draft-ietf-rtcweb-rtp-usage=
 </span><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">for usage of RFC 5285, to =
allow:</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(1) WebRTC applications to directly convey QoS related =
real-time traffic info to the network at points where RTP flow is =
directed to by TRAM Milstone 3, to be used by *<b>any network element =
implementing any suitable QoS methods for the particular network</b>* =
for </span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(2) *<b>all</b>* WebRTC browsers *<b>and</b>* clients, =
under *<b>all</b>* OSs, and *<b>all</b>* current and future IP network, =
to achieve best QoE </span><o:p></o:p></p><p class=3D"MsoNormal"><b><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">(3) *without* </span></b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">having to force WebRTC into application specific =
networks (such as IMS) instead of using the Internet (including =
OTT).</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">The only further activity required, is to call for =
ISPs=92 to review whether the traffic information transferred by RFC =
5285 is sufficient for current and future needs in their network as =
suggested in below repeated <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html"=
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a></sp=
an><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&lt;snip&gt;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">=85two parameters (e.g. two bytes each) are encoded into the RTP =
header extension:</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">A) The maximum bandwidth requirement: Two bytes could contain =
everything from some bps for real-time text to Gbps for future 3D =
supersize telepresence=85 on a logarithmic =
scale.</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">B) The quality characteristics for the stream, with the highest =
bit set to 1, we could allocate a bit each for quality type =
e.g:</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay =
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =
Prioritize X, Variation Y, that could be combined as required to =
describe the stream.</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">And with highest bit set to 0, there could instead be a number for =
special usage that does not fit the general description of the =
individual bits.</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;">&lt;/snip&gt;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Then this could be assigned numbers to have an RFC in =
place.</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">With TRAM milestone 3 also place, =
</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">market forces will drive ISPs and browser makers to =
implement just this, without even having it =
MUST-established.</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">=93Who does not want a =93WebRTC-Ready=94 Internet =
access?=94 and</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">=93Who wants to use Chrome, if Firefox, Internet =
Explorer or Safari comes with much better QoE?=94 and vice =
versa.</span><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">Please see further emails soon following this one, for =
details and history.</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">/Karl</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&q=
uot;;color:blue">&nbsp;</span><o:p></o:p></p><div><div =
style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm"><p class=3D"MsoNormal"><b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;">Fr=E5n:</span></b><span lang=3D"EN-US" =
style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;"> <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>=
] <b>F=F6r </b>Karl Stahl<br><b>Skickat:</b> den 22 oktober 2013 =
16:37<br><b>Till:</b> 'Harald Alvestrand'; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; 'Magnus =
Westerlund'<br><b>Kopia:</b> 'Colin Perkins'<br><b>=C4mne:</b> [rtcweb] =
[avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p></div=
></div><p class=3D"MsoNormal"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">Harald, I mostly agree with the quality requirements of =
different real-time traffic that the WebRTC browser/application may use. =
But rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. &nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">There is actually a good =
way to show these needs to the network (without using the PT, or =
diffserv bits, which aren=92t sufficient anyway). =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">Let's use the RTP header extension field that also is =
visible outside the encrypted payload. A week ago came <a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">ht=
tp://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a> =
&nbsp;that outlines the usage of the extension field for classification =
of traffic! This document does not yet outline what to put in there and =
how to encode it though.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">Today's <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http://=
tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a> discusses other =
webrtc usages of the RTP header extension in 5.2 (there can be many =
header extensions according to RFC 5285) and in 9 there is "WebRTC Use =
of RTP: Future Extensions".</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">So, it looks obvious to use =
the RTP header extension to show the characteristics and bandwidth =
requirements to the network. It should not introduce any backward =
incompatibilities either.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">Such marking is done in =
every RTP packet so it can be set individually for each stream and could =
even be changed during a session (e.g. when limiting the bandwidth based =
on RTCP feedback). RFC 5286 also specifies how RTP extension header =
usage can be negotiated in SDP. I think this could be easily done by the =
WebRTC browser for "all current and future needs" if properly specified =
now.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">I suggest that two parameters (e.g. two bytes each) are =
encoded into the RTP header extension:</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">A) The maximum bandwidth =
requirement: Two bytes could contain everything from some bps for =
real-time text to Gbps for future 3D supersize telepresence=85 on a =
logarithmic scale.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">B) The quality characteristics for the stream, with the =
highest bit set to 1, we could allocate a bit each quality =
e.g:</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">Best Effort, Audio, Video, Supplemental Video, Gaming, =
Data, Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable =
Delivery, Prioritize X, Variation Y, that could be combined as required =
to describe the stream.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">And with highest bit set to =
0, there could instead be a number for special usage that does not fit =
the general description of the individual bits.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">Please note the totally =
different requirements a diffserv and an RSVP network have to know, so =
let=92s put all into these bytes. (E.g. a diffserv network don't need =
the bandwidth usage, but RSVP reservation networks (e.g. cable and 3G/4G =
OTT) do. There one should initially reserve the maximum bandwidth =
indicated, but can later re-reserve.)</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">/Karl</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">PS </span><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Microsoft=
 seems to have done work in this field, defining a proprietary attribute =
=93MS Service Quality=94; </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">However =
that seems to apply to the TURN server allocation request and would =
therefore:</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">--- =
Apply to the whole UDP flow, and could not be set for each stream =
individually (with different requirements), and</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">--- =
Does not handle the bandwidth requirement for incoming real-time traffic =
(required to reserve in RSVP type of networks)</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">However =
the quality attributes conveyed and their encoding may &nbsp;be =
considered.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</s=
pan><o:p></o:p></p><p class=3D"MsoNormal"><span lang=3D"EN-US">This is =
2.2.2.19 MS-Service Quality Attribute from </span><o:p></o:p></p><p =
class=3D"MsoNormal"><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).as=
px" =
title=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx">http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx<=
/a>&nbsp;<o:p></o:p></p><p class=3D"MsoNormal">&nbsp;<o:p></o:p></p><p =
class=3D"MsoNormal"><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">MS-Servic=
e Quality Attribute</span></em><o:p></o:p></p><p =
class=3D"MsoNormal"><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">The =
MS-Service Quality attribute is used to convey information about the =
data stream that the protocol client is intending to transfer over an =
allocated port. The protocol client SHOULD&lt;21&gt; include this =
attribute as part of an Allocate request message. A TURN server SHOULD =
use the information in this <span =
style=3D"background:yellow;mso-highlight:yellow">attribute to make =
decisions about resource allocation, bandwidth prioritization, and data =
delivery methods</span>. If the attribute is not present in the Allocate =
request message, the TURN server SHOULD assume that the data stream is =
audio with best effort delivery. The format of this attribute is as =
follows... </span></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">...</span=
></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">The =
following stream types are supported in this extension. All other stream =
types are reserved for future use.</span></em><o:p></o:p></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family:Symbol">=A7 =
</span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0001":=
 Audio</span></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span =
style=3D"font-family:Symbol">=A7 </span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0002":=
 Video</span></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span =
style=3D"font-family:Symbol">=A7 </span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0003":=
 Supplemental Video</span></em><o:p></o:p></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family:Symbol">=A7 =
</span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0004":=
 Data</span></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span =
lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Service =
Quality (2 bytes): The service quality level required by the protocol =
client for the stream.</span></em><o:p></o:p></p><p =
class=3D"MsoNormal"><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">The =
following service quality levels are supported in this extension. All =
other service quality levels are reserved for future =
use.</span></em><o:p></o:p></p><p class=3D"MsoNormal"><em><span =
style=3D"font-family:Symbol">=A7 </span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0000":=
 Best effort delivery.</span></em><o:p></o:p></p><p =
class=3D"MsoNormal"><em><span style=3D"font-family:Symbol">=A7 =
</span></em><em><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">"0x0001":=
 Reliable delivery.</span></em><o:p></o:p></p><p class=3D"MsoNormal"><span=
 lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText">-----Ursprungligt =
meddelande-----<o:p></o:p></p><p class=3D"MsoPlainText">Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>=
] F=F6r Harald Alvestrand<o:p></o:p></p><p class=3D"MsoPlainText">Skickat:=
 den 8 oktober 2013 13:01<o:p></o:p></p><p class=3D"MsoPlainText">Till: =
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">=C4mne: Re: [rtcweb] Payload =
Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">On 10/08/2013 09:17 AM, =
Karl Stahl wrote:</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt; Hej Magnus,</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; Also, are you =
really interested in knowing that it is VP9 vs H.264, =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;&gt; isn't</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; the questions this is =
video of this priority that is important?</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; I think you need to =
more carefully consider what are the goals you </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt;&gt; try =
to</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt; achieve them.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; Actually, my concern is =
to get an idea of the maximum bandwidth that </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; could be required for a =
WebRTC (ICE) setup media flow. Both voice and </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; video should be =
prioritized over data (their individual priority is of =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
less importance as long as there is sufficient bandwidth for =
both).</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">You don't know that without knowing what the application =
is for.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">In, for instance, a shooter game with voice backchannels, =
the movement and event information (data) is MORE time sensitive than =
the voice data.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">&gt;&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; With diffserv you don=92t=
 need to know the bandwidth requirement, but </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; with RSVP reservation =
(like in cable and mobile networks) you need to </span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; know how much to =
reserve. Voice is like 100's kbit/s, video VP8 or =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
H.264 is like 3,5 mbps.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p=
 class=3D"MsoPlainText"><span lang=3D"EN-US">Again, without knowing the =
application, you don't know that.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">The application could decide =
to use QCIF or HD, and the bandwidth variation of screencast =
(semi-static with sudden, large changes) is completely different from =
that of a talking head, which is again completely different from a =
high-movement scene.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">&gt;&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; To add to the =
complication of codec variants, the video codecs in =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
question for WebRTC have variable bandwidth, and when there is a poor =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
connection we see Chrome reducing the video window size to reduce the =
bandwidth used...</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&gt;&nbsp;</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; I think the payload =
type field at best can reflect a maximum bandwidth =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
to initially reserve bandwidth for, and thereafter make new =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
reservations if the bandwidth changes during the call. So could we =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
change RTP to show maximum bandwidth instead of payload type in that =
</span><o:p></o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">&gt; =
field outside the encrypted payload :) ... Or maybe that is not a =
joke?</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">I think these ruminations only lead to one =
conclusion:</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">You can't tell what the needed bandwidth is up front =
without asking the application.</span><o:p></o:p></p><p =
class=3D"MsoPlainText"><span lang=3D"EN-US">You can't tell what the =
right priority ranking is without asking the =
application.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 lang=3D"EN-US">If you need to know the bandwidth or the priority up =
front, the application has to tell you. Anything else is pure =
heuristics.</span><o:p></o:p></p><p class=3D"MsoPlainText"><span =
lang=3D"EN-US">&nbsp;</span><o:p></o:p></p><p class=3D"MsoPlainText"><span=
 =
lang=3D"EN-US">_______________________________________________</span><o:p>=
</o:p></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">rtcweb mailing =
list</span><o:p></o:p></p><p class=3D"MsoPlainText"><a =
href=3D"mailto:rtcweb@ietf.org"><span =
lang=3D"EN-US">rtcweb@ietf.org</span></a><o:p></o:p></p><p =
class=3D"MsoPlainText"><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span =
lang=3D"EN-US">https://www.ietf.org/mailman/listinfo/rtcweb</span></a><o:p=
></o:p></p></blockquote></div><p class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">&nbsp;</span></p><div><div><p =
class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">&nbsp;</span></p></div><div><p =
class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">--&nbsp;<o:p></o:p></span></p></div><div><p=
 class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">Colin =
Perkins<o:p></o:p></span></p></div><div><p class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;"><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a><o:p></o:p></span>=
</p></div><div><p class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">&nbsp;</span></p></div><p =
class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">&nbsp;</span></p></div><p =
class=3D"MsoNormal"><span =
style=3D"font-size:12.0pt;font-family:&quot;Times New =
Roman&quot;,&quot;serif&quot;">&nbsp;</span></p></div></div></div></blockq=
uote></div><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div><div><br></d=
iv></span><br class=3D"Apple-interchange-newline">

</div>
<br></div></body></html>=

--Apple-Mail=_B2EC0CF9-4C12-4B53-B965-573E2AE51D13--


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>, IESG IESG <iesg@ietf.org>
Thread-Topic: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:

> To the TRAM WG and RTCWEB WG and ADs:
> =20
> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed an=
d encouraged to route quality demanding WebRTC media into their IP pipes th=
at are capable of transporting real-time traffic without quality issues, us=
ing TURN servers.
> =20

Reading the charter, the above is *not* at all a clear objective of the WG =
(note I am not the chair of this WG or the responsible AD).=20

That said, I think you have pointed out this charter is abysmally vague - i=
t does not say what the WG is not going to do. If I decided to do BGP for r=
outing updates over TURN it would be within the scope of this charter.=20

My advice to the responsible AD is recharter this WG before IETF 90 or clos=
e it. I would be glad to help write a charter that is not an infinite blank=
 cheque.=20







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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>, IESG IESG <iesg@ietf.org>
Thread-Topic: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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resend because the IETF mail lists said I had to many people in the to line=
=20


On Feb 26, 2014, at 9:05 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>=20
>=20
> On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:
>=20
>> To the TRAM WG and RTCWEB WG and ADs:
>>=20
>> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed a=
nd encouraged to route quality demanding WebRTC media into their IP pipes t=
hat are capable of transporting real-time traffic without quality issues, u=
sing TURN servers.
>>=20
>=20
> Reading the charter, the above is *not* at all a clear objective of the W=
G (note I am not the chair of this WG or the responsible AD).=20
>=20
> That said, I think you have pointed out this charter is abysmally vague -=
 it does not say what the WG is not going to do. If I decided to do BGP for=
 routing updates over TURN it would be within the scope of this charter.=20
>=20
> My advice to the responsible AD is recharter this WG before IETF 90 or cl=
ose it. I would be glad to help write a charter that is not an infinite bla=
nk cheque.=20
>=20
>=20
>=20
>=20
>=20
>=20


From nobody Tue Feb 25 17:08:40 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>, IESG IESG <iesg@ietf.org>
Thread-Topic: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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resend because the IETF mail lists said I had to many people in the to line=
=20


On Feb 26, 2014, at 9:05 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>=20
>=20
> On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:
>=20
>> To the TRAM WG and RTCWEB WG and ADs:
>>=20
>> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed a=
nd encouraged to route quality demanding WebRTC media into their IP pipes t=
hat are capable of transporting real-time traffic without quality issues, u=
sing TURN servers.
>>=20
>=20
> Reading the charter, the above is *not* at all a clear objective of the W=
G (note I am not the chair of this WG or the responsible AD).=20
>=20
> That said, I think you have pointed out this charter is abysmally vague -=
 it does not say what the WG is not going to do. If I decided to do BGP for=
 routing updates over TURN it would be within the scope of this charter.=20
>=20
> My advice to the responsible AD is recharter this WG before IETF 90 or cl=
ose it. I would be glad to help write a charter that is not an infinite bla=
nk cheque.=20
>=20
>=20
>=20
>=20
>=20
>=20


From nobody Tue Feb 25 19:49:11 2014
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Subject: Re: [rtcweb] [tram] Payload Types assignments
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Karl,

what on Earth does TRAM milestone 3 have to do with whatever it is
you're proposing?

TRAM milestone 3 is:

Sep 2014 - Send new proposed standard TURN server discovery mechanism
for enterprises and ISPs to IESG

There's no (zero, nada, none) mention of QoS within that charter.


