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From: Tim Panton <tim@phonefromhere.com>
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I (personally) am against making g711 or g722 MTI audio codecs. There are ne=
twork environments where they will behave significantly worse than opus (in s=
ilk mode). Neither codec will adapt to variable available bandwidth or cope w=
ell with packet loss -as seen on the edge of wifi networks and on 3G network=
s .=20

We should not be encouraging the use of legacy codecs by making them MTI.=20=


When I mentioned H261 as a compromise MTI video codec, I was told that bette=
r codecs existed and that we should adopt them. Same goes for audio.=20

Tim. =20

Sent from my iPhone

On 30 Jul 2012, at 20:48, "Kevin P. Fleming" <kpfleming@digium.com> wrote:

> On 07/30/2012 12:21 PM, Roman Shpount wrote:
>> Should we consider adding G.722 as a mandatory to implement audio codec?
>> It is low complexity and carries no royalty or license restrictions, it
>> is very good quality for voice audio communications at the
>> same bandwidths as G.711, and it is very widely implemented by the
>> desktop IP phones.
>=20
> I would certainly support making G.722 mandatory to implement.
>=20
> --=20
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming=

> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From roman@telurix.com  Wed Aug  1 08:24:18 2012
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Subject: Re: [rtcweb] Should we also make G.722 a mandatory to implement codec?
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I personally think it would be a mistake not to require G.711 and G.722.
Even though OPUS can be better under some circumstances, it is much more
expensive to encode, has no deployed base, and might proof to be patent
encumbered.

The CPU resources to encode or transcode to OPUS are several orders of
magnitude more then for G.711 and G.722. If we need to communicate with
legacy equipment, the cost to transcode from OPUS to G.711 or G.722 might
be prohibitive. Also, if we need to transcode to other low bitrate codecs,
such as AMR, it would require half the resources to transcode between G.722
and AMR vs OPUS and AMR.

Even though OPUS is supposed to be an unencumbered codec, it is not
unfeasible for some party to step forward and claim intellectual property
rigts that OPUS is using. In this case we might have no recourse but to pay
royalties. G.711 and G.722 are guaranteed not to be encumbered in any way
since they been in use long enough for any IP claims to expire.

Finally, OPUS is still in a draft phase, so there is no or little
deployment experience with it. We might still discover things that will
negatively affect it in some way. G.711 and G.722 are very widely deployed
and all the issues related to it are well known. In a lot of cases they are
simply good enough.

P.S. This is in no way an argument against OPUS. I think we should make it
mandatory and hope for the best, but I think we should also have easy to
implement, widely deployed, unencumbered codecs supported as well.
_____________
Roman Shpount


On Wed, Aug 1, 2012 at 6:33 AM, Tim Panton <tim@phonefromhere.com> wrote:

> I (personally) am against making g711 or g722 MTI audio codecs. There are
> network environments where they will behave significantly worse than opus
> (in silk mode). Neither codec will adapt to variable available bandwidth or
> cope well with packet loss -as seen on the edge of wifi networks and on 3G
> networks .
>
> We should not be encouraging the use of legacy codecs by making them MTI.
>
> When I mentioned H261 as a compromise MTI video codec, I was told that
> better codecs existed and that we should adopt them. Same goes for audio.
>
> Tim.
>
> Sent from my iPhone
>
> On 30 Jul 2012, at 20:48, "Kevin P. Fleming" <kpfleming@digium.com> wrote:
>
> > On 07/30/2012 12:21 PM, Roman Shpount wrote:
> >> Should we consider adding G.722 as a mandatory to implement audio codec?
> >> It is low complexity and carries no royalty or license restrictions, it
> >> is very good quality for voice audio communications at the
> >> same bandwidths as G.711, and it is very widely implemented by the
> >> desktop IP phones.
> >
> > I would certainly support making G.722 mandatory to implement.
> >
> > --
> > Kevin P. Fleming
> > Digium, Inc. | Director of Software Technologies
> > Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
> kpfleming
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> > Check us out at www.digium.com & www.asterisk.org
> > _______________________________________________
> > 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
>

--e89a8ff245df474fdf04c635e3c8
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I personally think it would be a mistake not to require G.711 and G.722. Ev=
en though OPUS can be better under some circumstances, it is much more expe=
nsive to encode, has no deployed base, and might proof to be patent encumbe=
red.<br>
<br>The CPU resources to encode or transcode to OPUS are several orders of =
magnitude more then for G.711 and G.722. If we need to communicate with leg=
acy equipment, the cost to transcode from OPUS to G.711 or G.722 might be p=
rohibitive. Also, if we need to transcode to other low bitrate codecs, such=
 as AMR, it would require half the resources to transcode between G.722 and=
 AMR vs OPUS and AMR.<br>
<br>Even though OPUS is supposed to be an unencumbered codec, it is not unf=
easible for some party to step forward and claim intellectual property rigt=
s that OPUS is using. In this case we might have no recourse but to pay roy=
alties. G.711 and G.722 are guaranteed not to be encumbered in any way sinc=
e they been in use long enough for any IP claims to expire.<br>
<br>Finally, OPUS is still in a draft phase, so there is no or little deplo=
yment experience with it. We might still discover things that will negative=
ly affect it in some way. G.711 and G.722 are very widely deployed and all =
the issues related to it are well known. In a lot of cases they are simply =
good enough.<br>
<br>P.S. This is in no way an argument against OPUS. I think we should make=
 it mandatory and hope for the best, but I think we should also have easy t=
o implement, widely deployed, unencumbered codecs supported as well.<br cle=
ar=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Wed, Aug 1, 2012 at 6:33 AM, Tim Pant=
on <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere.com" target=3D=
"_blank">tim@phonefromhere.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
I (personally) am against making g711 or g722 MTI audio codecs. There are n=
etwork environments where they will behave significantly worse than opus (i=
n silk mode). Neither codec will adapt to variable available bandwidth or c=
ope well with packet loss -as seen on the edge of wifi networks and on 3G n=
etworks .<br>

<br>
We should not be encouraging the use of legacy codecs by making them MTI.<b=
r>
<br>
When I mentioned H261 as a compromise MTI video codec, I was told that bett=
er codecs existed and that we should adopt them. Same goes for audio.<br>
<br>
Tim.<br>
<br>
Sent from my iPhone<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On 30 Jul 2012, at 20:48, &quot;Kevin P. Fleming&quot; &lt;<a href=3D"mailt=
o:kpfleming@digium.com">kpfleming@digium.com</a>&gt; wrote:<br>
<br>
&gt; On 07/30/2012 12:21 PM, Roman Shpount wrote:<br>
&gt;&gt; Should we consider adding G.722 as a mandatory to implement audio =
codec?<br>
&gt;&gt; It is low complexity and carries no royalty or license restriction=
s, it<br>
&gt;&gt; is very good quality for voice audio communications at the<br>
&gt;&gt; same bandwidths as G.711, and it is very widely implemented by the=
<br>
&gt;&gt; desktop IP phones.<br>
&gt;<br>
&gt; I would certainly support making G.722 mandatory to implement.<br>
&gt;<br>
&gt; --<br>
&gt; Kevin P. Fleming<br>
&gt; Digium, Inc. | Director of Software Technologies<br>
&gt; Jabber: <a href=3D"mailto:kfleming@digium.com">kfleming@digium.com</a>=
 | SIP: <a href=3D"mailto:kpfleming@digium.com">kpfleming@digium.com</a> | =
Skype: kpfleming<br>
&gt; 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
&gt; Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">ww=
w.digium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank=
">www.asterisk.org</a><br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<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>

--e89a8ff245df474fdf04c635e3c8--

From mandyam@quicinc.com  Tue Jul 31 11:05:11 2012
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWeb Data Stream and RTP Synchronization - new I.-D.
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Hi Harald,
Thanks for your feedback.  Our intention was to leverage the latest Editor'=
s Draft of the WebRTC API as is.  This should have been stated more clearly=
 in the I.-D. - I apologize.

Therefore JS does not have access to timestamp data in RTP (AFAICT) with th=
e current PeerConnection API.  Ergo for application-generated data, a JS me=
chanism such as Date().getTime() could be used to embed timestamp data with=
in the application data sent over SCTP.  This is far from perfect and does =
not provide perfect synchronization to an accompanying RTP stream.

We were not intending to extend the data channel mechanism as defined by I-=
D.-jesup-rtcweb-data-protocol to include timestamp data, if that is what yo=
u were referring to.  I'm not sure that solves the problem either, because =
there are no guarantees that such a mechanism would leverage the same times=
tamp generation as the RTP stream would.

We feel that the best way to assure tight synchronization between the RTP s=
tream and any opaque data is through the RTP extension header.

-Giri

From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org]<mailto:[mailto:rtcweb-bounces@ietf.org]> On Behalf Of H=
arald Alvestrand
Sent: Tuesday, July 31, 2012 9:04 AM
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWeb Data Stream and RTP Synchronization - new I.-D=
.