On 02/26/2014 12:49 AM, Colin Perkins wrote:
> Karl,
>
> I sympathise with your goal, but I really do not think RTP header
> extensions are appropriate for this purpose.
>
> Ignoring the semantic mismatch, in order to identify an RTP header
> extension, you need access to the signalling data, and if you have
> access to the signalling, you can signal the QoS parameters without
> using RTP. 
>
> You state that only the payload is encrypted in SRTP. That is not
> necessarily true. RTP header extensions can be encrypted when RFC 6904
> is in use, and rtcweb-rtp-usage draft recommends this be done in some
> cases. The signalling channel is also likely encrypted end-to-end in
> new applications, so making it difficult to extract the information
> you need to parse the RTP header extension, even if it is unencrypted. 
>
> Besides these focussed issues, I would also urge you to consider the
> much broader comments Magnus made. The QoS problem is broad, and a
> point solution based on RTP header extensions - even if it were
> workable, which I doubt - would address only a small part of the
> problem space.
>
> Colin
>
>
>
>
> On 25 Feb 2014, at 01:45, Karl Stahl <karl.stahl@intertex.se
> <mailto:karl.stahl@intertex.se>> wrote:
>>
>> Colin,
>>
>>  
>>
>> If the below were the case, it would be “DPI guesswork” that I also
>> advice against. RTP doesn’t even have unique protocol header within
>> UDP – it can even be confused with other UDP payload.
>>
>>  
>>
>> However, if the RTP is captured in a TURN-flow, as in TRAM Milestone
>> 3, the network point that this flow is directed to and can apply QoS
>> methods relevant to the network (which is not diffserve in Mobile OTT
>> and Cable Networks) has not a too difficult tasks. Linking each
>> RTP-flow by its ID and sequence number, and picking exactly the right
>> traffic type and bandwidth parameters is doable (see inline below, we
>> at Ingate do it already)!
>>
>>  
>>
>> Also DiffServ DSCP-bits are seldom maintained crossing network
>> boundaries, thus carrying no relevant information at the receiving
>> end, while the RTP extension header remains unchanged end-to-end. In
>> DSCP-bits, there is no bandwidth requirement information when
>> entering networks requiring reservation. That is always (and
>> dynamically set) available in the RTP extension header for each packet.
>>
>>  
>>
>> And, I am sure you are aware of the difficulties of getting DSCP-bits
>> through OS sockets, which is even worse with multiple streams over
>> the same UDP-port
>>
>>  
>>
>> Further, defining the QoS RTP extension header as in RFC5285, does
>> not in anyway conflict with other RTP extension headers or DSCP
>> transfer or settings.
>>
>>  
>>
>> /Karl
>>
>>  
>>
>> *Från:*Colin Perkins [mailto:csp@csperkins.org]
>> *Skickat:* den 24 februari 2014 23:52
>> *Till:* Karl Stahl
>> *Kopia:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>; Magnus Westerlund;
>> tram@ietf.org <mailto:tram@ietf.org>; Harald Alvestrand
>> *Ämne:* Re: [rtcweb] [tram] Payload Types assignments
>>
>>  
>>
>> Karl,
>>
>>  
>>
>> I strongly disagree with this suggestion. An RTP header extension,
>> located at an unknown and variable offset into a packet that does not
>> have a well-defined magic number in the header,
>>
>> This is how to find the traffic info:
>>
>>    The payload of the classifier header extension element can be encoded
>>
>>    using either the one-byte or two-byte header defined in [rfc5285
>> <http://tools.ietf.org/html/rfc5285>] and
>>
>>    shown below in Figure 1 and 2 below.
>>
>>  
>>
>>       0                   1                   2                   3
>>
>>       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>>
>>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>
>>      |  ID   | len=1 |   Namespace   |    Value      |    0 (pad)    |
>>
>>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>
>>  
>>
>>              Figure 1: Classifier Using the One-Byte Header
>>
>>  
>>
>>       0                   1                   2                   3
>>
>>       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>>
>>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>
>>      |      ID       |    len=2      |   Namespace   |    Value      |
>>
>>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>>
>>  
>>
>>  
>>
>>              Figure 2: Classifier Using the Two-Byte Header
>>
>>  
>>
>> indicated using a dynamically assigned identifier that is conveyed in
>> an out-of-band
>>
>> --- It is the TURN flow!!! Requested by the ICE protocol for the
>> specific media to come!
>>
>> and encrypted signalling channel,
>>
>> --- Only the payload is encrypted in SRTP – leaving id, seq no and
>> header ext to be used as intended!
>>
>> --- Thus being the appropriate place…
>>
>> is not an appropriate place to put QoS information that has to be
>> processed on a per-packet basis.
>>
>> If you want DiffServ, you know where to find it.
>>
>> --- True, but if it cannot be used…
>>
>> Colin
>>
>>  
>>
>>  
>>
>> On 24 Feb 2014, at 22:09, Karl Stahl <karl.stahl@intertex.se
>> <mailto:karl.stahl@intertex.se>> wrote:
>>
>>     I suggest to the RTCWEB WG that the below from the September and
>>     October discussions on the relevant [rtcweb] [avtext]
>>     [mmusic]lists
>>     http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>>     is introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC
>>     5285, to allow:
>>
>>      
>>
>>     (1) WebRTC applications to directly convey QoS related real-time
>>     traffic info to the network at points where RTP flow is directed
>>     to by TRAM Milstone 3, to be used by **any network element
>>     implementing any suitable QoS methods for the particular
>>     network** for
>>
>>     (2) **all** WebRTC browsers **and** clients, under **all** OSs,
>>     and **all** current and future IP network, to achieve best QoE
>>
>>     *(3) *without* *having to force WebRTC into application specific
>>     networks (such as IMS) instead of using the Internet (including OTT).
>>
>>      
>>
>>     The only further activity required, is to call for ISPs’ to
>>     review whether the traffic information transferred by RFC 5285 is
>>     sufficient for current and future needs in their network as
>>     suggested in below repeated
>>     http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>>
>>     <snip>
>>
>>     …two parameters (e.g. two bytes each) are encoded into the RTP
>>     header extension:
>>
>>      
>>
>>     A) The maximum bandwidth requirement: Two bytes could contain
>>     everything from some bps for real-time text to Gbps for future 3D
>>     supersize telepresence… on a logarithmic scale.
>>
>>      
>>
>>     B) The quality characteristics for the stream, with the highest
>>     bit set to 1, we could allocate a bit each for quality type e.g:
>>
>>     Best Effort, Audio, Video, Supplemental Video, Gaming, Data,
>>     Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable
>>     Delivery, Prioritize X, Variation Y, that could be combined as
>>     required to describe the stream.
>>
>>      
>>
>>     And with highest bit set to 0, there could instead be a number
>>     for special usage that does not fit the general description of
>>     the individual bits.
>>
>>     </snip>
>>
>>      
>>
>>     Then this could be assigned numbers to have an RFC in place.
>>
>>      
>>
>>     With TRAM milestone 3 also place,
>>
>>     market forces will drive ISPs and browser makers to implement
>>     just this, without even having it MUST-established.
>>
>>      
>>
>>     “Who does not want a “WebRTC-Ready” Internet access?” and
>>
>>     “Who wants to use Chrome, if Firefox, Internet Explorer or Safari
>>     comes with much better QoE?” and vice versa.
>>
>>      
>>
>>     Please see further emails soon following this one, for details
>>     and history.
>>
>>      
>>
>>     /Karl
>>
>>      
>>
>>      
>>
>>     *Från:*rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>>     [mailto:rtcweb-bounces@ietf.org] *För *Karl Stahl
>>     *Skickat:* den 22 oktober 2013 16:37
>>     *Till:* 'Harald Alvestrand'; rtcweb@ietf.org
>>     <mailto:rtcweb@ietf.org>; 'Magnus Westerlund'
>>     *Kopia:* 'Colin Perkins'
>>     *Ämne:* [rtcweb] [avtext] Payload Types assignments was Re: SV:
>>     [mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
>>
>>      
>>
>>     Harald, I mostly agree with the quality requirements of different
>>     real-time traffic that the WebRTC browser/application may use.
>>     But rather than asking the application, let's convey the
>>     bandwidth and priority requirements to the network. Just like
>>     with the Payload type (that is hard to squeeze that information
>>     into) it must be visible to the network (and not changed by the
>>     network, like diffserv bits are). Such marking must also be
>>     available for incoming traffic, which is especially important in
>>     RSVP type of networks, that has to reserve bandwidth for it.  
>>
>>      
>>
>>     There is actually a good way to show these needs to the network
>>     (without using the PT, or diffserv bits, which aren’t sufficient
>>     anyway).
>>
>>      
>>
>>     Let's use the RTP header extension field that also is visible
>>     outside the encrypted payload. A week ago came
>>     http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00
>>      that outlines the usage of the extension field for
>>     classification of traffic! This document does not yet outline
>>     what to put in there and how to encode it though.
>>
>>      
>>
>>     Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10
>>     discusses other webrtc usages of the RTP header extension in 5.2
>>     (there can be many header extensions according to RFC 5285) and
>>     in 9 there is "WebRTC Use of RTP: Future Extensions".
>>
>>      
>>
>>     So, it looks obvious to use the RTP header extension to show the
>>     characteristics and bandwidth requirements to the network. It
>>     should not introduce any backward incompatibilities either.
>>
>>      
>>
>>     Such marking is done in every RTP packet so it can be set
>>     individually for each stream and could even be changed during a
>>     session (e.g. when limiting the bandwidth based on RTCP
>>     feedback). RFC 5286 also specifies how RTP extension header usage
>>     can be negotiated in SDP. I think this could be easily done by
>>     the WebRTC browser for "all current and future needs" if properly
>>     specified now.
>>
>>      
>>
>>     I suggest that two parameters (e.g. two bytes each) are encoded
>>     into the RTP header extension:
>>
>>      
>>
>>     A) The maximum bandwidth requirement: Two bytes could contain
>>     everything from some bps for real-time text to Gbps for future 3D
>>     supersize telepresence… on a logarithmic scale.
>>
>>      
>>
>>     B) The quality characteristics for the stream, with the highest
>>     bit set to 1, we could allocate a bit each quality e.g:
>>
>>     Best Effort, Audio, Video, Supplemental Video, Gaming, Data,
>>     Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable
>>     Delivery, Prioritize X, Variation Y, that could be combined as
>>     required to describe the stream.
>>
>>      
>>
>>     And with highest bit set to 0, there could instead be a number
>>     for special usage that does not fit the general description of
>>     the individual bits.
>>
>>      
>>
>>     Please note the totally different requirements a diffserv and an
>>     RSVP network have to know, so let’s put all into these bytes.
>>     (E.g. a diffserv network don't need the bandwidth usage, but RSVP
>>     reservation networks (e.g. cable and 3G/4G OTT) do. There one
>>     should initially reserve the maximum bandwidth indicated, but can
>>     later re-reserve.)
>>
>>      
>>
>>     /Karl
>>
>>      
>>
>>     PS Microsoft seems to have done work in this field, defining a
>>     proprietary attribute “MS Service Quality”;
>>
>>     However that seems to apply to the TURN server allocation request
>>     and would therefore:
>>
>>     --- Apply to the whole UDP flow, and could not be set for each
>>     stream individually (with different requirements), and
>>
>>     --- Does not handle the bandwidth requirement for incoming
>>     real-time traffic (required to reserve in RSVP type of networks)
>>
>>     However the quality attributes conveyed and their encoding may
>>      be considered.
>>
>>      
>>
>>     This is 2.2.2.19 MS-Service Quality Attribute from
>>
>>     http://msdn.microsoft.com/en-us/library/cc431507(v=office.12).aspx <http://msdn.microsoft.com/en-us/library/cc431507%28v=office.12%29.aspx> 
>>
>>      
>>
>>     /MS-Service Quality Attribute/
>>
>>     /The MS-Service Quality attribute is used to convey information
>>     about the data stream that the protocol client is intending to
>>     transfer over an allocated port. The protocol client SHOULD<21>
>>     include this attribute as part of an Allocate request message. A
>>     TURN server SHOULD use the information in this attribute to make
>>     decisions about resource allocation, bandwidth prioritization,
>>     and data delivery methods. If the attribute is not present in the
>>     Allocate request message, the TURN server SHOULD assume that the
>>     data stream is audio with best effort delivery. The format of
>>     this attribute is as follows... /
>>
>>     /.../
>>
>>     /The following stream types are supported in this extension. All
>>     other stream types are reserved for future use./
>>
>>     /§ //"0x0001": Audio/
>>
>>     /§ //"0x0002": Video/
>>
>>     /§ //"0x0003": Supplemental Video/
>>
>>     /§ //"0x0004": Data/
>>
>>     /Service Quality (2 bytes): The service quality level required by
>>     the protocol client for the stream./
>>
>>     /The following service quality levels are supported in this
>>     extension. All other service quality levels are reserved for
>>     future use./
>>
>>     /§ //"0x0000": Best effort delivery./
>>
>>     /§ //"0x0001": Reliable delivery./
>>
>>      
>>
>>      
>>
>>     -----Ursprungligt meddelande-----
>>
>>     Från: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>>     [mailto:rtcweb-bounces@ietf.org] För Harald Alvestrand
>>
>>     Skickat: den 8 oktober 2013 13:01
>>
>>     Till: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>
>>     Ämne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic]
>>     WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
>>
>>      
>>
>>     On 10/08/2013 09:17 AM, Karl Stahl wrote:
>>
>>     > Hej Magnus,
>>
>>     > 
>>
>>     >> Also, are you really interested in knowing that it is VP9 vs H.264,
>>
>>     >> isn't
>>
>>     > the questions this is video of this priority that is important?
>>
>>     >> I think you need to more carefully consider what are the goals you
>>
>>     >> try to
>>
>>     > achieve them.
>>
>>     > 
>>
>>     > Actually, my concern is to get an idea of the maximum bandwidth
>>     that
>>
>>     > could be required for a WebRTC (ICE) setup media flow. Both
>>     voice and
>>
>>     > video should be prioritized over data (their individual priority
>>     is of
>>
>>     > less importance as long as there is sufficient bandwidth for both).
>>
>>      
>>
>>     You don't know that without knowing what the application is for.
>>
>>     In, for instance, a shooter game with voice backchannels, the
>>     movement and event information (data) is MORE time sensitive than
>>     the voice data.
>>
>>      
>>
>>     > 
>>
>>     > With diffserv you don’t need to know the bandwidth requirement, but
>>
>>     > with RSVP reservation (like in cable and mobile networks) you
>>     need to
>>
>>     > know how much to reserve. Voice is like 100's kbit/s, video VP8 or
>>
>>     > H.264 is like 3,5 mbps.
>>
>>      
>>
>>     Again, without knowing the application, you don't know that.
>>
>>     The application could decide to use QCIF or HD, and the bandwidth
>>     variation of screencast (semi-static with sudden, large changes)
>>     is completely different from that of a talking head, which is
>>     again completely different from a high-movement scene.
>>
>>      
>>
>>     > 
>>
>>     > To add to the complication of codec variants, the video codecs in
>>
>>     > question for WebRTC have variable bandwidth, and when there is a
>>     poor
>>
>>     > connection we see Chrome reducing the video window size to
>>     reduce the bandwidth used...
>>
>>     > 
>>
>>     > I think the payload type field at best can reflect a maximum
>>     bandwidth
>>
>>     > to initially reserve bandwidth for, and thereafter make new
>>
>>     > reservations if the bandwidth changes during the call. So could we
>>
>>     > change RTP to show maximum bandwidth instead of payload type in
>>     that
>>
>>     > field outside the encrypted payload :) ... Or maybe that is not
>>     a joke?
>>
>>      
>>
>>     I think these ruminations only lead to one conclusion:
>>
>>      
>>
>>     You can't tell what the needed bandwidth is up front without
>>     asking the application.
>>
>>     You can't tell what the right priority ranking is without asking
>>     the application.
>>
>>      
>>
>>     If you need to know the bandwidth or the priority up front, the
>>     application has to tell you. Anything else is pure heuristics.
>>
>>      
>>
>>     _______________________________________________
>>
>>     rtcweb mailing list
>>
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>  
>>
>>  
>>
>> -- 
>>
>> Colin Perkins
>>
>> http://csperkins.org/
>>
>>  
>>
>>  
>>
>>  
>>
>
>
>
> -- 
> Colin Perkins
> http://csperkins.org/
>
>
>