On 07/30/2012 12:11 PM, Mandyam, Giridhar wrote:
Hello,

Please note the internet draft recently uploaded:  http://tools.ietf.org/ht=
ml/draft-mandyam-rtcweb-data-synch-00 entitled "RTCWeb Data Stream and RTP =
Synchronization."   All comments, suggestions, requests for clarification, =
corrections are welcome.


Hello - I scanned your draft quickly, and I see a lot of talk about signali=
ng that synchronization between data tracks and RTP tracks is requested, bu=
t nothing about how you intended to represent timestamps in SCTP.

Could you clarify what timestamps you intended to synchronize?

                 Harald



Thanks,

-Giri Mandyam, Qualcomm Innovation Center




_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Harald,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for your feedback.=
&nbsp; Our intention was to leverage the latest Editor&#8217;s Draft of the=
 WebRTC API as is.&nbsp; This should have been stated more clearly in the
 I.-D. &#8211; I apologize.<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">Therefore JS does not hav=
e access to timestamp data in RTP (AFAICT) with the current PeerConnection =
API.&nbsp; Ergo for application-generated data, a JS mechanism
 such as Date().getTime() could be used to embed timestamp data within the =
application data sent over SCTP.&nbsp; This is far from perfect and does no=
t provide perfect synchronization to an accompanying RTP stream.<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">We were not intending to =
extend the data channel mechanism as defined by I-D.-jesup-rtcweb-data-prot=
ocol to include timestamp data, if that is what you were
 referring to.&nbsp; I&#8217;m not sure that solves the problem either, bec=
ause there are no guarantees that such a mechanism would leverage the same =
timestamp generation as the RTP stream would.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">We feel that the best way=
 to assure tight synchronization between the RTP stream and any opaque data=
 is through the RTP extension header.<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">-Giri<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext">
<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> <a h=
ref=3D"mailto:[mailto:rtcweb-bounces@ietf.org]">
[mailto:rtcweb-bounces@ietf.org]</a> <b>On Behalf Of </b>Harald Alvestrand<=
br>
<b>Sent:</b> Tuesday, July 31, 2012 9:04 AM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] RTCWeb Data Stream and RTP Synchronization - n=
ew I.-D.<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 07/30/2012 12:11 PM, Mandyam, Giridhar wrote:<o:p=
></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hello,</span><o:p></o:p><=
/p>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:#1F497D">Please note the internet draft recently upl=
oaded:&nbsp; </span><a href=3D"http://tools.ietf.org/html/draft-mandyam-rtc=
web-data-synch-00"><span style=3D"font-size:11.0pt;font-family:&quot;Calibr=
i&quot;,&quot;sans-serif&quot;">http://tools.ietf.org/html/draft-mandyam-rt=
cweb-data-synch-00</span></a><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"> entitled &#8220;RT=
CWeb Data Stream and RTP Synchronization.&#8220; &nbsp;&nbsp;All comments, =
suggestions, requests for clarification, corrections are welcome.</span><o:=
p></o:p></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></pre>
</div>
</blockquote>
<p class=3D"MsoNormal">Hello - I scanned your draft quickly, and I see a lo=
t of talk about signaling that synchronization between data tracks and RTP =
tracks is requested, but nothing about how you intended to represent timest=
amps in SCTP.<br>
<br>
Could you clarify what timestamps you intended to synchronize?<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
<br>
<o:p></o:p></p>
<div>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:#1F497D">Thanks,</span><o:p></o:p></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:#1F497D">-Giri Mandyam, Qualcomm Innovation Center</=
span><o:p></o:p></pre>
</div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>rtcweb mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre=
>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
</div>
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From miguelecasassanchez@gmail.com  Thu Aug  2 06:01:13 2012
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Date: Thu, 2 Aug 2012 15:01:03 +0200
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From: Miguel Casas-Sanchez <miguelecasassanchez@gmail.com>
To: David Singer <singer@apple.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Google statement on codecs
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You can count on a +1 from me, for what matters. I agree on the base
statement, but what's more important, I've seen what happens if you
follow down the licesing/royalty way of ITU and the likes, or, like
some of us call here, the lawyer's way... I bet we all know a bit
what's this about.
M