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">Karl,<br>
      <br>
      what on Earth does TRAM milestone 3 have to do with whatever it is
      you're proposing?<br>
      <br>
      TRAM milestone 3 is: <br>
      <br>
      Sep 2014 - Send new proposed standard TURN server discovery
      mechanism for enterprises and ISPs to IESG<br>
      <br>
      There's no (zero, nada, none) mention of QoS within that charter.<br>
      <br>
      <br>
      On 02/26/2014 12:49 AM, Colin Perkins wrote:<br>
    </div>
    <blockquote
      cite="mid:D06C438A-8894-402C-AE9F-D7787ECF77B3@csperkins.org"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      Karl,
      <div><br>
      </div>
      <div>I sympathise with your goal, but I really do not think RTP
        header extensions are appropriate for this purpose.</div>
      <div><br>
      </div>
      <div>Ignoring the semantic mismatch, in order to identify an RTP
        header extension, you need access to the signalling data, and if
        you have access to the signalling, you can signal the QoS
        parameters without using RTP. </div>
      <div><br>
      </div>
      <div>You state that only the payload is encrypted in SRTP. That is
        not necessarily true. RTP header extensions can be encrypted
        when RFC 6904 is in use, and rtcweb-rtp-usage draft recommends
        this be done in some cases. The signalling channel is also
        likely encrypted end-to-end in new applications, so making it
        difficult to extract the information you need to parse the RTP
        header extension, even if it is unencrypted. </div>
      <div><br>
      </div>
      <div>Besides these focussed issues, I would also urge you to
        consider the much broader comments Magnus made. The QoS problem
        is broad, and a point solution based on RTP header extensions -
        even if it were workable, which I doubt - would address only a
        small part of the problem space.</div>
      <div><br>
      </div>
      <div>Colin</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
        <div>
          <div>On 25 Feb 2014, at 01:45, Karl Stahl &lt;<a
              moz-do-not-send="true"
              href="mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt;
            wrote:</div>
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                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">Colin,<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">If the below were the case, it would be
                    “DPI guesswork” that I also advice against. RTP
                    doesn’t even have unique protocol header within UDP
                    – it can even be confused with other UDP payload. <o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">However, if the RTP is captured in a
                    TURN-flow, as in TRAM Milestone 3, the network point
                    that this flow is directed to and can apply QoS
                    methods relevant to the network (which is not
                    diffserve in Mobile OTT and Cable Networks) has not
                    a too difficult tasks. Linking each RTP-flow by its
                    ID and sequence number, and picking exactly the
                    right traffic type and bandwidth parameters is
                    doable (see inline below, we at Ingate do it
                    already)!<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">Also DiffServ DSCP-bits are seldom
                    maintained crossing network boundaries, thus
                    carrying no relevant information at the receiving
                    end, while the RTP extension header remains
                    unchanged end-to-end. In DSCP-bits, there is no
                    bandwidth requirement information when entering
                    networks requiring reservation. That is always (and
                    dynamically set) available in the RTP extension
                    header for each packet.<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">And, I am sure you are aware of the
                    difficulties of getting DSCP-bits through OS
                    sockets, which is even worse with multiple streams
                    over the same UDP-port<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">Further, defining the QoS RTP extension
                    header as in RFC5285, does not in anyway conflict
                    with other RTP extension headers or DSCP transfer or
                    settings.<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US">/Karl<o:p></o:p></span></p>
                <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                    lang="EN-US"> </span></p>
                <div>
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                    <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Från:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                        Colin Perkins [<a moz-do-not-send="true"
                          href="mailto:csp@csperkins.org">mailto:csp@csperkins.org</a>]
                        <br>
                        <b>Skickat:</b> den 24 februari 2014 23:52<br>
                        <b>Till:</b> Karl Stahl<br>
                        <b>Kopia:</b> <a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>;
                        Magnus Westerlund; <a moz-do-not-send="true"
                          href="mailto:tram@ietf.org">tram@ietf.org</a>;
                        Harald Alvestrand<br>
                        <b>Ämne:</b> Re: [rtcweb] [tram] Payload Types
                        assignments<o:p></o:p></span></p>
                  </div>
                </div>
                <p class="MsoNormal"><o:p> </o:p></p>
                <p class="MsoNormal">Karl,<o:p></o:p></p>
                <div>
                  <p class="MsoNormal"><o:p> </o:p></p>
                </div>
                <div>
                  <p class="MsoNormal">I strongly disagree with this
                    suggestion. An RTP header extension, located at an
                    unknown and variable offset into a packet that does
                    not have a well-defined magic number in the header,
                    <span style="color:blue"><o:p></o:p></span></p>
                  <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                      lang="EN-US">This is how to find the traffic info:</span><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"><o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">   The payload of the
                      classifier header extension element can be encoded<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">   using either the one-byte
                      or two-byte header defined in [<a
                        moz-do-not-send="true"
                        href="http://tools.ietf.org/html/rfc5285"
                        title="&quot;A General Mechanism for RTP Header
                        Extensions&quot;">rfc5285</a>] and<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">   shown below in Figure 1
                      and 2 below.<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"> </span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">      0                  
                      1                   2                   3<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">      0 1 2 3 4 5 6 7 8 9 0 1
                      2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">    
                      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">     |  ID   | len=1 |  
                      Namespace   |    Value      |    0 (pad)    |<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">    
                      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"> </span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">             Figure 1:
                      Classifier Using the One-Byte Header<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"> </span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">      0    
                                    1                  
                      2                   3<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">      0 1 2 3 4 5 6 7 8 9 0 1
                      2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">    
                      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">     |      ID       |   
                      len=2      |   Namespace   |    Value      |<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">    
                      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"> </span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN"> </span></p>
                  <p class="MsoNormal" style="page-break-before:always"><span
                      style="font-size:12.0pt;font-family:&quot;Courier
                      New&quot;" lang="EN">             Figure 2:
                      Classifier Using the Two-Byte Header<o:p></o:p></span></p>
                  <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                      lang="EN-US"> </span></p>
                  <p class="MsoNormal"><span lang="EN-US">indicated
                      using a dynamically assigned identifier that is
                      conveyed in an out-of-band <span
                        style="color:blue"><o:p></o:p></span></span></p>
                  <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                      lang="EN-US">--- It is the TURN flow!!! Requested
                      by the ICE protocol for the specific media to
                      come!<o:p></o:p></span></p>
                  <p class="MsoNormal"><span lang="EN-US">and encrypted
                      signalling channel, <span style="color:blue"><o:p></o:p></span></span></p>
                  <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                      lang="EN-US">--- Only the payload is encrypted in
                      SRTP – leaving id, seq no and header ext to be
                      used as intended!<o:p></o:p></span></p>
                  <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                      lang="EN-US">--- Thus being the appropriate place…<o:p></o:p></span></p>
                  <p class="MsoNormal"><span lang="EN-US">is not an
                      appropriate place to put QoS information that has
                      to be processed on a per-packet basis. <span
                        style="color:blue"><o:p></o:p></span></span></p>
                  <p class="MsoNormal"><span lang="EN-US">If you want
                      DiffServ, you know where to find it.<o:p></o:p></span></p>
                </div>
                <div>
                  <p class="MsoNormal"><span style="color:blue"
                      lang="EN-US">--- True, but if it cannot be used…</span><span
                      lang="EN-US"><o:p></o:p></span></p>
                </div>
                <div>
                  <p class="MsoNormal">Colin<o:p></o:p></p>
                </div>
                <div>
                  <p class="MsoNormal"><o:p> </o:p></p>
                </div>
                <div>
                  <p class="MsoNormal"><o:p> </o:p></p>
                  <div>
                    <div>
                      <p class="MsoNormal">On 24 Feb 2014, at 22:09,
                        Karl Stahl &lt;<a moz-do-not-send="true"
                          href="mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt;
                        wrote:<o:p></o:p></p>
                    </div>
                    <blockquote
                      style="margin-top:5.0pt;margin-bottom:5.0pt">
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">I suggest to the RTCWEB WG that
                          the below from the September and October
                          discussions on the relevant </span><span
                          lang="EN-US">[rtcweb] [avtext] [mmusic]</span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> lists <a moz-do-not-send="true"
href="http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a>
                          is introduced into </span><span lang="EN-US">draft-ietf-rtcweb-rtp-usage
                        </span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">for usage of RFC 5285, to allow:</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">(1) WebRTC applications to
                          directly convey QoS related real-time traffic
                          info to the network at points where RTP flow
                          is directed to by TRAM Milstone 3, to be used
                          by *<b>any network element implementing any
                            suitable QoS methods for the particular
                            network</b>* for </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">(2) *<b>all</b>* WebRTC browsers
                          *<b>and</b>* clients, under *<b>all</b>* OSs,
                          and *<b>all</b>* current and future IP
                          network, to achieve best QoE </span><o:p></o:p></p>
                      <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                            lang="EN-US">(3) *without* </span></b><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">having to force WebRTC into
                          application specific networks (such as IMS)
                          instead of using the Internet (including OTT).</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">The only further activity
                          required, is to call for ISPs’ to review
                          whether the traffic information transferred by
                          RFC 5285 is sufficient for current and future
                          needs in their network as suggested in below
                          repeated <a moz-do-not-send="true"
                            href="http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a></span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">&lt;snip&gt;</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">…two parameters (e.g. two bytes
                          each) are encoded into the RTP header
                          extension:</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">A) The maximum bandwidth
                          requirement: Two bytes could contain
                          everything from some bps for real-time text to
                          Gbps for future 3D supersize telepresence… on
                          a logarithmic scale.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">B) The quality characteristics
                          for the stream, with the highest bit set to 1,
                          we could allocate a bit each for quality type
                          e.g:</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">Best Effort, Audio, Video,
                          Supplemental Video, Gaming, Data, Delay
                          Insensitive (e.g. video streaming), Minimum
                          Delay, Reliable Delivery, Prioritize X,
                          Variation Y, that could be combined as
                          required to describe the stream.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">And with highest bit set to 0,
                          there could instead be a number for special
                          usage that does not fit the general
                          description of the individual bits.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">&lt;/snip&gt;</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">Then this could be assigned
                          numbers to have an RFC in place.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">With TRAM milestone 3 also place,
                        </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">market forces will drive ISPs and
                          browser makers to implement just this, without
                          even having it MUST-established.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">“Who does not want a
                          “WebRTC-Ready” Internet access?” and</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">“Who wants to use Chrome, if
                          Firefox, Internet Explorer or Safari comes
                          with much better QoE?” and vice versa.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">Please see further emails soon
                          following this one, for details and history.</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US">/Karl</span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <div>
                        <div style="border:none;border-top:solid #B5C4DF
                          1.0pt;padding:3.0pt 0cm 0cm 0cm">
                          <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"
                                lang="EN-US">Från:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"
                              lang="EN-US"> <a moz-do-not-send="true"
                                href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                              [<a moz-do-not-send="true"
                                href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                              <b>För </b>Karl Stahl<br>
                              <b>Skickat:</b> den 22 oktober 2013 16:37<br>
                              <b>Till:</b> 'Harald Alvestrand'; <a
                                moz-do-not-send="true"
                                href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>;
                              'Magnus Westerlund'<br>
                              <b>Kopia:</b> 'Colin Perkins'<br>
                              <b>Ämne:</b> [rtcweb] [avtext] Payload
                              Types assignments was Re: SV: [mmusic]
                              WGLC of
                              draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p>
                        </div>
                      </div>
                      <p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Harald,
                          I mostly agree with the quality requirements
                          of different real-time traffic that the WebRTC
                          browser/application may use. But rather than
                          asking the application, let's convey the
                          bandwidth and priority requirements to the
                          network. Just like with the Payload type (that
                          is hard to squeeze that information into) it
                          must be visible to the network (and not
                          changed by the network, like diffserv bits
                          are). Such marking must also be available for
                          incoming traffic, which is especially
                          important in RSVP type of networks, that has
                          to reserve bandwidth for it.  </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">There
                          is actually a good way to show these needs to
                          the network (without using the PT, or diffserv
                          bits, which aren’t sufficient anyway). </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Let's
                          use the RTP header extension field that also
                          is visible outside the encrypted payload. A
                          week ago came <a moz-do-not-send="true"
                            href="http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a>
                           that outlines the usage of the extension
                          field for classification of traffic! This
                          document does not yet outline what to put in
                          there and how to encode it though.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Today's
                          <a moz-do-not-send="true"
                            href="http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a>
                          discusses other webrtc usages of the RTP
                          header extension in 5.2 (there can be many
                          header extensions according to RFC 5285) and
                          in 9 there is "WebRTC Use of RTP: Future
                          Extensions".</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">So, it
                          looks obvious to use the RTP header extension
                          to show the characteristics and bandwidth
                          requirements to the network. It should not
                          introduce any backward incompatibilities
                          either.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Such
                          marking is done in every RTP packet so it can
                          be set individually for each stream and could
                          even be changed during a session (e.g. when
                          limiting the bandwidth based on RTCP
                          feedback). RFC 5286 also specifies how RTP
                          extension header usage can be negotiated in
                          SDP. I think this could be easily done by the
                          WebRTC browser for "all current and future
                          needs" if properly specified now.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">I
                          suggest that two parameters (e.g. two bytes
                          each) are encoded into the RTP header
                          extension:</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">A) The
                          maximum bandwidth requirement: Two bytes could
                          contain everything from some bps for real-time
                          text to Gbps for future 3D supersize
                          telepresence… on a logarithmic scale.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">B) The
                          quality characteristics for the stream, with
                          the highest bit set to 1, we could allocate a
                          bit each quality e.g:</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Best
                          Effort, Audio, Video, Supplemental Video,
                          Gaming, Data, Delay Insensitive (e.g. video
                          streaming), Minimum Delay, Reliable Delivery,
                          Prioritize X, Variation Y, that could be
                          combined as required to describe the stream.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">And
                          with highest bit set to 0, there could instead
                          be a number for special usage that does not
                          fit the general description of the individual
                          bits.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Please
                          note the totally different requirements a
                          diffserv and an RSVP network have to know, so
                          let’s put all into these bytes. (E.g. a
                          diffserv network don't need the bandwidth
                          usage, but RSVP reservation networks (e.g.
                          cable and 3G/4G OTT) do. There one should
                          initially reserve the maximum bandwidth
                          indicated, but can later re-reserve.)</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">/Karl</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">PS </span><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">Microsoft seems to have done work
                          in this field, defining a proprietary
                          attribute “MS Service Quality”; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span
                          style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">However that seems to apply to
                          the TURN server allocation request and would
                          therefore:</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span
                          style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">--- Apply to the whole UDP flow,
                          and could not be set for each stream
                          individually (with different requirements),
                          and</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span
                          style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">--- Does not handle the bandwidth
                          requirement for incoming real-time traffic
                          (required to reserve in RSVP type of networks)</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span
                          style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US">However the quality attributes
                          conveyed and their encoding may  be
                          considered.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span
                          style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                          lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoNormal"><span lang="EN-US">This is
                          2.2.2.19 MS-Service Quality Attribute from </span><o:p></o:p></p>
                      <p class="MsoNormal"><a moz-do-not-send="true"
href="http://msdn.microsoft.com/en-us/library/cc431507%28v=office.12%29.aspx"
title="http://msdn.microsoft.com/en-us/library/cc431507(v=office.12).aspx">http://msdn.microsoft.com/en-us/library/cc431507(v=office.12).aspx</a> <o:p></o:p></p>
                      <p class="MsoNormal"> <o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">MS-Service Quality Attribute</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">The MS-Service Quality
                            attribute is used to convey information
                            about the data stream that the protocol
                            client is intending to transfer over an
                            allocated port. The protocol client
                            SHOULD&lt;21&gt; include this attribute as
                            part of an Allocate request message. A TURN
                            server SHOULD use the information in this <span
style="background:yellow;mso-highlight:yellow">attribute to make
                              decisions about resource allocation,
                              bandwidth prioritization, and data
                              delivery methods</span>. If the attribute
                            is not present in the Allocate request
                            message, the TURN server SHOULD assume that
                            the data stream is audio with best effort
                            delivery. The format of this attribute is as
                            follows... </span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">...</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">The following stream types are
                            supported in this extension. All other
                            stream types are reserved for future use.</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0001": Audio</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0002": Video</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0003": Supplemental Video</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0004": Data</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">Service Quality (2 bytes): The
                            service quality level required by the
                            protocol client for the stream.</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">The following service quality
                            levels are supported in this extension. All
                            other service quality levels are reserved
                            for future use.</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0000": Best effort delivery.</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><em><span
                            style="font-family:Symbol">§ </span></em><em><span
style="font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                            lang="EN-US">"0x0001": Reliable delivery.</span></em><o:p></o:p></p>
                      <p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText">-----Ursprungligt
                        meddelande-----<o:p></o:p></p>
                      <p class="MsoPlainText">Från: <a
                          moz-do-not-send="true"
                          href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                        [<a moz-do-not-send="true"
                          href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                        För Harald Alvestrand<o:p></o:p></p>
                      <p class="MsoPlainText">Skickat: den 8 oktober
                        2013 13:01<o:p></o:p></p>
                      <p class="MsoPlainText">Till: <a
                          moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Ämne:
                          Re: [rtcweb] Payload Types assignments was Re:
                          SV: [mmusic] WGLC of
                          draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">On
                          10/08/2013 09:17 AM, Karl Stahl wrote:</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          Hej Magnus,</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;&gt;
                          Also, are you really interested in knowing
                          that it is VP9 vs H.264, </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;&gt;
                          isn't</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          the questions this is video of this priority
                          that is important?</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;&gt;
                          I think you need to more carefully consider
                          what are the goals you </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;&gt;
                          try to</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          achieve them.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          Actually, my concern is to get an idea of the
                          maximum bandwidth that </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          could be required for a WebRTC (ICE) setup
                          media flow. Both voice and </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          video should be prioritized over data (their
                          individual priority is of </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          less importance as long as there is sufficient
                          bandwidth for both).</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">You
                          don't know that without knowing what the
                          application is for.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">In, for
                          instance, a shooter game with voice
                          backchannels, the movement and event
                          information (data) is MORE time sensitive than
                          the voice data.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          With diffserv you don’t need to know the
                          bandwidth requirement, but </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          with RSVP reservation (like in cable and
                          mobile networks) you need to </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          know how much to reserve. Voice is like 100's
                          kbit/s, video VP8 or </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          H.264 is like 3,5 mbps.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">Again,
                          without knowing the application, you don't
                          know that.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">The
                          application could decide to use QCIF or HD,
                          and the bandwidth variation of screencast
                          (semi-static with sudden, large changes) is
                          completely different from that of a talking
                          head, which is again completely different from
                          a high-movement scene.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; To
                          add to the complication of codec variants, the
                          video codecs in </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          question for WebRTC have variable bandwidth,
                          and when there is a poor </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          connection we see Chrome reducing the video
                          window size to reduce the bandwidth used...</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; I
                          think the payload type field at best can
                          reflect a maximum bandwidth </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt; to
                          initially reserve bandwidth for, and
                          thereafter make new </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          reservations if the bandwidth changes during
                          the call. So could we </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          change RTP to show maximum bandwidth instead
                          of payload type in that </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">&gt;
                          field outside the encrypted payload :) ... Or
                          maybe that is not a joke?</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">I think
                          these ruminations only lead to one conclusion:</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">You
                          can't tell what the needed bandwidth is up
                          front without asking the application.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">You
                          can't tell what the right priority ranking is
                          without asking the application.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">If you
                          need to know the bandwidth or the priority up
                          front, the application has to tell you.
                          Anything else is pure heuristics.</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US"> </span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">_______________________________________________</span><o:p></o:p></p>
                      <p class="MsoPlainText"><span lang="EN-US">rtcweb
                          mailing list</span><o:p></o:p></p>
                      <p class="MsoPlainText"><a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org"><span
                            lang="EN-US">rtcweb@ietf.org</span></a><o:p></o:p></p>
                      <p class="MsoPlainText"><a moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb"><span lang="EN-US">https://www.ietf.org/mailman/listinfo/rtcweb</span></a><o:p></o:p></p>
                    </blockquote>
                  </div>
                  <p class="MsoNormal"><span
                      style="font-size:12.0pt;font-family:&quot;Times
                      New Roman&quot;,&quot;serif&quot;"> </span></p>
                  <div>
                    <div>
                      <p class="MsoNormal" style="margin-bottom:12.0pt"><span
                          style="font-size:12.0pt;font-family:&quot;Times
                          New Roman&quot;,&quot;serif&quot;"> </span></p>
                    </div>
                    <div>
                      <p class="MsoNormal"><span
                          style="font-size:12.0pt;font-family:&quot;Times
                          New Roman&quot;,&quot;serif&quot;">-- <o:p></o:p></span></p>
                    </div>
                    <div>
                      <p class="MsoNormal"><span
                          style="font-size:12.0pt;font-family:&quot;Times
                          New Roman&quot;,&quot;serif&quot;">Colin
                          Perkins<o:p></o:p></span></p>
                    </div>
                    <div>
                      <p class="MsoNormal"><span
                          style="font-size:12.0pt;font-family:&quot;Times
                          New Roman&quot;,&quot;serif&quot;"><a
                            moz-do-not-send="true"
                            href="http://csperkins.org/">http://csperkins.org/</a><o:p></o:p></span></p>
                    </div>
                    <div>
                      <p class="MsoNormal"><span
                          style="font-size:12.0pt;font-family:&quot;Times
                          New Roman&quot;,&quot;serif&quot;"> </span></p>
                    </div>
                    <p class="MsoNormal"><span
                        style="font-size:12.0pt;font-family:&quot;Times
                        New Roman&quot;,&quot;serif&quot;"> </span></p>
                  </div>
                  <p class="MsoNormal"><span
                      style="font-size:12.0pt;font-family:&quot;Times
                      New Roman&quot;,&quot;serif&quot;"> </span></p>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
        <div apple-content-edited="true">
          <span class="Apple-style-span" style="border-collapse:
            separate; border-spacing: 0px;">
            <div><br class="Apple-interchange-newline">
              <br class="khtml-block-placeholder">
            </div>
            <div>-- </div>
            <div>Colin Perkins</div>
            <div><a moz-do-not-send="true" href="http://csperkins.org/">http://csperkins.org/</a></div>
            <div><br>
            </div>
          </span><br class="Apple-interchange-newline">
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

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Thanks Michael,

WG, please consider these open issues and try to form your own position
on them. They are intended to be discussed at the meeting. If you have
proposals on how to resolve them or other input you are very welcome to
provide that on the list.