On 1 August 2012 00:12, David Singer <singer@apple.com> wrote:
>
> On Jul 30, 2012, at 1:08 , Randell Jesup <randell-ietf@jesup.org> wrote:
>
>> On 7/29/2012 8:13 PM, Ralph Giles wrote:
>>> On 12-07-29 6:44 PM, Maire Reavy wrote:
>>>
>>>> Mandating encumbered codecswould force open-source projects to violate
>>>> the spec (by not implementing a mandatory codec).  For this reason and
>>>> the reasons Justin cites, we believe Opus, G.711 and VP8 are the best
>>>> mandatory-to-implement choices for the spec.
>>> +1
>>
>
> There are open-source implementations of encumbered codecs (e.g. x264); and I expect that there are proprietary codecs that are believed to be unencumbered :-).  Let's not confuse the two classes.
>
> David Singer
> Multimedia and Software Standards, Apple Inc.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: [rtcweb] NAT behavior heuristics
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In today's RTCWEB meeting I said that NAT heuristics do not work reliably,
especially if a NAT is busy (high CPU or lots of ports consumed), but there
are other situations with a NAT that cause heuristics to be inaccurate.  The
IETF document regarding this is http://tools.ietf.org/html/rfc5780, and be
sure to read its Applicability Statement in Section 1,
http://tools.ietf.org/html/rfc5780#section-1.

An explicit protocol, such Port Control Protocol (PCP, draft-ietf-pcp-base)
is the only reliable way to communicate with a NAT and reduce application
keepalive traffic.  Several of us have noticed the need to document exactly
how PCP can be reliably used to reduce UDP keepalive traffic.  We will write
down those details before the next IETF, probably in an Internet Draft.

-d



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Dear all,

the full recording (synchronized video, audio, slides and jabber room)
of the first RTCWEB session at IETF-84 is available.

You can watch it by accessing the following URL:
http://ietf84.conf.meetecho.com/index.php/Recorded_Sessions#IETF84_RTCWEB_I

For the chair(s): please feel free to put the link to the recording in 
the minutes, if you think this might be useful.

In case of problems with the playout, just drop an e-mail to 
team@meetecho.com.

Cheers,
the Meetecho team

-- 
Meetecho s.r.l.
Web Conferencing and Collaboration Tools
www.meetecho.com

From martin.thomson@gmail.com  Thu Aug  2 16:02:29 2012
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From: Martin Thomson <martin.thomson@gmail.com>
To: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] NAT behavior heuristics
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I assume that this applies only to the NAT that doesn't exist yet and
that we will have to live with status quo (and the current keep-alive
recommendations) until PCP becomes bountiful.

We might want to consider other options for things like power saving
in addition to this.  One option that springs to mind is the ability
to explicitly shut down streams that aren't in use and pay the price
for warm-up.  Optimizations to candidate pair select at warm-up might
be handy in this case.

On 2 August 2012 10:45, Dan Wing <dwing@cisco.com> wrote:
> In today's RTCWEB meeting I said that NAT heuristics do not work reliably,
> especially if a NAT is busy (high CPU or lots of ports consumed), but there
> are other situations with a NAT that cause heuristics to be inaccurate.  The
> IETF document regarding this is http://tools.ietf.org/html/rfc5780, and be
> sure to read its Applicability Statement in Section 1,
> http://tools.ietf.org/html/rfc5780#section-1.
>
> An explicit protocol, such Port Control Protocol (PCP, draft-ietf-pcp-base)
> is the only reliable way to communicate with a NAT and reduce application
> keepalive traffic.  Several of us have noticed the need to document exactly
> how PCP can be reliably used to reduce UDP keepalive traffic.  We will write
> down those details before the next IETF, probably in an Internet Draft.
>
> -d
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From dwing@cisco.com  Thu Aug  2 16:15:47 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Martin Thomson'" <martin.thomson@gmail.com>
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> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: Thursday, August 02, 2012 4:02 PM
> To: Dan Wing
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] NAT behavior heuristics
> 
> I assume that this applies only to the NAT that doesn't exist yet and
> that we will have to live with status quo (and the current keep-alive
> recommendations) until PCP becomes bountiful.

Yes.  PCP is new, somewhat like RTCWEB.

There is an incentive for the existing CGNs, deployed by almost all 
3G/LTE carriers around the world, to have their vendors add PCP 
support to those NATs, as it saves battery lifetime for their 
subscribers and reduces chatter on their network.  Incentives are 
well aligned for that to happen.

I agree that home NATs, enterprise NATs, and enterprise firewalls 
do not have those same incentives.

> We might want to consider other options for things like power saving
> in addition to this.  One option that springs to mind is the ability
> to explicitly shut down streams that aren't in use and pay the price
> for warm-up.  Optimizations to candidate pair select at warm-up might
> be handy in this case.