Thanks

Magnus

On 2014-02-26 00:44, Michael Tuexen wrote:
> Dear all,
> 
> Magnus asked me to send a list of open issues regarding data channels
> to the list. Here is my current list:
> 
> 
> * Priority
>   The W3C hasn't defined it yet. Neither for the (S)RTP media nor for the
>   data channels. We agreed on using a non strict policy for the data channels
>   (some sort of wighted fair queueing). That is all.
> 
> * Protocol
>   It seems not to be clear what needs to be provided when registering a
>   (sub)-protocol at IANA. And the name of the registry is unclear...
> 
> * SCTP parameters.
>   There was discussed the issue how to set SCTP parameters, especially path.max.retrans
>   and association.max.retrans. Also HB.Interval might be of interest.
>   RFC 4060 recommends path.max.retrans=5, association.max.retrans=10, but has multihoming
>   in mind. To avoid the dormant state, path.max.retrans = association.max.retrans should be used.
>   I would suggest 10 for this value. Should HEARTBEATs be disabled?
> 
> * U-C 7: Proxy browsing
> 
> * Alternate CC for SCTP
>   Currently there is only the standard CC. However, in some places negotiation of CC is
>   mentioned.
> 
> I'm currently going through the backlog of comments regarding the data channels
> ID and I'll try to address the issues. If I find other issues, I send an update
> to the above list.
> 
> Best regards
> Michael
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>
Thread-Topic: [rtcweb] [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
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Karl,=20

I am totally lost on this thread. Could you start a new tmead that summariz=
e what the issues is, what seems to be the point of debate, and what your v=
iew is on what we should do. I think that would help make progress.=20

Thank you,=20

Cullen=20


On Feb 25, 2014, at 8:20 AM, Karl Stahl <karl.stahl@intertex.se> wrote:

> Hi Alan,
> =20
> I have in previous email http://www.ietf.org/mail-archive/web/tram/curren=
t/msg00304.html suggested that both DISCUSS and draft-thomson-tram-turn-ban=
dwidth-00.txt is taken off the TRAM WG, but I think is shall be reintroduce=
d after you clarify as you do in
> http://www.ietf.org/mail-archive/web/tram/current/msg00300.html :
> Hi Karl, Thanks for your comments and feedback on the draft.
> You are correct in saying that the BANDWIDTH extension is not about QoS. =
 It is about fairness between users of a TURN server, and a TURN server bei=
ng able to indicate rate limiting policy to users. in the draft.
> =20
> The =93confusion=94 (to say least, see below) surrounding QoS within IETF=
 has lead me to protest and address the TRAM WG and RTCWEB WG and ADs with =
the advise in http://www.ietf.org/mail-archive/web/tram/current/msg00304.ht=
ml :
> =93The TRAM WG and RTCWEB WG chairs and ADs are advised to review whether=
 IETF standard work in these WGs by contributors having an interest in more=
 than one of current or emerging: ISPs, carrier equipment vendors or web br=
owser makers, may discriminate any with an interest only in one of these. T=
he same should apply to non-activity by WG contributors to remedy such disc=
rimination opened by allowed proprietary usage of RFCs such as RFC 5285.=94
> =20
> This is because the introduction of the QoS related usage of RFC 5285 int=
o draft-ietf-rtcweb-rtp-usage in combinations with the Milestone 3 objectiv=
es of TRAM, immediately would allow for:
> (1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in addi=
tion to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP variants) to qu=
ickly allow us to finally enjoy the high quality and connectivity (NAT/fire=
wall traversal) of real-time communication services now possible, for =20
> (2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), und=
er *all* OSs, and *all* current and future IP networks
> (3) *without* having to be forced into application specific networks (PST=
N, IMS) instead of the Internet (including OTT).
> =20
> While the =93confusion=94 surrounding the QoS discussion in TRAM and RTCW=
EB combined with thegeneral QoS attitude within IETF =93it is all about ban=
dwidth=94 and =93it will go away with time=94 otherwise:
> (i) will *not allow* ISP=92s to use already available and currently deplo=
yable quality IP pipes for real-time traffic to also be used for WebRTC gen=
erated real-time traffic.
> (ii) will *not allow* some network types (e.g. Cable Networks and Mobile =
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming vi=
deo and file sharing. That all networks *will be inhibited* from the very c=
ommon and used method of simply providing an extra IP pipe (often provided =
over the same wire but level-2 separated) dedicated for real-time usage
> (*by resisting TRAM implementation of step B*)
> *while* newer, fiber only type of networks, still can borrow bandwidth at=
 no extra cost, *by proprietary usage* of RFC 5285 by browsers.
> (iii) will leave most ISPs to *only use* raw bandwidth capacity increase =
that may have to be 10-folded to reach sufficient QoE when WebRTC usage bec=
omes popular (if at all possible, since unmanaged IP pipes intermittently a=
re filled, whatever bandwidth is available).
> =20
> As Good doers, we should not allow this to happen.
> =20
> /Karl
> =20
> =20
> Fr=E5n: Alan Johnston [mailto:alan.b.johnston@gmail.com]=20
> Skickat: den 21 februari 2014 18:59
> Till: Pal Martinsen (palmarti)
> Kopia: tram@ietf.org; Oleg Moskalenko; Karl Stahl; Yoakum, John H (John)
> =C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
> =20
> I guess the way I would expect this to progress is for QoS discussions to=
 happen in another working group.  Once that other working group came to co=
nsensus on an approach, and if that approach required STUN or TURN extensio=
ns, then we would discuss mechanisms and possible milestones in TRAM.
> =20
> - Alan -
> =20
>=20
> On Fri, Feb 21, 2014 at 3:37 AM, Pal Martinsen (palmarti) <palmarti@cisco=
.com> wrote:
> Hi,
> =20
> I agree the full QoS discussion should _not_ happen in TRAM. If you are i=
nterested in helping out in that area I suggest you looking into the AEON m=
ailing list at: https://www.ietf.org/mailman/listinfo/aeon . They are curre=
ntly working on a problem-statement draft and a use-case draft, any input t=
o those would be very helpful. (http://tools.ietf.org/html/draft-eckel-aeon=
-use-cases-01, http://tools.ietf.org/html/draft-eckel-aeon-problem-statemen=
t-00).
> =20
> That said, STUN have a few nice characteristics that makes it a perfect c=
andidate for transporting some of the QoS information.  IMHO that would be =
extending the STUN spec and should be within the TRAM charter.  The main go=
al of draft-martinsen-tram-discuss was to show how already existing QoS mec=
hanisms could be transported with STUN to provide more value, and to start =
the discussion if TRAM is the appropriate place to have those on the wire f=
ormat discussions.
> =20
> .-.
> P=E5l-Erik
> =20
> =20
> On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <yoakum@avaya.com> wro=
te:
>=20
>=20
> +1
> I fully agree with the comments that QOS should be a low priority for the=
 initial focus of the TRAM efforts.  There are other groups doing QOS work =
and frankly I engage in WebRTC multimedia interactions daily over the Inter=
net, enterprise VPNs, and various combinations and seldom suffer egregious =
quality issues.  I am more concerned about carriers doing things to regulat=
e or degrade WebRTC flows than a failure of existing Internet mechanisms to=
 enable them.
> =20
> Significant focus on QOS before we better enable TURN to be easily used i=
n a browser environment taking advantage or normal web characteristics (as =
opposed to historic telephony constructs) would seem to be highly distracti=
ng at this point.
> =20
> =20
> Cheers,
> John
> =20
> AVAYA
> 1.919.425.8446
> =20
> From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
> Sent: Thursday, February 20, 2014 12:43 PM
> To: Alan Johnston
> Cc: Karl Stahl; tram@ietf.org
> Subject: Re: [tram] Fwd: I-D Action: draft-thomson-tram-turn-bandwidth-00=
.txt
> =20
> =20
> =20
>=20
> On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <alan.b.johnston@gmail.com=
> wrote:
> =20
> =20
> =20
> Personally, I am not sure how much QoS is actually in scope for TRAM. Hav=
e you been following RMCAT where congestion avoidance for RTP is being deve=
loped?  I see some overlap in your goals and the goals of that work.
> -
> =20
> =20
> I'd concentrate on the TURN application-level functionality, for now, and=
 I'd leave QoS for the future discussions.
>=20
> _______________________________________________
> tram mailing list
> tram@ietf.org
> https://www.ietf.org/mailman/listinfo/tram
> =20
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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> That said, I think you have pointed out this charter is abysmally
> vague - it does not say what the WG is not going to do. If I
> decided to do BGP for routing updates over TURN it would be
> within the scope of this charter.

Cullen, is it your opinion that a working group is always free to do
anything that's not explicitly forbidden in its charter?

I think that a working group is limited to doing what *is* in its
charter, and we sometimes explicitly forbid things for emphasis.

The TRAM charter says this:

   The work will include the addition of DTLS
   as an additional transport, authentication mechanisms, and
   extensions to TURN and STUN.

I don't think that would support, for example, BGP over TURN.

Barry


From nobody Wed Feb 26 00:36:54 2014
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From: Spencer Dawkins <spencerdawkins.ietf@gmail.com>
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Subject: [rtcweb] BCP over TURN will not be in scope ... and MPLS over UDP over TURN will not be in scope ... and ...
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So ... TRAM was chartered last Thursday ...

I'm gonna let Gonzalo (Yet Another Soon-to-be Former AD Serving as WG 
Chair) and Simon actually chair their working group for at least a full 
week before intruding further, but for now, let's assume that the 
working group wants to work on the milestones they signed up for last 
week, and see how badly that assumption blows up, rather than talk about 
rechartering every time someone proposes a topic that's out of scope.

I look forward to attending the first working group session next week, 
and to balloting on six TRAM drafts before IETF 92.

Thanks,

Spencer, as the responsible AD


From nobody Wed Feb 26 00:36:57 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Barry Leiba <barryleiba@computer.org>
Thread-Topic: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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On Feb 26, 2014, at 9:12 AM, Barry Leiba <barryleiba@computer.org> wrote:

>> That said, I think you have pointed out this charter is abysmally
>> vague - it does not say what the WG is not going to do. If I
>> decided to do BGP for routing updates over TURN it would be
>> within the scope of this charter.
>=20
> Cullen, is it your opinion that a working group is always free to do
> anything that's not explicitly forbidden in its charter?
>=20
> I think that a working group is limited to doing what *is* in its
> charter, and we sometimes explicitly forbid things for emphasis.
>=20
> The TRAM charter says this:
>=20
>   The work will include the addition of DTLS
>   as an additional transport, authentication mechanisms, and
>   extensions to TURN and STUN.
>=20
> I don't think that would support, for example, BGP over TURN.
>=20
> Barry

BGP over TURN sounds like a turn extension to me. Now you might think that =
is a silly one and I would agree but I assure you Karl does not think the Q=
oS extensions are silly extensions to TURN.=20

So in your opinion are they in scope or out, I think that is the heart of t=
he issue right now. I=92m of the opinion they are in scope so I look forwar=
d to agenda time be allocated for them.=20

Cullen


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Spencer Dawkins <spencerdawkins.ietf@gmail.com>
Thread-Topic: BCP over TURN will not be in scope ... and MPLS over UDP over TURN will not be in scope ... and ...
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On Feb 26, 2014, at 10:52 AM, Spencer Dawkins <spencerdawkins.ietf@gmail.co=
m> wrote:

> So ... TRAM was chartered last Thursday ...
>=20
> I'm gonna let Gonzalo (Yet Another Soon-to-be Former AD Serving as WG Cha=
ir) and Simon actually chair their working group for at least a full week b=
efore intruding further, but for now, let's assume that the working group w=
ants to work on the milestones they signed up for last week, and see how ba=
dly that assumption blows up, rather than talk about rechartering every tim=
e someone proposes a topic that's out of scope.
>=20
> I look forward to attending the first working group session next week, an=
d to balloting on six TRAM drafts before IETF 92.
>=20
> Thanks,
>=20
> Spencer, as the responsible AD

Makes sense - I get the desire to let some work get done.=20

I worry that every time someone says something is is not in scope of this W=
G, it is not going to feel like an open consensus decisions but is instead =
going to feel like a arbitrary fiat decisions by a chair or IESG made in a =
dark and smoky room - I hope I am wrong.=20



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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "tram@ietf.org" <tram@ietf.org>, "tsv-ads@ietf.org" <tsv-ads@ietf.org>
Subject: [rtcweb] Last note about QoS from the responsible AD before the rant ...
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Using smaller words (since we're all very busy):

QoS is out of scope for TRAM.

It is in scope for TSVWG.

It will not be necessary helpful to continue to talk about QoS in TRAM.

Thanks for taking the conversation to a mailing list where it's in scope.

Spencer, as responsible AD

On Tuesday, February 25, 2014, Cullen Jennings (fluffy) <fluffy@cisco.com>
wrote:

>
> Karl,
>
> I am totally lost on this thread. Could you start a new tmead that
> summarize what the issues is, what seems to be the point of debate, and
> what your view is on what we should do. I think that would help make
> progress.
>
> Thank you,
>
> Cullen
>
>
> On Feb 25, 2014, at 8:20 AM, Karl Stahl <karl.stahl@intertex.se<javascrip=
t:;>>
> wrote:
>
> > Hi Alan,
> >
> > I have in previous email
> http://www.ietf.org/mail-archive/web/tram/current/msg00304.html suggested
> that both DISCUSS and draft-thomson-tram-turn-bandwidth-00.txt is taken o=
ff
> the TRAM WG, but I think is shall be reintroduced after you clarify as yo=
u
> do in
> > http://www.ietf.org/mail-archive/web/tram/current/msg00300.html :
> > Hi Karl, Thanks for your comments and feedback on the draft.
> > You are correct in saying that the BANDWIDTH extension is not about QoS=
.
>  It is about fairness between users of a TURN server, and a TURN server
> being able to indicate rate limiting policy to users. in the draft.
> >
> > The "confusion" (to say least, see below) surrounding QoS within IETF
> has lead me to protest and address the TRAM WG and RTCWEB WG and ADs with
> the advise in
> http://www.ietf.org/mail-archive/web/tram/current/msg00304.html :
> > "The TRAM WG and RTCWEB WG chairs and ADs are advised to review whether
> IETF standard work in these WGs by contributors having an interest in mor=
e
> than one of current or emerging: ISPs, carrier equipment vendors or web
> browser makers, may discriminate any with an interest only in one of thes=
e.
> The same should apply to non-activity by WG contributors to remedy such
> discrimination opened by allowed proprietary usage of RFCs such as RFC
> 5285."
> >
> > This is because the introduction of the QoS related usage of RFC 5285
> into draft-ietf-rtcweb-rtp-usage in combinations with the Milestone 3
> objectives of TRAM, immediately would allow for:
> > (1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in
> addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP variants)
> to quickly allow us to finally enjoy the high quality and connectivity
> (NAT/firewall traversal) of real-time communication services now possible=
,
> for
> > (2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC),
> under *all* OSs, and *all* current and future IP networks
> > (3) *without* having to be forced into application specific networks
> (PSTN, IMS) instead of the Internet (including OTT).
> >
> > While the "confusion" surrounding the QoS discussion in TRAM and RTCWEB
> combined with thegeneral QoS attitude within IETF "it is all about
> bandwidth" and "it will go away with time" otherwise:
> > (i) will *not allow* ISP's to use already available and currently
> deployable quality IP pipes for real-time traffic to also be used for
> WebRTC generated real-time traffic.
> > (ii) will *not allow* some network types (e.g. Cable Networks and Mobil=
e
> OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
> video and file sharing. That all networks *will be inhibited* from the ve=
ry
> common and used method of simply providing an extra IP pipe (often provid=
ed
> over the same wire but level-2 separated) dedicated for real-time usage
> > (*by resisting TRAM implementation of step B*)
> > *while* newer, fiber only type of networks, still can borrow bandwidth
> at no extra cost, *by proprietary usage* of RFC 5285 by browsers.
> > (iii) will leave most ISPs to *only use* raw bandwidth capacity increas=
e
> that may have to be 10-folded to reach sufficient QoE when WebRTC usage
> becomes popular (if at all possible, since unmanaged IP pipes
> intermittently are filled, whatever bandwidth is available).
> >
> > As Good doers, we should not allow this to happen.
> >
> > /Karl
> >
> >
> > Fr=E5n: Alan Johnston [mailto:alan.b.johnston@gmail.com <javascript:;>]
> > Skickat: den 21 februari 2014 18:59
> > Till: Pal Martinsen (palmarti)
> > Kopia: tram@ietf.org <javascript:;>; Oleg Moskalenko; Karl Stahl;
> Yoakum, John H (John)
> > =C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
> >
> > I guess the way I would expect this to progress is for QoS discussions
> to happen in another working group.  Once that other working group came t=
o
> consensus on an approach, and if that approach required STUN or TURN
> extensions, then we would discuss mechanisms and possible milestones in
> TRAM.
> >
> > - Alan -
> >
> >
> > On Fri, Feb 21, 2014 at 3:37 AM, Pal Martinsen (palmarti) <
> palmarti@cisco.com <javascript:;>> wrote:
> > Hi,
> >
> > I agree the full QoS discussion should _not_ happen in TRAM. If you are
> interested in helping out in that area I suggest you looking into the AEO=
N
> mailing list at: https://www.ietf.org/mailman/listinfo/aeon . They are
> currently working on a problem-statement draft and a use-case draft, any
> input to those would be very helpful. (
> http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01,
> http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00).
> >
> > That said, STUN have a few nice characteristics that makes it a perfect
> candidate for transporting some of the QoS information.  IMHO that would =
be
> extending the STUN spec and should be within the TRAM charter.  The main
> goal of draft-martinsen-tram-discuss was to show how already existing QoS
> mechanisms could be transported with STUN to provide more value, and to
> start the discussion if TRAM is the appropriate place to have those on th=
e
> wire format discussions.
> >
> > .-.
> > P=E5l-Erik
> >
> >
> > On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) <yoakum@avaya.com<ja=
vascript:;>>
> wrote:
> >
> >
> > +1
> > I fully agree with the comments that QOS should be a low priority for
> the initial focus of the TRAM efforts.  There are other groups doing QOS
> work and frankly I engage in WebRTC multimedia interactions daily over th=
e
> Internet, enterprise VPNs, and various combinations and seldom suffer
> egregious quality issues.  I am more concerned about carriers doing thing=
s
> to regulate or degrade WebRTC flows than a failure of existing Internet
> mechanisms to enable them.
> >
> > Significant focus on QOS before we better enable TURN to be easily used
> in a browser environment taking advantage or normal web characteristics (=
as
> opposed to historic telephony constructs) would seem to be highly
> distracting at this point.
> >
> >
> > Cheers,
> > John
> >
> > AVAYA
> > 1.919.425.8446
> >
> > From: tram [mailto:tram-bounces@ietf.org <javascript:;>] On Behalf Of
> Oleg Moskalenko
> > Sent: Thursday, February 20, 2014 12:43 PM
> > To: Alan Johnston
> > Cc: Karl Stahl; tram@ietf.org <javascript:;>
> > Subject: Re: [tram] Fwd: I-D Action:
> draft-thomson-tram-turn-bandwidth-00.txt
> >
> >
> >
> >
> > On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston <
> alan.b.johnston@gmail.com <javascript:;>> wrote:
> >
> >
> >
> > Personally, I am not sure how much QoS is actually in scope for TRAM.
> Have you been following RMCAT where congestion avoidance for RTP is being
> developed?  I see some overlap in your goals and the goals of that work.
> > -
> >
> >
> > I'd concentrate on the TURN application-level functionality, for now,
> and I'd leave QoS for the future discussions.
> >
> > _______________________________________________
> > tram mailing list
> > tram@ietf.org <javascript:;>
> > https://www.ietf.org/mailman/listinfo/tram
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org <javascript:;>
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf303f672e6a778604f34f4302
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Using smaller words (since we&#39;re all very busy):&nbsp;<div><br></div><d=
iv>QoS is out of scope for TRAM.</div><div><br></div><div>It is in scope fo=
r TSVWG.</div><div><br></div><div>It will not be necessary helpful&nbsp;to =
continue to talk about QoS in TRAM.</div>
<div><br></div><div>Thanks for taking the conversation to a mailing list wh=
ere it&#39;s in scope.</div><div><br></div><div>Spencer, as responsible AD<=
/div><div><br>On Tuesday, February 25, 2014, Cullen Jennings (fluffy) &lt;<=
a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Karl,<br>
<br>
I am totally lost on this thread. Could you start a new tmead that summariz=
e what the issues is, what seems to be the point of debate, and what your v=
iew is on what we should do. I think that would help make progress.<br>

<br>
Thank you,<br>
<br>
Cullen<br>
<br>
<br>
On Feb 25, 2014, at 8:20 AM, Karl Stahl &lt;<a href=3D"javascript:;" onclic=
k=3D"_e(event, &#39;cvml&#39;, &#39;karl.stahl@intertex.se&#39;)">karl.stah=
l@intertex.se</a>&gt; wrote:<br>
<br>
&gt; Hi Alan,<br>
&gt;<br>
&gt; I have in previous email <a href=3D"http://www.ietf.org/mail-archive/w=
eb/tram/current/msg00304.html" target=3D"_blank">http://www.ietf.org/mail-a=
rchive/web/tram/current/msg00304.html</a> suggested that both DISCUSS and d=
raft-thomson-tram-turn-bandwidth-00.txt is taken off the TRAM WG, but I thi=
nk is shall be reintroduced after you clarify as you do in<br>

&gt; <a href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00300.=
html" target=3D"_blank">http://www.ietf.org/mail-archive/web/tram/current/m=
sg00300.html</a> :<br>
&gt; Hi Karl, Thanks for your comments and feedback on the draft.<br>
&gt; You are correct in saying that the BANDWIDTH extension is not about Qo=
S. &nbsp;It is about fairness between users of a TURN server, and a TURN se=
rver being able to indicate rate limiting policy to users. in the draft.<br=
>

&gt;<br>
&gt; The &ldquo;confusion&rdquo; (to say least, see below) surrounding QoS =
within IETF has lead me to protest and address the TRAM WG and RTCWEB WG an=
d ADs with the advise in <a href=3D"http://www.ietf.org/mail-archive/web/tr=
am/current/msg00304.html" target=3D"_blank">http://www.ietf.org/mail-archiv=
e/web/tram/current/msg00304.html</a> :<br>

&gt; &ldquo;The TRAM WG and RTCWEB WG chairs and ADs are advised to review =
whether IETF standard work in these WGs by contributors having an interest =
in more than one of current or emerging: ISPs, carrier equipment vendors or=
 web browser makers, may discriminate any with an interest only in one of t=
hese. The same should apply to non-activity by WG contributors to remedy su=
ch discrimination opened by allowed proprietary usage of RFCs such as RFC 5=
285.&rdquo;<br>

&gt;<br>
&gt; This is because the introduction of the QoS related usage of RFC 5285 =
into draft-ietf-rtcweb-rtp-usage in combinations with the Milestone 3 objec=
tives of TRAM, immediately would allow for:<br>
&gt; (1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in a=
ddition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP variants) to=
 quickly allow us to finally enjoy the high quality and connectivity (NAT/f=
irewall traversal) of real-time communication services now possible, for<br=
>

&gt; (2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under *all* OSs, and *all* current and future IP networks<br>
&gt; (3) *without* having to be forced into application specific networks (=
PSTN, IMS) instead of the Internet (including OTT).<br>
&gt;<br>
&gt; While the &ldquo;confusion&rdquo; surrounding the QoS discussion in TR=
AM and RTCWEB combined with thegeneral QoS attitude within IETF &ldquo;it i=
s all about bandwidth&rdquo; and &ldquo;it will go away with time&rdquo; ot=
herwise:<br>
&gt; (i) will *not allow* ISP&rsquo;s to use already available and currentl=
y deployable quality IP pipes for real-time traffic to also be used for Web=
RTC generated real-time traffic.<br>
&gt; (ii) will *not allow* some network types (e.g. Cable Networks and Mobi=
le OTT) to borrow bandwidth from data traffic and QoS insensitive streaming=
 video and file sharing. That all networks *will be inhibited* from the ver=
y common and used method of simply providing an extra IP pipe (often provid=
ed over the same wire but level-2 separated) dedicated for real-time usage<=
br>

&gt; (*by resisting TRAM implementation of step B*)<br>
&gt; *while* newer, fiber only type of networks, still can borrow bandwidth=
 at no extra cost, *by proprietary usage* of RFC 5285 by browsers.<br>
&gt; (iii) will leave most ISPs to *only use* raw bandwidth capacity increa=
se that may have to be 10-folded to reach sufficient QoE when WebRTC usage =
becomes popular (if at all possible, since unmanaged IP pipes intermittentl=
y are filled, whatever bandwidth is available).<br>

&gt;<br>
&gt; As Good doers, we should not allow this to happen.<br>
&gt;<br>
&gt; /Karl<br>
&gt;<br>
&gt;<br>
&gt; Fr=E5n: Alan Johnston [mailto:<a href=3D"javascript:;" onclick=3D"_e(e=
vent, &#39;cvml&#39;, &#39;alan.b.johnston@gmail.com&#39;)">alan.b.johnston=
@gmail.com</a>]<br>
&gt; Skickat: den 21 februari 2014 18:59<br>
&gt; Till: Pal Martinsen (palmarti)<br>
&gt; Kopia: <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &=
#39;tram@ietf.org&#39;)">tram@ietf.org</a>; Oleg Moskalenko; Karl Stahl; Yo=
akum, John H (John)<br>
&gt; =C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.tx=
t<br>
&gt;<br>
&gt; I guess the way I would expect this to progress is for QoS discussions=
 to happen in another working group. &nbsp;Once that other working group ca=
me to consensus on an approach, and if that approach required STUN or TURN =
extensions, then we would discuss mechanisms and possible milestones in TRA=
M.<br>

&gt;<br>
&gt; - Alan -<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Feb 21, 2014 at 3:37 AM, Pal Martinsen (palmarti) &lt;<a href=
=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;palmarti@cisco.=
com&#39;)">palmarti@cisco.com</a>&gt; wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt; I agree the full QoS discussion should _not_ happen in TRAM. If you ar=
e interested in helping out in that area I suggest you looking into the AEO=
N mailing list at: <a href=3D"https://www.ietf.org/mailman/listinfo/aeon" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/aeon</a> . They are =
currently working on a problem-statement draft and a use-case draft, any in=
put to those would be very helpful. (<a href=3D"http://tools.ietf.org/html/=
draft-eckel-aeon-use-cases-01" target=3D"_blank">http://tools.ietf.org/html=
/draft-eckel-aeon-use-cases-01</a>, <a href=3D"http://tools.ietf.org/html/d=
raft-eckel-aeon-problem-statement-00" target=3D"_blank">http://tools.ietf.o=
rg/html/draft-eckel-aeon-problem-statement-00</a>).<br>

&gt;<br>
&gt; That said, STUN have a few nice characteristics that makes it a perfec=
t candidate for transporting some of the QoS information. &nbsp;IMHO that w=
ould be extending the STUN spec and should be within the TRAM charter. &nbs=
p;The main goal of draft-martinsen-tram-discuss was to show how already exi=
sting QoS mechanisms could be transported with STUN to provide more value, =
and to start the discussion if TRAM is the appropriate place to have those =
on the wire format discussions.<br>

&gt;<br>
&gt; .-.<br>
&gt; P=E5l-Erik<br>
&gt;<br>
&gt;<br>
&gt; On 20 Feb 2014, at 19:55 pm, Yoakum, John H (John) &lt;<a href=3D"java=
script:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;yoakum@avaya.com&#39;)"=
>yoakum@avaya.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; +1<br>
&gt; I fully agree with the comments that QOS should be a low priority for =
the initial focus of the TRAM efforts. &nbsp;There are other groups doing Q=
OS work and frankly I engage in WebRTC multimedia interactions daily over t=
he Internet, enterprise VPNs, and various combinations and seldom suffer eg=
regious quality issues. &nbsp;I am more concerned about carriers doing thin=
gs to regulate or degrade WebRTC flows than a failure of existing Internet =
mechanisms to enable them.<br>

&gt;<br>
&gt; Significant focus on QOS before we better enable TURN to be easily use=
d in a browser environment taking advantage or normal web characteristics (=
as opposed to historic telephony constructs) would seem to be highly distra=
cting at this point.<br>

&gt;<br>
&gt;<br>
&gt; Cheers,<br>
&gt; John<br>
&gt;<br>
&gt; AVAYA<br>
&gt; 1.919.425.8446<br>
&gt;<br>
&gt; From: tram [mailto:<a href=3D"javascript:;" onclick=3D"_e(event, &#39;=
cvml&#39;, &#39;tram-bounces@ietf.org&#39;)">tram-bounces@ietf.org</a>] On =
Behalf Of Oleg Moskalenko<br>
&gt; Sent: Thursday, February 20, 2014 12:43 PM<br>
&gt; To: Alan Johnston<br>
&gt; Cc: Karl Stahl; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvm=
l&#39;, &#39;tram@ietf.org&#39;)">tram@ietf.org</a><br>
&gt; Subject: Re: [tram] Fwd: I-D Action: draft-thomson-tram-turn-bandwidth=
-00.txt<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston &lt;<a href=3D"javascri=
pt:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;alan.b.johnston@gmail.com&#=
39;)">alan.b.johnston@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Personally, I am not sure how much QoS is actually in scope for TRAM. =
Have you been following RMCAT where congestion avoidance for RTP is being d=
eveloped? &nbsp;I see some overlap in your goals and the goals of that work=
.<br>

&gt; -<br>
&gt;<br>
&gt;<br>
&gt; I&#39;d concentrate on the TURN application-level functionality, for n=
ow, and I&#39;d leave QoS for the future discussions.<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; tram mailing list<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;tra=
m@ietf.org&#39;)">tram@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tram" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tram</a><br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;rtc=
web@ietf.org&#39;)">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>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;rtcweb@i=
etf.org&#39;)">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>

--20cf303f672e6a778604f34f4302--


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RTCWEB WG,

This is one of the documents that this WG's Data Channel is normatively
dependent on. If we don't help review the work of other WGs we will not
finish ourselves.

So please review and send comments that you done and provide any
comments you have.

Cheers

Magnus Westerlund

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A gentle reminder, I've seen no notes of support, or comments on this 
document during the WGLC so far.

Could people please send comments to the list if they think this is 
ready to publish?

Gorry

On 07/02/2014 17:18, Gorry Fairhurst wrote:
> This email announces the start of a working group last call
> of draft-ietf-tsvwg-sctp-dtls-encaps-03,
> "DTLS Encapsulation of SCTP Packets". This document was
> discussed at the WG meeting in Vancouver and was thought at that
> meeting to be ready for WG review, this email starts this process by
> requesting review comments.
>
> Please send any comments, notes of support, or concerns to the TSVWG list.
>
> The draft is available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps
>
> The last call will run for TWO weeks, ending 28th February 2014.
>
> James, David, and Gorry
> (TSVWG Chairs)
> tsvwg-chairs@ietf.org
>




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To: "'Yoakum, John H \(John\)'" <yoakum@avaya.com>, "'Oleg Moskalenko'" <mom040267@gmail.com>, "'Alan Johnston'" <alan.b.johnston@gmail.com>
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Subject: [rtcweb]  [tram] The way to "Interfacing to QoS", A level 3-5 IP/IETF/WebRTC-thing how to inface to lower level's QoS-stuff
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(After having typed this up; I see today=92s charter discussion etc. =
that I
haven=92t read yet, but it will hopefully will be clearer by pointing at =
this
entry.)

=20

Hi John, and=20

Collin, P=E5l, Magnus, Charles,  Simon,

=20

In my step A) B) and D) efforts in
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html for the
=93mission to bring quality to real-time traffic over our best effort =
Internet
is a huge mission, I am convinced it not a huge task =96 We just have to =
be a
bit clever here :). I only see these few standard steps required, before =
it
can happen!=94

=20

To accomplish these (1), (2), (3)-things for:

(1) *all* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in
addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to
quickly allow us to finally enjoy the high quality and connectivity
(NAT/firewall traversal) of real-time communication services now =
possible,
for =20

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks=20

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of the Internet (including OTT).

=20

There are a lot of Mission Impossible, Pointers, Good arguments and =
other
things to consider in the mails from you guys that I have not (yet)
answered.

=20

Allow me to come back to those under THIS THREAD/Subject after having
figured out how to do it in a structured way, with the aim of collecting =
the
summary in a new "Interfacing to QoS" thread, that only will relate to =
how
IP/IETF/WebRTC level 3-5 =96 *can interface* to the QoS-stuff (whether =
within
IETF, in the telephony world or elsewhere e.g. IEEE Ethernet, mainly
relating to lower level).

=20

I think this is doable *without* ending up in any of the (i), (ii),
(iii)-things=20

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. That all networks *will be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

*while* newer, fiber only type of networks, still can borrow bandwidth =
at no
extra cost, *by proprietary usage* of RFC 5285 by browsers.=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may have to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

(?) and possible other Evil -things.

=20

I refer to =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html
for these =93-things=94 and further.