Such as draft-westerlund-avtext-rtp-stream-pause?

-d


> On 2 August 2012 10:45, Dan Wing <dwing@cisco.com> wrote:
> > In today's RTCWEB meeting I said that NAT heuristics do not work
> reliably,
> > especially if a NAT is busy (high CPU or lots of ports consumed), but
> there
> > are other situations with a NAT that cause heuristics to be
> inaccurate.  The
> > IETF document regarding this is http://tools.ietf.org/html/rfc5780,
> and be
> > sure to read its Applicability Statement in Section 1,
> > http://tools.ietf.org/html/rfc5780#section-1.
> >
> > An explicit protocol, such Port Control Protocol (PCP, draft-ietf-
> pcp-base)
> > is the only reliable way to communicate with a NAT and reduce
> application
> > keepalive traffic.  Several of us have noticed the need to document
> exactly
> > how PCP can be reliably used to reduce UDP keepalive traffic.  We
> will write
> > down those details before the next IETF, probably in an Internet
> Draft.
> >
> > -d
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Thu Aug  2 16:20:55 2012
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On 2 August 2012 16:15, Dan Wing <dwing@cisco.com> wrote:
>> We might want to consider other options for things like power saving
>> in addition to this.  One option that springs to mind is the ability
>> to explicitly shut down streams that aren't in use and pay the price
>> for warm-up.  Optimizations to candidate pair select at warm-up might
>> be handy in this case.
>
> Such as draft-westerlund-avtext-rtp-stream-pause?

That would work at the RTP layer.  That could flow down to the UDP
flow, but resumption is non-trivial.  If you lose the NAT bindings as
a result, you would have to be prepared to restart the flow.  The
(previously) active flow might simply have to represent the highest
priority candidate pair in an ICE reboot.

From dwing@cisco.com  Thu Aug  2 16:46:06 2012
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> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: Thursday, August 02, 2012 4:21 PM
> To: Dan Wing
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] NAT behavior heuristics
> 
> On 2 August 2012 16:15, Dan Wing <dwing@cisco.com> wrote:
> >> We might want to consider other options for things like power saving
> >> in addition to this.  One option that springs to mind is the ability
> >> to explicitly shut down streams that aren't in use and pay the price
> >> for warm-up.  Optimizations to candidate pair select at warm-up
> might
> >> be handy in this case.
> >
> > Such as draft-westerlund-avtext-rtp-stream-pause?
> 
> That would work at the RTP layer.  That could flow down to the UDP
> flow, but resumption is non-trivial.  If you lose the NAT bindings as
> a result, you would have to be prepared to restart the flow.  The
> (previously) active flow might simply have to represent the highest
> priority candidate pair in an ICE reboot.

Thanks.

Perhaps modeling the long-duration stopped flow as a long-duration
mobility event (draft-wing-mmusic-ice-mobility) might be worthwhile;
they feel the same to me at the moment.

-d



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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>, "Dan Wing (dwing)" <dwing@cisco.com>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Subject: Re: [rtcweb] NAT behavior heuristics
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|We might want to consider other options for things like power saving
|in addition to this.  One option that springs to mind is the ability
|to explicitly shut down streams that aren't in use and pay the price
|for warm-up. =20

I think that can be done by slightly tweaking the consent freshness test in=
 draft-muthu-behave-consent-freshness-01. We can also combine it with the s=
ession liveness test and optimize everything. Here is the algorithm I have =
in mind:

The browser starts two timers after ICE concludes:
- A consent timer t1 for 15 sec (configurable in the browser)
- A liveness timer t2 for 5 sec (configurable by the JS)

When t1 expires:
If nothing except consent freshness request(s) was sent in the last x sec t=
hen=20
   goto power_saving_mode.
Else // We has been sending media on the candidate pair.
   If a STUN transaction isn't already active then=20
      Send a consent freshness request.
   If the transaction fails then=20
      goto power_saving_mode.
   Else // The transaction succeeds
      Reset t1.
        =20
When t2 expires:
If nothing was received since the last liveness test then=20
   If a STUN transaction isn't already active then
      Send a liveness request.=20
   If the transaction fails then
      Notify the JS.
   Else // The transaction succeeds
      Reset t2.
     =20
power_saving_mode:
   Stop sending everything on that candidate pair.=20
   Stop both timers.
   Notify the JS.=20

Notes:
The JS is responsible for restarting ICE in the power_saving_mode.
Consent freshness and liveness requests are STUN binding requests.
The default value for t1 is the same as the default for ICE keepalives (sec=
tion 10 of RFC5245).
The default value for x is 60 sec (configurable by the JS).