=20

=20

John, Alan, Henry (Sinnreich), Henning, Richard, Cullen, J=F6rgen =
(Bj=F6rkner),
Lars (Berggren) and other good friends since the early SIP age (before =
SIP
was =93hijacked=94 into application specific network and proprietary =
usage and
nowadays rarely is used the IETF-way):=20

=20

I want to point out how =93confusing=94 the QoS-issue has been and as =
hard as I
will the fight Mission Impossible attitude, I will also fight the =93it =
is all
about bandwidth=94 and =93it will go away with time=94 attitude=85  =
(also in the
above http reference)

=20

I agree with parts of what John says below but the remedy to achieve the
Good (1), (2), (3)-things instead of the Evil (i), (ii), (iii)-things, =
is
NOT to ignore QoS-things =96 It is the opposite!=20

=20

For you having more and relevant input =96 That has not been said, and =
that
you think I will not address anyway =96 in this "Interfacing to QoS" =
path,
please provide, but I will not be able to process much more than already
have been said =96 and I will use http back-pointers for what has =
already been
discussed in more detail.

=20

If you want to point out QoS-stuff Requiring More Input-from/Feedback-to =
the
level 3 and above stuff IN ADDITION to what already has been discussed =
here
http://www.ietf.org/mail-archive/web/tram/current/msg00302.html =20

Please also point out WHAT and WHY that can/or cannot be handled by the
methods proposed here and whether other existing methods can be used to
achieve the mission of "Interfacing to QoS" from IP =3D level 3 to lower
levels.

=20

I will do some thinking about what LISTs different things shall be =
brought
to and then split the [TRAM] [WEBRTC] and advice regarding involving =
other
list will be considered. Such will probably show up and happen during =
this
mission.

=20

=20

Another thing from
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html: What =
name=20

Footnote **[Karl]* For similar reasons I have started to say things like
=93the box containing the TURN server=94. We should not change the TURN =
server
spec into including specific QoS methods to apply (like reservation,
changing DSCP bits or applying traffic shaping mechanism). What is =
happening
here is that we share the traffic information that the TURN server gets =
hold
of, with a QoS applying function (that will be different for different
networks) in =93the same box=94. With =93the same box=94 I mean that we =
for now
don=92t care about the interface/commands for sharing information =
between the
TURN server and the QoS applying function. There may be a future need to
specify/standardize this, but if we start doing that now and in TRAM, we
risk ending up in a 10-year process before having something in place =
(which
without a spec automatically will happen in vendor=92s product =
development, by
putting the TURN server into already available =93QoS applying =
function=94 boxes
(like firewalls or the mobile DPI/PCRF combination).
=20
*What we need to consider and specify, is only what traffic info is =
required
(to be provided by the application/browser) for =93QoS applying =
functions=94 in
different networks (including the very common reservation types) to do =
job
(i.e. giving us good QoE for real-time applications).*

=20

And Henry, I will have to continue to use the S(BC)-word and the =
I(MS)-word,
since you did not succeed to =93eradicate=94 them ;-)  Let=92s hope we =
find a word
for the above box (quite undefined on the QoS-side =96 It is only the =
TURN
side and traffic info we are up to designing here) that does not need to =
be
=93eradicated=94. So no WebRTC-SBC I guess (hopefully it will not even =
be
WebRTC-specific), and ICE-SBC sounds crazy (since ICE was developed to
remove the need of SIP SBCs), the same applies to TURN-SBC (which =
implies
that this box hopefully is not even ICE-specific). TURN-QoS-Interface =
box is
relevant but long=85=20

=20

/Karl

=20

PS: To avoid being stopped by the =93to many recipients=94 spam filter, =
the Guys
(including Mary) from=20

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html=20

and the ones mentioned in the email are copied separately.

=20

=20

Fr=E5n: Yoakum, John H (John) [mailto:yoakum@avaya.com]=20
Skickat: den 20 februari 2014 19:56
Till: Oleg Moskalenko; Alan Johnston
Kopia: Karl Stahl; tram@ietf.org
=C4mne: RE: [tram] Fwd: I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt

+1

I fully agree with the comments that QOS should be a low priority for =
the
initial focus of the TRAM efforts.  There are other groups doing QOS =
work
and frankly I engage in WebRTC multimedia interactions daily over the
Internet, enterprise VPNs, and various combinations and seldom suffer
egregious quality issues.  I am more concerned about carriers doing =
things
to regulate or degrade WebRTC flows than a failure of existing Internet
mechanisms to enable them.

=20

Significant focus on QOS before we better enable TURN to be easily used =
in a
browser environment taking advantage or normal web characteristics (as
opposed to historic telephony constructs) would seem to be highly
distracting at this point.

=20

Cheers,

John

=20

AVAYA
1.919.425.8446=20

=20

From: tram [mailto:tram-bounces@ietf.org] On Behalf Of Oleg Moskalenko
Sent: Thursday, February 20, 2014 12:43 PM
To: Alan Johnston
Cc: Karl Stahl; tram@ietf.org
Subject: Re: [tram] Fwd: I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

=20

On Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston =
<alan.b.johnston@gmail.com>
wrote:

=20

Personally, I am not sure how much QoS is actually in scope for TRAM. =
Have
you been following RMCAT where congestion avoidance for RTP is being
developed?  I see some overlap in your goals and the goals of that work.

=20

I'd concentrate on the TURN application-level functionality, for now, =
and
I'd leave QoS for the future discussions.


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<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(A=
fter having typed this up; I see today&#8217;s charter discussion etc. =
that I haven&#8217;t read yet, but it will hopefully will be clearer by =
pointing at this entry.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 John, and <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Co=
llin, P=E5l, Magnus, Charles,=A0 Simon,<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>In=
 my step A) B) and D) efforts in </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a> =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
r the</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'> </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet is a huge mission, I am convinced =
it not a huge task &#8211; We just have to be a bit clever here :). I =
only see these few standard steps required, before it can =
happen!</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>To=
 accomplish these (1), (2), (3)-things for:<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
*<b>all</b>* protocols using IETF RTP/SRTP and/or IETF TURN/STU/ICE (in =
addition to WebRTC: e.g. SIP, Skype and Facetime, and most VoIP =
variants) to quickly allow us to finally enjoy the high quality and =
connectivity (NAT/firewall traversal) of real-time communication =
services now possible, for &nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
*<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of the Internet (including OTT).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
ere are a lot of Mission Impossible, Pointers, Good arguments and other =
things to consider in the mails from you guys that I have not (yet) =
answered.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Al=
low me to come back to those under THIS THREAD/Subject after having =
figured out how to do it in a structured way, with the aim of collecting =
the summary in a new &quot;Interfacing to QoS&quot; thread, that only =
will relate to how IP/IETF/WebRTC level 3-5 &#8211; *<b>can =
interface</b>* to the QoS-stuff (whether within IETF, in the telephony =
world or elsewhere e.g. IEEE Ethernet, mainly relating to lower =
level).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
think this is doable *<b>without</b>* ending up in any of the =
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
),</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i),</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
<b>(iii)</b></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>-t=
hings <o:p></o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time traffic.<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. That all networks *<b>will be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time usage<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>*<=
b>while</b>* newer, fiber only type of networks, still can borrow =
bandwidth at no extra cost, *<b>by proprietary usage</b>* of RFC 5285 by =
browsers. <o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii)</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may have to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(?=
) and possible other Evil -things.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
refer to <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html</a> =
for these &#8220;-things&#8221; and further.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Jo=
hn, Alan, Henry (Sinnreich), Henning, Richard, Cullen, J=F6rgen =
(Bj=F6rkner), Lars (Berggren) and other good friends since the early SIP =
age (before SIP was &#8220;hijacked&#8221; into application specific =
network and proprietary usage and nowadays rarely is used the IETF-way): =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
want to point out how &#8220;confusing&#8221; the QoS-issue has been and =
as hard as I will the fight Mission Impossible attitude, I will also =
fight the</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
&#8220;it is all about bandwidth&#8221; and &#8220;it will go away with =
time&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
attitude&#8230; =A0(also in the above http =
reference)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
agree with parts of what John says below but the remedy to achieve the =
Good (1), (2), (3)-things instead of the Evil </span><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
),</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i),</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
<b>(iii)</b></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>-t=
hings, is NOT to ignore QoS-things &#8211; It is the opposite! =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
r you having more and relevant input &#8211; That has not been said, and =
that you think I will not address anyway &#8211; in this =
&quot;Interfacing to QoS&quot; path, please provide, but I will not be =
able to process much more than already have been said &#8211; and I will =
use http back-pointers for what has already been discussed in more =
detail.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>If=
 you want to point out QoS-stuff Requiring More Input-from/Feedback-to =
the level 3 and above stuff IN ADDITION to what already has been =
discussed here <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00302.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00302.html</a> =
=A0<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease also point out WHAT and WHY that can/or cannot be handled by the =
methods proposed here and whether other existing methods can be used to =
achieve the mission of &quot;Interfacing to QoS&quot; from IP =3D level =
3 to lower levels.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
will do some thinking about what LISTs different things shall be brought =
to and then split the [TRAM] [WEBRTC] and advice regarding involving =
other list will be considered. Such will probably show up and happen =
during this mission.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>An=
other thing from <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a>: =
What name <o:p></o:p></span></p><pre><span lang=3DEN-US>Footnote =
**[Karl]* For similar reasons I have started to say things like =
&#8220;the box containing the TURN server&#8221;. We should not change =
the TURN server spec into including specific QoS methods to apply (like =
reservation, changing DSCP bits or applying traffic shaping mechanism). =
What is happening here is that we share the traffic information that the =
TURN server gets hold of, with a QoS applying function (that will be =
different for different networks) in &#8220;the same box&#8221;. With =
&#8220;the same box&#8221; I mean that we for now don&#8217;t care about =
the interface/commands for sharing information between the TURN server =
and the QoS applying function. There may be a future need to =
specify/standardize this, but if we start doing that now and in TRAM, we =
risk ending up in a 10-year process before having something in place =
(which without a spec automatically will happen in vendor&#8217;s =
product development, by putting the TURN server into already available =
&#8220;QoS applying function&#8221; boxes (like firewalls or the mobile =
DPI/PCRF combination).<o:p></o:p></span></pre><pre><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></pre><pre><span lang=3DEN-US>*What =
we need to consider and specify, is only what traffic info is required =
(to be provided by the application/browser) for &#8220;QoS applying =
functions&#8221; in different networks (including the very common =
reservation types) to do job (i.e. giving us good QoE for real-time =
applications).*<o:p></o:p></span></pre><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>An=
d Henry, I will have to continue to use the S(BC)-word and the =
I(MS)-word, since you did not succeed to &#8220;eradicate&#8221; them =
;-) =A0Let&#8217;s hope we find a word for the above box (quite =
undefined on the QoS-side &#8211; It is only the TURN side and traffic =
info we are up to designing here) that does not need to be =
&#8220;eradicated&#8221;. So no WebRTC-SBC I guess (hopefully it will =
not even be WebRTC-specific), and ICE-SBC sounds crazy (since ICE was =
developed to remove the need of SIP SBCs), the same applies to TURN-SBC =
(which implies that this box hopefully is not even ICE-specific). =
TURN-QoS-Interface box is relevant but long&#8230; =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>PS=
: To avoid being stopped by the &#8220;to many recipients&#8221; spam =
filter, the Guys (including Mary) from <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d the ones mentioned in the email are copied =
separately.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Yoakum, =
John H (John) [mailto:yoakum@avaya.com] <br><b>Skickat:</b> den 20 =
februari 2014 19:56<br><b>Till:</b> Oleg Moskalenko; Alan =
Johnston<br><b>Kopia:</b> Karl Stahl; tram@ietf.org<br><b>=C4mne:</b> =
RE: [tram] Fwd: I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p></div></div=
><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I fully agree with the comments that QOS should be a low priority for =
the initial focus of the TRAM efforts.&nbsp; There are other groups =
doing QOS work and frankly I engage in WebRTC multimedia interactions =
daily over the Internet, enterprise VPNs, and various combinations and =
seldom suffer egregious quality issues.&nbsp; I am more concerned about =
carriers doing things to regulate or degrade WebRTC flows than a failure =
of existing Internet mechanisms to enable them.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Significant focus on QOS before we better enable TURN to be easily =
used in a browser environment taking advantage or normal web =
characteristics (as opposed to historic telephony constructs) would seem =
to be highly distracting at this point.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Ch=
eers,<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><i><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Jo=
hn<o:p></o:p></span></i></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:navy'><o:=
p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:red'>AVAY=
A</span><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><br></span><span lang=3DEN-US =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif";color:teal'>1.9=
19.425.8446</span><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><span lang=3DEN-US =
style=3D'font-size:8.0pt;font-family:"Arial","sans-serif";color:red'><o:p=
></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> tram [<a =
href=3D"mailto:tram-bounces@ietf.org">mailto:tram-bounces@ietf.org</a>] =
<b>On Behalf Of </b>Oleg Moskalenko<br><b>Sent:</b> Thursday, February =
20, 2014 12:43 PM<br><b>To:</b> Alan Johnston<br><b>Cc:</b> Karl Stahl; =
<a href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><b>Subject:</b> =
Re: [tram] Fwd: I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p><div><div><=
div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:blue'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>On =
Thu, Feb 20, 2014 at 6:26 AM, Alan Johnston &lt;<a =
href=3D"mailto:alan.b.johnston@gmail.com" =
target=3D"_blank">alan.b.johnston@gmail.com</a>&gt; =
wrote:<o:p></o:p></span></p><div><div><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US>Personally, I am not sure how much =
QoS is actually in scope for TRAM. Have you been following RMCAT where =
congestion avoidance for RTP is being developed? &nbsp;I see some =
overlap in your goals and the goals of that =
work.<o:p></o:p></span></p></div></div></div><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span lang=3DEN-US>I'd concentrate on the =
TURN application-level functionality, for now, and I'd leave QoS for the =
future discussions.<o:p></o:p></span></p></div></div></div></body></html>
------=_NextPart_000_06F9_01CF3312.7C8EE080--


From nobody Wed Feb 26 09:36:45 2014
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Spencer Dawkins <spencerdawkins.ietf@gmail.com>
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Cc: "tram@ietf.org" <tram@ietf.org>, Spencer Dawkins <spencer@wonderhamster.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, IESG IESG <iesg@ietf.org>, Barry Leiba <barryleiba@computer.org>
Subject: Re: [rtcweb] BCP over TURN will not be in scope ... and MPLS over UDP over TURN will not be in scope ... and ...
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Hi Cullen,

I am keeping the cc: as is in order to close this thread but let's try
and avoid cross-posting in future discussions.

Thanks for your input. I agree that explicit charters are better than
implicit ones, especially around areas that can be controversial. Given
that we are not going to recharter the WG in the next few days, we are
going to focus the session in London on items that are *explicitly*
chartered.

After London, if we need to tighten the charter somehow, we will do that.

Cheers,

Gonzalo

On 26/02/2014 10:26 AM, Cullen Jennings (fluffy) wrote:
> 
> On Feb 26, 2014, at 10:52 AM, Spencer Dawkins <spencerdawkins.ietf@gmail.com> wrote:
> 
>> So ... TRAM was chartered last Thursday ...
>>
>> I'm gonna let Gonzalo (Yet Another Soon-to-be Former AD Serving as WG Chair) and Simon actually chair their working group for at least a full week before intruding further, but for now, let's assume that the working group wants to work on the milestones they signed up for last week, and see how badly that assumption blows up, rather than talk about rechartering every time someone proposes a topic that's out of scope.
>>
>> I look forward to attending the first working group session next week, and to balloting on six TRAM drafts before IETF 92.
>>
>> Thanks,
>>
>> Spencer, as the responsible AD
> 
> Makes sense - I get the desire to let some work get done. 
> 
> I worry that every time someone says something is is not in scope of this WG, it is not going to feel like an open consensus decisions but is instead going to feel like a arbitrary fiat decisions by a chair or IESG made in a dark and smoky room - I hope I am wrong. 
> 
> 


From nobody Wed Feb 26 11:38:33 2014
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Randell,

I agree that most of this discussion belongs elsewhere.
But there are a few bits that probably do belong here.

CLUE needs *something* that provides an e2e reliable channel. It could 
be a lot of things, e.g. TCP or TLS. We decided to go with the 
SCTP-based data channel that RTCWEB is defining so that we had the 
potential for a Webrtc endpoint to participate in CLUE without a gateway 
on the clue channel.

That means we will also be using the same stack between non-webrtc clue 
endpoints that are using SIP for signaling.

We are still undecided on whether to use DCEP or SDP O/A to negotiate 
the channel we use. If we use DCEP it will be straightforward for a 
webrtc endpoint. But that is a little unnatural for the sip endpoints. 
For them, the SDP O/A approach would be more natural.

It doesn't matter much if the *only* channel being used is the clue 
channel. It will start to matter if there is a desire to use other 
channels for features that can coexist with clue. E.g. BFCP. (But now we 
see a proposal to put BFCP over websocket, so maybe it won't ever be an 
issue for us.)

The important thing is that if clue decides to use SDP O/A for this, 
(the ejzak draft) then we must be confident that it will be feasible for 
a webrtc endpoint, via javascript, to participate in this negotiation 
and use it to set up the clue data channel. That is the part that 
belongs in rtcweb.