By carefully choose t1, t2 and x we can:
1. Make it work for existing NATs.
2. Interoperate with existing ICE/ICE-lite implementations.
3. Save battery.
4. Achieve good user experience.

Comments welcome.

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f Martin Thomson
|Sent: Friday, August 03, 2012 4:32 AM
|To: Dan Wing (dwing)
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] NAT behavior heuristics
|
|I assume that this applies only to the NAT that doesn't exist yet and
|that we will have to live with status quo (and the current keep-alive
|recommendations) until PCP becomes bountiful.
|
|We might want to consider other options for things like power saving
|in addition to this.  One option that springs to mind is the ability
|to explicitly shut down streams that aren't in use and pay the price
|for warm-up.  Optimizations to candidate pair select at warm-up might
|be handy in this case.
|
|On 2 August 2012 10:45, Dan Wing <dwing@cisco.com> wrote:
|> In today's RTCWEB meeting I said that NAT heuristics do not work reliabl=
y,
|> especially if a NAT is busy (high CPU or lots of ports consumed), but th=
ere
|> are other situations with a NAT that cause heuristics to be inaccurate. =
 The
|> IETF document regarding this is http://tools.ietf.org/html/rfc5780, and =
be
|> sure to read its Applicability Statement in Section 1,
|> http://tools.ietf.org/html/rfc5780#section-1.
|>
|> An explicit protocol, such Port Control Protocol (PCP, draft-ietf-pcp-ba=
se)
|> is the only reliable way to communicate with a NAT and reduce applicatio=
n
|> keepalive traffic.  Several of us have noticed the need to document exac=
tly
|> how PCP can be reliably used to reduce UDP keepalive traffic.  We will w=
rite
|> down those details before the next IETF, probably in an Internet Draft.
|>
|> -d
|>
|>
|> _______________________________________________
|> 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: Likepeng <likepeng@huawei.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Fw: New Version Notification for draft-li-rtcweb-jsep-xmpp-mapping-00.txt
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Subject: Re: [rtcweb] NAT behavior heuristics
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Don't forget that if you want to shut down a flow according to RTP 
rules, you have to stop sending RTCP on it, which, again according to 
RTP rules, will declare the flow ended after a timeout.

If you want to have the flow "active but not sending", you have to go on 
sending RTCP SR's, reporting that "I still have this flow, but I 
deliberately did not send any packets on it" - which of course will keep 
the radio awake.

If the flow is regarded as ended at all layers, resumption becomes a 
non-issue; you just add a new flow when the time comes.