	Thanks,
	Paul

On 2/25/14 5:41 PM, Randell Jesup wrote:
> On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>>
>> I still don't think the information belong to SDP - they define
>> protocol characteristics. The protocol specification should define
>> re-transmission timers etc.
>>
>> Anyway, we can argue about it later. At this moment I think the focus
>> should be on whether to move forward with the mechanism to begin with,
>> and the main feature of the mechanism - which is about negotiating
>> usage of channels :)
>
> This (whether to move forward) is my concern (in addition to Mary's
> about this being the wrong group - MMUSIC is where the SDP should be and
> has been).
>
> The apparent justification for this document is "This document defines
> SDP-based out-of-band negotiation procedures to establish WebRTC data
> channels for transport of well-defined sub- protocols." This seems a
> pretty sparse justification.
>
> We very purposely avoided using SDP for channel setup (after MUCH
> discussion at Boston and elsewhere).  That isn't to say that for
> non-webrtc usage, one could define an SDP used to externally-negotiate
> the settings for DataChannels, which this appears to be.  But I think
> there should a clearer statement as to "why" we should invest the time
> to standardize this now, not just that we "can".  For example, if CLUE
> really wants to use this instead of in-band opens, then maybe there's a
> reason to do so - but don't expect webrtc endpoints to implement this.
> And if so, it should be done in CLUE by request to MMUSIC.
>
> At this point I don't see need to move forward with this, and certainly
> not here.  And I certainly don't want this to delay finalizing the stuff
> needed in MMUSIC for WebRTC use of DataChannels
>


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Michael,

There have been no replies to my comments:

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11518.html
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11519.html

The first of those is about channels, the 2nd about DCEP. One of my 
issues is that the partitioning between those two drafts is awkward. At 
the least some things from DCEP should be moved to the channel draft. It 
might be better to just merge those two drafts, or do a serious refactoring.

	Thanks,
	Paul


On 2/25/14 6:44 PM, Michael Tuexen wrote:
> Dear all,
>
> Magnus asked me to send a list of open issues regarding data channels
> to the list. Here is my current list:
>
>
> * Priority
>    The W3C hasn't defined it yet. Neither for the (S)RTP media nor for the
>    data channels. We agreed on using a non strict policy for the data channels
>    (some sort of wighted fair queueing). That is all.
>
> * Protocol
>    It seems not to be clear what needs to be provided when registering a
>    (sub)-protocol at IANA. And the name of the registry is unclear...
>
> * SCTP parameters.
>    There was discussed the issue how to set SCTP parameters, especially path.max.retrans
>    and association.max.retrans. Also HB.Interval might be of interest.
>    RFC 4060 recommends path.max.retrans=5, association.max.retrans=10, but has multihoming
>    in mind. To avoid the dormant state, path.max.retrans = association.max.retrans should be used.
>    I would suggest 10 for this value. Should HEARTBEATs be disabled?
>
> * U-C 7: Proxy browsing
>
> * Alternate CC for SCTP
>    Currently there is only the standard CC. However, in some places negotiation of CC is
>    mentioned.
>
> I'm currently going through the backlog of comments regarding the data channels
> ID and I'll try to address the issues. If I find other issues, I send an update
> to the above list.
>
> Best regards
> Michael
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

Let's also keep in mind that there has been suggestions to negotiate the ch=
annels using SDP, but in addition also use DCP (I assume that is what your =
DCEP abbreviation stands for :) to explicitly open the negotiated channels.

But, again, as long as JavaScript can access the information (stream id val=
ue etc), in order to add it to the SDP sent on the wire, and as long as the=
 usage of DCP is optional (it was earlier indicated that it IS optional) I =
don't think there should be any impacts on WebRTC.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
Sent: 26 February 2014 21:38
To: rtcweb@ietf.org
Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-c=
hannel-sdpneg-00

Randell,

I agree that most of this discussion belongs elsewhere.
But there are a few bits that probably do belong here.

CLUE needs *something* that provides an e2e reliable channel. It could be a=
 lot of things, e.g. TCP or TLS. We decided to go with the SCTP-based data =
channel that RTCWEB is defining so that we had the potential for a Webrtc e=
ndpoint to participate in CLUE without a gateway on the clue channel.

That means we will also be using the same stack between non-webrtc clue end=
points that are using SIP for signaling.

We are still undecided on whether to use DCEP or SDP O/A to negotiate the c=
hannel we use. If we use DCEP it will be straightforward for a webrtc endpo=
int. But that is a little unnatural for the sip endpoints.=20
For them, the SDP O/A approach would be more natural.

It doesn't matter much if the *only* channel being used is the clue channel=
. It will start to matter if there is a desire to use other channels for fe=
atures that can coexist with clue. E.g. BFCP. (But now we see a proposal to=
 put BFCP over websocket, so maybe it won't ever be an issue for us.)

The important thing is that if clue decides to use SDP O/A for this, (the e=
jzak draft) then we must be confident that it will be feasible for a webrtc=
 endpoint, via javascript, to participate in this negotiation and use it to=
 set up the clue data channel. That is the part that belongs in rtcweb.

	Thanks,
	Paul

On 2/25/14 5:41 PM, Randell Jesup wrote:
> On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>>
>> I still don't think the information belong to SDP - they define=20
>> protocol characteristics. The protocol specification should define=20
>> re-transmission timers etc.
>>
>> Anyway, we can argue about it later. At this moment I think the focus=20
>> should be on whether to move forward with the mechanism to begin=20
>> with, and the main feature of the mechanism - which is about=20
>> negotiating usage of channels :)
>
> This (whether to move forward) is my concern (in addition to Mary's=20
> about this being the wrong group - MMUSIC is where the SDP should be=20
> and has been).
>
> The apparent justification for this document is "This document defines=20
> SDP-based out-of-band negotiation procedures to establish WebRTC data=20
> channels for transport of well-defined sub- protocols." This seems a=20
> pretty sparse justification.
>
> We very purposely avoided using SDP for channel setup (after MUCH=20
> discussion at Boston and elsewhere).  That isn't to say that for=20
> non-webrtc usage, one could define an SDP used to externally-negotiate=20
> the settings for DataChannels, which this appears to be.  But I think=20
> there should a clearer statement as to "why" we should invest the time=20
> to standardize this now, not just that we "can".  For example, if CLUE=20
> really wants to use this instead of in-band opens, then maybe there's=20
> a reason to do so - but don't expect webrtc endpoints to implement this.
> And if so, it should be done in CLUE by request to MMUSIC.
>
> At this point I don't see need to move forward with this, and=20
> certainly not here.  And I certainly don't want this to delay=20
> finalizing the stuff needed in MMUSIC for WebRTC use of DataChannels
>

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


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On 2/26/2014 2:50 PM, Paul Kyzivat wrote:
> Michael,
>
> There have been no replies to my comments:
>
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11518.html
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11519.html
>
> The first of those is about channels, the 2nd about DCEP. One of my 
> issues is that the partitioning between those two drafts is awkward. 
> At the least some things from DCEP should be moved to the channel 
> draft. It might be better to just merge those two drafts, or do a 
> serious refactoring.

There seem to have been a number of responses to your second posting at 
least.

> How about:

    Each SCTP user message contains a so called Payload Protocol
    Identifier (PPID) that is passed to SCTP by the data channel layer
    and sent to its peer.  This value is used to multiplex WebRTC Data
    Channel Establishment Protocol messages (defined in [I-D.ietf-rtcweb-
    data-protocol]) with messages containing user data on a data
    channel. The PPID is also used to distinguish UTF-8 encoded user
    data and binary encoded user data.

That seems reasonable and clearer.


> * Section 6.5
>
> This section contains:

    Data channels can be opened by using internal or external
    negotiation.  The details are out of scope of this document.

    A simple protocol for internal negotiation is specified in
    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.

> But internal and external negotiation are not defined in this document.
>

 > I *thought* that internal negotiation was by definition negotiation 
by use of
 > rtcweb-data-protocol. (draft-ejzak-mmusic-data-channel-sdpneg-00 
thinks so too,
 > but calls it "in-band negotiation".) There should be a good 
definition of these
 > terms, or reference to one. And more discussion if there can be other 
kinds of
 > internal negotiation. (If so, how would one be chosen?)

Well, the text above says that ietf-rtcweb-data-protocol specifies an 
internal negotiation protocol,
so it's not that far off.  The split does allow someone to use an 
alternative negotiation protocol
(internal or external).

How about:

    Data channels can be opened by using negotiation within the SCTP association or external
    negotiation.  External negotiation is defined as any method which results in an agreement
    as to the parameters of a channel and the creation thereof.
    The details are out of scope of this document.

    A simple protocol for negotiation within the SCTP association is specified in
    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.


Perhaps someone can wordsmith "external negotiation" better.


> * Section 6.6:
>
> Say:

    All data sent on a Channel in both directions MUST be sent over the
    underlying stream using the reliability defined when the Channel was
    opened unless the options are changed, or per-message options are
    specified by a higher level.

 > Is the recipient to consider it an error if messages are received 
with options different from those
 > defined for the channel? Also, is it an error if messages are 
received with a PPID that isn't specified
 > in Section 8? (And what about PPID 50?)

> How are such channel errors to be treated?


I'd propose that if messages are received with options that don't match 
the initial definition, that fact should be ignored and the message 
processed.
If messages with an unknown PID are received, those messages should be 
ignored.

In both cases, logging of such an event to the application (or onerror 
calls in WebRTC W3 JS layers) is allowed but not required. Note that the 
onerror callback is currently under-defined, so this might change.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Subject: Re: [rtcweb] Open data channel issues
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On 2/26/14 4:02 PM, Randell Jesup wrote:
> On 2/26/2014 2:50 PM, Paul Kyzivat wrote:
>> Michael,
>>
>> There have been no replies to my comments:
>>
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11518.html
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11519.html
>>
>> The first of those is about channels, the 2nd about DCEP. One of my
>> issues is that the partitioning between those two drafts is awkward.
>> At the least some things from DCEP should be moved to the channel
>> draft. It might be better to just merge those two drafts, or do a
>> serious refactoring.
>
> There seem to have been a number of responses to your second posting at
> least.

Well - *one*, and my reply. My mistake - I lost track, and only checked 
the first one when I sent this. :-(

>> How about:
>
>     Each SCTP user message contains a so called Payload Protocol
>     Identifier (PPID) that is passed to SCTP by the data channel layer
>     and sent to its peer.  This value is used to multiplex WebRTC Data
>     Channel Establishment Protocol messages (defined in [I-D.ietf-rtcweb-
>     data-protocol]) with messages containing user data on a data
>     channel. The PPID is also used to distinguish UTF-8 encoded user
>     data and binary encoded user data.
>
> That seems reasonable and clearer.

Thanks.

>> * Section 6.5
>>
>> This section contains:
>
>     Data channels can be opened by using internal or external
>     negotiation.  The details are out of scope of this document.
>
>     A simple protocol for internal negotiation is specified in
>     [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>
>> But internal and external negotiation are not defined in this document.
>>
>
>  > I *thought* that internal negotiation was by definition negotiation
> by use of
>  > rtcweb-data-protocol. (draft-ejzak-mmusic-data-channel-sdpneg-00
> thinks so too,
>  > but calls it "in-band negotiation".) There should be a good
> definition of these
>  > terms, or reference to one. And more discussion if there can be other
> kinds of
>  > internal negotiation. (If so, how would one be chosen?)
>
> Well, the text above says that ietf-rtcweb-data-protocol specifies an
> internal negotiation protocol,
> so it's not that far off.  The split does allow someone to use an
> alternative negotiation protocol
> (internal or external).
>
> How about:
>
>     Data channels can be opened by using negotiation within the SCTP
> association or external
>     negotiation.  External negotiation is defined as any method which
> results in an agreement
>     as to the parameters of a channel and the creation thereof.
>     The details are out of scope of this document.
>
>     A simple protocol for negotiation within the SCTP association is
> specified in
>     [I-D.ietf-rtcweb-data-protocol] and MUST be supported.

> Perhaps someone can wordsmith "external negotiation" better.

ISTM the issue is that Internal vs. External isn't the key distinction. 
What is key is that there is one method that MUST be *supported* but 
need not be the only one used. Other mechanisms may be used. It doesn't 
really matter whether they are conducted internally (over the SCTP 
association) or externally (anything else).

E.g., I could use DCEP to establish one channel, and then use my own 
private protocol over that to negotiate other channels. Would that be 
internal or external negotiation? Does it matter?


>> * Section 6.6:
>>
>> Say:
>
>     All data sent on a Channel in both directions MUST be sent over the
>     underlying stream using the reliability defined when the Channel was
>     opened unless the options are changed, or per-message options are
>     specified by a higher level.
>
>  > Is the recipient to consider it an error if messages are received
> with options different from those
>  > defined for the channel? Also, is it an error if messages are
> received with a PPID that isn't specified
>  > in Section 8? (And what about PPID 50?)
>
>> How are such channel errors to be treated?
>
>
> I'd propose that if messages are received with options that don't match
> the initial definition, that fact should be ignored and the message
> processed.
> If messages with an unknown PID are received, those messages should be
> ignored.

I could live with that. It comes to a philosophical issue of whether you 
want to fail soft or fail hard.

Could just explicitly say that behavior when the per-message SCTP 
parameters are inconsistent with the channel is nonconformant to this 
specification and undefined. And that behavior when PPIDS other than 50, 
51, and 54 (did I remember right?) are used is also nonconformant and 
undefined.

Note that the current partitionion makes it hard to specify conforming 
use of PPID 50. From a channel perspective you can use it, but from a 
DCEP perspective you can only use it a certain way.

(This would be easier if the two were merged. Since DCEP is mandatory to 
implement, there seems little reason to keep them separate.

> In both cases, logging of such an event to the application (or onerror
> calls in WebRTC W3 JS layers) is allowed but not required. Note that the
> onerror callback is currently under-defined, so this might change.

If there is a goal to report errors in a defined way then that would 
change things.

	Thanks,
	Paul


From nobody Wed Feb 26 14:21:49 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Open data channel issues
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Hi,

>>> * Section 6.5
>>>
>>> This section contains:
>>
>>     Data channels can be opened by using internal or external
>>     negotiation.  The details are out of scope of this document.
>>
>>     A simple protocol for internal negotiation is specified in
>>     [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>
>>> But internal and external negotiation are not defined in this document.
>>>
>>
>>  > I *thought* that internal negotiation was by definition negotiation=20
>> by use of  > rtcweb-data-protocol.=20
>> (draft-ejzak-mmusic-data-channel-sdpneg-00
>> thinks so too,
>>  > but calls it "in-band negotiation".) There should be a good=20
>> definition of these  > terms, or reference to one. And more discussion=20
>> if there can be other kinds of  > internal negotiation. (If so, how=20
>> would one be chosen?)
>>
>> Well, the text above says that ietf-rtcweb-data-protocol specifies an=20
>> internal negotiation protocol, so it's not that far off.  The split=20
>> does allow someone to use an alternative negotiation protocol=20
>> (internal or external).
>>
>> How about:
>>
>>     Data channels can be opened by using negotiation within the SCTP=20
>> association or external
>>     negotiation.  External negotiation is defined as any method which=20
>> results in an agreement
>>     as to the parameters of a channel and the creation thereof.
>>     The details are out of scope of this document.
>>
>>     A simple protocol for negotiation within the SCTP association is=20
>> specified in
>>     [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>
>> Perhaps someone can wordsmith "external negotiation" better.
>
>ISTM the issue is that Internal vs. External isn't the key distinction.=20
>What is key is that there is one method that MUST be *supported* but need =
not be the only one >used. Other mechanisms may be used. It doesn't really =
matter whether they are conducted >internally (over the SCTP association) o=
r externally (anything else).

I think the MUST-be-supported-method is a separate question.

>E.g., I could use DCEP to establish one channel, and then use my own priva=
te protocol over that to >negotiate other channels. Would that be internal =
or external negotiation? Does it matter?

I guess they would both be in-band methods. Maybe using in-band and out-of-=
band terminology would be more clear - and more aligned with existing termi=
nology.

Regards,

Christer


From nobody Wed Feb 26 17:12:25 2014
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Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/26/14 3:02 PM, Christer Holmberg wrote:
> Hi,
>
> Let's also keep in mind that there has been suggestions to negotiate the channels using SDP, but in addition also use DCP (I assume that is what your DCEP abbreviation stands for :) to explicitly open the negotiated channels.

DCEP is the *new* acronym, from the most recent version.