On 08/03/2012 12:58 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:
> |We might want to consider other options for things like power saving
> |in addition to this.  One option that springs to mind is the ability
> |to explicitly shut down streams that aren't in use and pay the price
> |for warm-up.
>
> I think that can be done by slightly tweaking the consent freshness test in draft-muthu-behave-consent-freshness-01. We can also combine it with the session liveness test and optimize everything. Here is the algorithm I have in mind:
>
> The browser starts two timers after ICE concludes:
> - A consent timer t1 for 15 sec (configurable in the browser)
> - A liveness timer t2 for 5 sec (configurable by the JS)
>
> When t1 expires:
> If nothing except consent freshness request(s) was sent in the last x sec then
>     goto power_saving_mode.
> Else // We has been sending media on the candidate pair.
>     If a STUN transaction isn't already active then
>        Send a consent freshness request.
>     If the transaction fails then
>        goto power_saving_mode.
>     Else // The transaction succeeds
>        Reset t1.
>           
> When t2 expires:
> If nothing was received since the last liveness test then
>     If a STUN transaction isn't already active then
>        Send a liveness request.
>     If the transaction fails then
>        Notify the JS.
>     Else // The transaction succeeds
>        Reset t2.
>        
> power_saving_mode:
>     Stop sending everything on that candidate pair.
>     Stop both timers.
>     Notify the JS.
>
> Notes:
> The JS is responsible for restarting ICE in the power_saving_mode.
> Consent freshness and liveness requests are STUN binding requests.
> The default value for t1 is the same as the default for ICE keepalives (section 10 of RFC5245).
> The default value for x is 60 sec (configurable by the JS).
>
> By carefully choose t1, t2 and x we can:
> 1. Make it work for existing NATs.
> 2. Interoperate with existing ICE/ICE-lite implementations.
> 3. Save battery.
> 4. Achieve good user experience.
>
> Comments welcome.
>
> Muthu
>
> |-----Original Message-----
> |From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin Thomson
> |Sent: Friday, August 03, 2012 4:32 AM
> |To: Dan Wing (dwing)
> |Cc: rtcweb@ietf.org
> |Subject: Re: [rtcweb] NAT behavior heuristics
> |
> |I assume that this applies only to the NAT that doesn't exist yet and
> |that we will have to live with status quo (and the current keep-alive
> |recommendations) until PCP becomes bountiful.
> |
> |We might want to consider other options for things like power saving
> |in addition to this.  One option that springs to mind is the ability
> |to explicitly shut down streams that aren't in use and pay the price
> |for warm-up.  Optimizations to candidate pair select at warm-up might
> |be handy in this case.
> |
> |On 2 August 2012 10:45, Dan Wing <dwing@cisco.com> wrote:
> |> In today's RTCWEB meeting I said that NAT heuristics do not work reliably,
> |> especially if a NAT is busy (high CPU or lots of ports consumed), but there
> |> are other situations with a NAT that cause heuristics to be inaccurate.  The
> |> IETF document regarding this is http://tools.ietf.org/html/rfc5780, and be
> |> sure to read its Applicability Statement in Section 1,
> |> http://tools.ietf.org/html/rfc5780#section-1.
> |>
> |> An explicit protocol, such Port Control Protocol (PCP, draft-ietf-pcp-base)
> |> is the only reliable way to communicate with a NAT and reduce application
> |> keepalive traffic.  Several of us have noticed the need to document exactly
> |> how PCP can be reliably used to reduce UDP keepalive traffic.  We will write
> |> down those details before the next IETF, probably in an Internet Draft.
> |>
> |> -d
> |>
> |>
> |> _______________________________________________
> |> 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 mperumal@cisco.com  Fri Aug  3 17:04:56 2012
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Subject: Re: [rtcweb] NAT behavior heuristics
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|Don't forget that if you want to shut down a flow according to=20
|RTP rules, you have to stop sending RTCP on it, which, again=20
|according to RTP rules, will declare the flow ended after a=20
|timeout.

The idea is to completely shut down the flow. The RTP rules will ensure tha=
t the peer times out and releases its resources.