> But, again, as long as JavaScript can access the information (stream id value etc), in order to add it to the SDP sent on the wire, and as long as the usage of DCP is optional (it was earlier indicated that it IS optional) I don't think there should be any impacts on WebRTC.
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 26 February 2014 21:38
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
>
> Randell,
>
> I agree that most of this discussion belongs elsewhere.
> But there are a few bits that probably do belong here.
>
> CLUE needs *something* that provides an e2e reliable channel. It could be a lot of things, e.g. TCP or TLS. We decided to go with the SCTP-based data channel that RTCWEB is defining so that we had the potential for a Webrtc endpoint to participate in CLUE without a gateway on the clue channel.
>
> That means we will also be using the same stack between non-webrtc clue endpoints that are using SIP for signaling.
>
> We are still undecided on whether to use DCEP or SDP O/A to negotiate the channel we use. If we use DCEP it will be straightforward for a webrtc endpoint. But that is a little unnatural for the sip endpoints.
> For them, the SDP O/A approach would be more natural.
>
> It doesn't matter much if the *only* channel being used is the clue channel. It will start to matter if there is a desire to use other channels for features that can coexist with clue. E.g. BFCP. (But now we see a proposal to put BFCP over websocket, so maybe it won't ever be an issue for us.)
>
> The important thing is that if clue decides to use SDP O/A for this, (the ejzak draft) then we must be confident that it will be feasible for a webrtc endpoint, via javascript, to participate in this negotiation and use it to set up the clue data channel. That is the part that belongs in rtcweb.
>
> 	Thanks,
> 	Paul
>
> On 2/25/14 5:41 PM, Randell Jesup wrote:
>> On 2/24/2014 10:21 AM, Christer Holmberg wrote:
>>>
>>> I still don't think the information belong to SDP - they define
>>> protocol characteristics. The protocol specification should define
>>> re-transmission timers etc.
>>>
>>> Anyway, we can argue about it later. At this moment I think the focus
>>> should be on whether to move forward with the mechanism to begin
>>> with, and the main feature of the mechanism - which is about
>>> negotiating usage of channels :)
>>
>> This (whether to move forward) is my concern (in addition to Mary's
>> about this being the wrong group - MMUSIC is where the SDP should be
>> and has been).
>>
>> The apparent justification for this document is "This document defines
>> SDP-based out-of-band negotiation procedures to establish WebRTC data
>> channels for transport of well-defined sub- protocols." This seems a
>> pretty sparse justification.
>>
>> We very purposely avoided using SDP for channel setup (after MUCH
>> discussion at Boston and elsewhere).  That isn't to say that for
>> non-webrtc usage, one could define an SDP used to externally-negotiate
>> the settings for DataChannels, which this appears to be.  But I think
>> there should a clearer statement as to "why" we should invest the time
>> to standardize this now, not just that we "can".  For example, if CLUE
>> really wants to use this instead of in-band opens, then maybe there's
>> a reason to do so - but don't expect webrtc endpoints to implement this.
>> And if so, it should be done in CLUE by request to MMUSIC.
>>
>> At this point I don't see need to move forward with this, and
>> certainly not here.  And I certainly don't want this to delay
>> finalizing the stuff needed in MMUSIC for WebRTC use of DataChannels
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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On 2/26/14 5:21 PM, Christer Holmberg wrote:
> Hi,
>
>>>> * Section 6.5
>>>>
>>>> This section contains:
>>>
>>>      Data channels can be opened by using internal or external
>>>      negotiation.  The details are out of scope of this document.
>>>
>>>      A simple protocol for internal negotiation is specified in
>>>      [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>>
>>>> But internal and external negotiation are not defined in this document.
>>>>
>>>
>>>   > I *thought* that internal negotiation was by definition negotiation
>>> by use of  > rtcweb-data-protocol.
>>> (draft-ejzak-mmusic-data-channel-sdpneg-00
>>> thinks so too,
>>>   > but calls it "in-band negotiation".) There should be a good
>>> definition of these  > terms, or reference to one. And more discussion
>>> if there can be other kinds of  > internal negotiation. (If so, how
>>> would one be chosen?)
>>>
>>> Well, the text above says that ietf-rtcweb-data-protocol specifies an
>>> internal negotiation protocol, so it's not that far off.  The split
>>> does allow someone to use an alternative negotiation protocol
>>> (internal or external).
>>>
>>> How about:
>>>
>>>      Data channels can be opened by using negotiation within the SCTP
>>> association or external
>>>      negotiation.  External negotiation is defined as any method which
>>> results in an agreement
>>>      as to the parameters of a channel and the creation thereof.
>>>      The details are out of scope of this document.
>>>
>>>      A simple protocol for negotiation within the SCTP association is
>>> specified in
>>>      [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>>
>>> Perhaps someone can wordsmith "external negotiation" better.
>>
>> ISTM the issue is that Internal vs. External isn't the key distinction.
>> What is key is that there is one method that MUST be *supported* but need not be the only one >used. Other mechanisms may be used. It doesn't really matter whether they are conducted >internally (over the SCTP association) or externally (anything else).
>
> I think the MUST-be-supported-method is a separate question.
>
>> E.g., I could use DCEP to establish one channel, and then use my own private protocol over that to >negotiate other channels. Would that be internal or external negotiation? Does it matter?
>
> I guess they would both be in-band methods. Maybe using in-band and out-of-band terminology would be more clear - and more aligned with existing terminology.

My point is that the real distinction is between the one std mechanism 
and everything else. For purposes of discussions here it doesn't matter 
whether the non-std mechanisms are in-band or out of band.

(Of course to the people using them the distinction is significant. But 
it is their problem, not ours.)

	Thanks,
	Paul


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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+ mmusic
- rtcweb (in bcc)

Hi Paul,
I wanted to close this email loop. Please see my inline comments.

> >
> > [Raju] If I understand correctly, these parameters simply define the
> > data channel transport properties. Yes, they may have some correletion
> with
> > subprotocol, but that's not a one-to-one mapping.
> > For example, a given subprotocol could run on a reliable or
> > unreliable transport; depending on app needs it may use unreliable
> > transport and deal with reliablity at subprotocol stack (or ignore it).
>=20
> In some cases. But in general it is easy and common to define a protocol
> that needs a reliable transport, and it won't work without one.
>=20
> What if every application in the world that uses TCP had the option to
> specify if it was reliable or not? How many would screw it up?
>=20
[Raju]=20
Application subprotocol users need to know the transport needs for the prot=
ocol and select a single transport if multiple options exist. This is simil=
ar to how users select TCP or UDP when implementing any Layer5 protocol (e.=
g. SIP) on top of Layer 4.

If APIs abstract this subprotocol then the transport selection can be hidde=
n under the API (if only one choice exists) and API user won't have to spec=
ify a transport.=20

We really appreciate your time in providing valuable comments on these draf=
ts!

Best Regards!
Raju


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+ rtcweb as it impacts DCP draft.

Please see comments inline.

> I would like some clarity about how this interacts with DCEP.
>=20
> IMO these are not mutually exclusive. But they must be used with care.

[Raju] I debated on this with myself before concluding in my earlier
reply that they are mutually exclusive. My mistake, instead of=20
concluding I should have given options. To make this internal or=20
external per data channel granuality we need to add more=20
restrictions/notes to these draft.=20

http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-03#section-4 say=
s:
   "To avoid glare in opening Channels, each side MUST use either even or
   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers."

Where as draft-ejzak-mmusic-data-channel-sdpneg determines this=20
based on SDP offerer (even) and answerer (odd).

What if the SDP answerer becomes TLS client? The stream #id assignments
conflict with SDP offer/answer.
So, to avoid such conflict we need to add the following clarifications:
1. draft-ejzak-mmusic-data-channel-sdpneg
   If DCP is used along with external negotiation, when creating
   data channel the client should always specify the stream id=20
   to use. Leaving selection to data channel stack (DCSK) will=20
   result in conflicts as DCSK (browser) will assign stream ids=20
   per DTLS role.

2. draft-ietf-rtcweb-data-protocol-03
   Item (1) above conflicts with DCP's odd/even rule, for which
   the draft says "MUST" ("each side MUST use either even or=20
   odd Streams").
   This section 4 (and other sections) text has to be updated
   to "SHOULD" or "MAY".

If we all agree to these updates then I am ok to make=20
internal or external negotiation granularity per data channel.

Best Regards
Raju

>=20
> So, when a channel that was in the SDP previously then disappears from
> the SDP it is necessary to close the corresponding channel. But channels
> that were opened via DCEP should not be closed.
>=20
> That also means that one should never mention a DCEP channel in SDP.
>=20
> 	Thanks,
> 	Paul
>=20
> On 2/26/14 3:12 PM, Makaraju, Maridi Raju (Raju) wrote:
> > Please find my comments inline.
> >
> > But, there is no text on rejecting a data channel in the SDP answer.
> >
> > The answer can of course reject the whole m- line, using port zero in
> > the Answer m- line. But, how does the answerer reject an individual
> > sub-protocol data channel?
> >
> > Only attributes for "accepted data channels" are included in the SDP
> > answer, which indicates back to the offerer which ones are rejected
> > (they just aren't there).  This should be made clearer in the text.
> >
> >     Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted
> >     data channels:
> >
> >     "This line MUST be replicated without changes in the SDP answer, if
> >
> >                      the answerer accepts the offered data channel."
> >
> >     I guess one way to reject would be to simply not include the dcmap
> >     attribute in the Answer, and the Offerer then needs to figure out
> >     what was rejected based on the stream values not present in the
> Answer.
> >
> > Yes.  That is what was intended.
> >
> > /[Raju] In addition to the above mentiooed cases, if we all agree we
> > need to add the following cases in the next draft:/
> >
> > */1./*SDP offer has no a=3Ddcmap. This has two use cases:
> >
> > a.Initial SDP offer: no data channel streams (DCS) are open yet.
> >
> > b.Subsequent SDP offer: Remove all previously opened data
> channels/streams.
> >
> > In either case, the SCTP association is kept open (and ICE procedures
> > are performed) even when no data channels are in use.
> >
> > */2./*Creation of new DCS(s) using SDP offer (initial or subsequent)
> >
> > a.Offerer has to create the data channels locally first by doing
> > createDataChannel or equivalent API into data channel stack before
> > sending offer.
> >
> > */3./*SDP offer was rejected
> >
> > a.DCSs are kept per previous SDP offer/answer. This means DCSs that are
> > to be deleted has to be deleted (i.e. calling createDataChannel or
> > equivalent API) only after successful SDP answer.
> >
> > b.If any new DCSs are attempted to be created per new SDP offer, these
> > have to be closed now.
> >
> > */4./*Per
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-
> createdatachannel
> > createDataChannel() API the 'external' negotiation field is per data
> > channel.  But this draft only allows external negotiation to be used fo=
r
> > all or no streams; it can't be selective. This restriction is also
> > needed so that the SDP offer/answer odd/even stream id allocation won't
> > conflict DCP's TLS role based odd/even mechanism.
> >
> > Best Regards!
> >
> > Raju
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Paul Kyzivat" <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Date: Thu, 27 Feb 2014 21:51:11 +0000
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Subject: Re: [rtcweb] [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi,

Isn't it enough to, in the data channel protocol spec, say that the odd/eve=
n rule applies unless the stream id is explicitly negotiated using some oth=
er mechanism?

Or something like that...

Regards,

Christer




-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta Makaraju,=
 Maridi Raju (Raju)
L=E4hetetty: 27. helmikuuta 2014 21:55
Vastaanottaja: Paul Kyzivat; mmusic@ietf.org; rtcweb@ietf.org
Aihe: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejecti=
on of channel

+ rtcweb as it impacts DCP draft.

Please see comments inline.

> I would like some clarity about how this interacts with DCEP.
>=20
> IMO these are not mutually exclusive. But they must be used with care.

[Raju] I debated on this with myself before concluding in my earlier reply =
that they are mutually exclusive. My mistake, instead of concluding I shoul=
d have given options. To make this internal or external per data channel gr=
anuality we need to add more restrictions/notes to these draft.=20

http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-03#section-4 say=
s:
   "To avoid glare in opening Channels, each side MUST use either even or
   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers."

Where as draft-ejzak-mmusic-data-channel-sdpneg determines this based on SD=
P offerer (even) and answerer (odd).

What if the SDP answerer becomes TLS client? The stream #id assignments con=
flict with SDP offer/answer.
So, to avoid such conflict we need to add the following clarifications:
1. draft-ejzak-mmusic-data-channel-sdpneg
   If DCP is used along with external negotiation, when creating
   data channel the client should always specify the stream id=20
   to use. Leaving selection to data channel stack (DCSK) will=20
   result in conflicts as DCSK (browser) will assign stream ids=20
   per DTLS role.

2. draft-ietf-rtcweb-data-protocol-03
   Item (1) above conflicts with DCP's odd/even rule, for which
   the draft says "MUST" ("each side MUST use either even or=20
   odd Streams").
   This section 4 (and other sections) text has to be updated
   to "SHOULD" or "MAY".

If we all agree to these updates then I am ok to make internal or external =
negotiation granularity per data channel.

Best Regards
Raju

>=20
> So, when a channel that was in the SDP previously then disappears from=20
> the SDP it is necessary to close the corresponding channel. But=20
> channels that were opened via DCEP should not be closed.
>=20
> That also means that one should never mention a DCEP channel in SDP.
>=20
> 	Thanks,
> 	Paul
>=20
> On 2/26/14 3:12 PM, Makaraju, Maridi Raju (Raju) wrote:
> > Please find my comments inline.
> >
> > But, there is no text on rejecting a data channel in the SDP answer.
> >
> > The answer can of course reject the whole m- line, using port zero=20
> > in the Answer m- line. But, how does the answerer reject an=20
> > individual sub-protocol data channel?
> >
> > Only attributes for "accepted data channels" are included in the SDP=20
> > answer, which indicates back to the offerer which ones are rejected=20
> > (they just aren't there).  This should be made clearer in the text.
> >
> >     Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted
> >     data channels:
> >
> >     "This line MUST be replicated without changes in the SDP answer,=20
> > if
> >
> >                      the answerer accepts the offered data channel."
> >
> >     I guess one way to reject would be to simply not include the dcmap
> >     attribute in the Answer, and the Offerer then needs to figure out
> >     what was rejected based on the stream values not present in the
> Answer.
> >
> > Yes.  That is what was intended.
> >
> > /[Raju] In addition to the above mentiooed cases, if we all agree we=20
> > need to add the following cases in the next draft:/
> >
> > */1./*SDP offer has no a=3Ddcmap. This has two use cases:
> >
> > a.Initial SDP offer: no data channel streams (DCS) are open yet.
> >
> > b.Subsequent SDP offer: Remove all previously opened data
> channels/streams.
> >
> > In either case, the SCTP association is kept open (and ICE=20
> > procedures are performed) even when no data channels are in use.
> >
> > */2./*Creation of new DCS(s) using SDP offer (initial or subsequent)
> >
> > a.Offerer has to create the data channels locally first by doing=20
> > createDataChannel or equivalent API into data channel stack before=20
> > sending offer.
> >
> > */3./*SDP offer was rejected
> >
> > a.DCSs are kept per previous SDP offer/answer. This means DCSs that=20
> > are to be deleted has to be deleted (i.e. calling createDataChannel=20
> > or equivalent API) only after successful SDP answer.
> >
> > b.If any new DCSs are attempted to be created per new SDP offer,=20
> > these have to be closed now.
> >
> > */4./*Per
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-
> createdatachannel
> > createDataChannel() API the 'external' negotiation field is per data=20
> > channel.  But this draft only allows external negotiation to be used=20
> > for all or no streams; it can't be selective. This restriction is=20
> > also needed so that the SDP offer/answer odd/even stream id=20
> > allocation won't conflict DCP's TLS role based odd/even mechanism.
> >
> > Best Regards!
> >
> > Raju
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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


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Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi Christer,

> Hi,
>=20
> Isn't it enough to, in the data channel protocol spec, say that the odd/e=
ven
> rule applies unless the stream id is explicitly negotiated using some oth=
er
> mechanism?
>=20
[Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-chan=
nel-sdpneg should still need to say that "when both external and DCP are us=
ed, even for DCP created streams the stream ids have to be specified and ma=
naged by the application (else DCP stack will default to odd/even rule per =
DTLS role which may conflict with SDP offer/answer rule)".

Thanks for agreeing to add this to the draft!

Raju


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Raju,

On 2/27/14 5:59 PM, Makaraju, Maridi Raju (Raju) wrote:
> Hi Christer,
>
>> Hi,
>>
>> Isn't it enough to, in the data channel protocol spec, say that the odd/even
>> rule applies unless the stream id is explicitly negotiated using some other
>> mechanism?
>>
> [Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-channel-sdpneg should still need to say that "when both external and DCP are used, even for DCP created streams the stream ids have to be specified and managed by the application (else DCP stack will default to odd/even rule per DTLS role which may conflict with SDP offer/answer rule)".

Can we assume there must be a mechanism for using DCEP and still 
controlling which channels are used?

It might help if the O/A negotiation also followed the even/odd rule.
But there are difficulties in knowing who is even and who is odd for the 
first O/A.

If the offerer always creates the channel before sending the offer, then 
that will prevent the two mechanisms from stepping on one another's 
stream ids.

	Thanks,
	Paul

> Thanks for agreeing to add this to the draft!
>
> Raju
>
>


From nobody Fri Feb 28 08:46:52 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Date: Fri, 28 Feb 2014 16:46:33 +0000
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Subject: Re: [rtcweb] [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi Paul,


> >> Isn't it enough to, in the data channel protocol spec, say that the
> odd/even
> >> rule applies unless the stream id is explicitly negotiated using some
> other
> >> mechanism?
> >>
> > [Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-
> channel-sdpneg should still need to say that "when both external and DCP =
are
> used, even for DCP created streams the stream ids have to be specified an=
d
> managed by the application (else DCP stack will default to odd/even rule =
per
> DTLS role which may conflict with SDP offer/answer rule)".
>=20
> Can we assume there must be a mechanism for using DCEP and still
> controlling which channels are used?

[Raju] Yes, we can assume that and it is already supported by :
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.5
and
http://dev.w3.org/2011/webrtc/editor/webrtc.html#widl-RTCDataChannel-id
I believe it is safe to assume non-browser implementations of=20
data channel stack gives an option for application to select=20
stream ids independent of DCEP or external negotiation.


>=20
> It might help if the O/A negotiation also followed the even/odd rule.
[Raju] May be I am missing something here. Richard's draft already has
a specific even/odd rule for O/A. But the issue is TLS based rule may
conflict with the O/A rule as TLS roles may change depending on a=3Dsetup.

> But there are difficulties in knowing who is even and who is odd for the
> first O/A.

[Raju] Not related to first or subsequent O/A, but rather a conflict=20
between TLS roles vs. O/A.

>=20
> If the offerer always creates the channel before sending the offer, then
> that will prevent the two mechanisms from stepping on one another's
> stream ids.
[Raju]=20
In some cases, initial offer may not have a data channel stream created.=20
I don't know how this can prevent stepping on each other?!
May be I am missing something here?! In most cases TLS role is=20
determined after O/A is completed, but offerer creates a data=20
channel before answer, then how does it know even/odd rule?=20
So, trying to use TLS role for external negotiation is problematic.

Thanks
Raju