Muthu=20

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f Harald Alvestrand
|Sent: Friday, August 03, 2012 8:57 PM
|To: rtcweb@ietf.org
|Subject: Re: [rtcweb] NAT behavior heuristics
|
|Don't forget that if you want to shut down a flow according to RTP
|rules, you have to stop sending RTCP on it, which, again according to
|RTP rules, will declare the flow ended after a timeout.
|
|If you want to have the flow "active but not sending", you have to go on
|sending RTCP SR's, reporting that "I still have this flow, but I
|deliberately did not send any packets on it" - which of course will keep
|the radio awake.
|
|If the flow is regarded as ended at all layers, resumption becomes a
|non-issue; you just add a new flow when the time comes.
|
|On 08/03/2012 12:58 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:
|> |We might want to consider other options for things like power saving
|> |in addition to this.  One option that springs to mind is the ability
|> |to explicitly shut down streams that aren't in use and pay the price
|> |for warm-up.
|>
|> I think that can be done by slightly tweaking the consent freshness test=
 in draft-muthu-behave-
|consent-freshness-01. We can also combine it with the session liveness tes=
t and optimize everything.
|Here is the algorithm I have in mind:
|>
|> The browser starts two timers after ICE concludes:
|> - A consent timer t1 for 15 sec (configurable in the browser)
|> - A liveness timer t2 for 5 sec (configurable by the JS)
|>
|> When t1 expires:
|> If nothing except consent freshness request(s) was sent in the last x se=
c then
|>     goto power_saving_mode.
|> Else // We has been sending media on the candidate pair.
|>     If a STUN transaction isn't already active then
|>        Send a consent freshness request.
|>     If the transaction fails then
|>        goto power_saving_mode.
|>     Else // The transaction succeeds
|>        Reset t1.
|>
|> When t2 expires:
|> If nothing was received since the last liveness test then
|>     If a STUN transaction isn't already active then
|>        Send a liveness request.
|>     If the transaction fails then
|>        Notify the JS.
|>     Else // The transaction succeeds
|>        Reset t2.
|>
|> power_saving_mode:
|>     Stop sending everything on that candidate pair.
|>     Stop both timers.
|>     Notify the JS.
|>
|> Notes:
|> The JS is responsible for restarting ICE in the power_saving_mode.
|> Consent freshness and liveness requests are STUN binding requests.
|> The default value for t1 is the same as the default for ICE keepalives (=
section 10 of RFC5245).
|> The default value for x is 60 sec (configurable by the JS).
|>
|> By carefully choose t1, t2 and x we can:
|> 1. Make it work for existing NATs.
|> 2. Interoperate with existing ICE/ICE-lite implementations.
|> 3. Save battery.
|> 4. Achieve good user experience.
|>
|> Comments welcome.
|>
|> Muthu
|>
|> |-----Original Message-----
|> |From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behal=
f Of Martin Thomson
|> |Sent: Friday, August 03, 2012 4:32 AM
|> |To: Dan Wing (dwing)
|> |Cc: rtcweb@ietf.org
|> |Subject: Re: [rtcweb] NAT behavior heuristics
|> |
|> |I assume that this applies only to the NAT that doesn't exist yet and
|> |that we will have to live with status quo (and the current keep-alive
|> |recommendations) until PCP becomes bountiful.
|> |
|> |We might want to consider other options for things like power saving
|> |in addition to this.  One option that springs to mind is the ability
|> |to explicitly shut down streams that aren't in use and pay the price
|> |for warm-up.  Optimizations to candidate pair select at warm-up might
|> |be handy in this case.
|> |
|> |On 2 August 2012 10:45, Dan Wing <dwing@cisco.com> wrote:
|> |> In today's RTCWEB meeting I said that NAT heuristics do not work reli=
ably,
|> |> especially if a NAT is busy (high CPU or lots of ports consumed), but=
 there
|> |> are other situations with a NAT that cause heuristics to be inaccurat=
e.  The
|> |> IETF document regarding this is http://tools.ietf.org/html/rfc5780, a=
nd be
|> |> sure to read its Applicability Statement in Section 1,
|> |> http://tools.ietf.org/html/rfc5780#section-1.
|> |>
|> |> An explicit protocol, such Port Control Protocol (PCP, draft-ietf-pcp=
-base)
|> |> is the only reliable way to communicate with a NAT and reduce applica=
tion
|> |> keepalive traffic.  Several of us have noticed the need to document e=
xactly
|> |> how PCP can be reliably used to reduce UDP keepalive traffic.  We wil=
l write
|> |> down those details before the next IETF, probably in an Internet Draf=
t.
|> |>
|> |> -d
|> |>
|> |>
|> |> _______________________________________________
|> |> rtcweb mailing list
|> |> rtcweb@ietf.org
|> |> https://www.ietf.org/mailman/listinfo/rtcweb
|> |_______________________________________________
|> |rtcweb mailing list
|> |rtcweb@ietf.org
|> |https://www.ietf.org/mailman/listinfo/rtcweb
|> _______________________________________________
|> rtcweb mailing list
|> rtcweb@ietf.org
|> https://www.ietf.org/mailman/listinfo/rtcweb
|
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From randell-ietf@jesup.org  Sun Aug  5 00:20:12 2012
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Subject: Re: [rtcweb] NAT behavior heuristics
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On 8/2/2012 7:15 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>> Sent: Thursday, August 02, 2012 4:02 PM
>> To: Dan Wing
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] NAT behavior heuristics
>>
>> I assume that this applies only to the NAT that doesn't exist yet and
>> that we will have to live with status quo (and the current keep-alive
>> recommendations) until PCP becomes bountiful.
>
> Yes.  PCP is new, somewhat like RTCWEB.
>
> There is an incentive for the existing CGNs, deployed by almost all
> 3G/LTE carriers around the world, to have their vendors add PCP
> support to those NATs, as it saves battery lifetime for their
> subscribers and reduces chatter on their network.  Incentives are
> well aligned for that to happen.
>
> I agree that home NATs, enterprise NATs, and enterprise firewalls
> do not have those same incentives.

And that's a rub, since in many/most cases, the 3G/LTE people will 
likely be talking to non-3G/LTE people, and if either side needs 
keepalives, then radio will be kept active.  Note we're talking 
long-term inactive media flows and an inactive (or rarely active) 
datachannel, such as a client using PeerConnection and DataChannels to 
keep a registration or empty conference alive, or various non-phone-like 
applications.

An alternative mechanism for keepalives might help - you can use 
short-TTL packets to prop the local router without letting the packet go 
all the way to the other end.  If the fixed-station PC uses this TTL 
trick, and the mobile unit uses PCP, the mobile unit can keep its radio off.

Short-TTL can be handy for reducing loads on servers, especially where 
the port needs to stay open with no real traffic for long periods (think 
SIP).

The local router is rarely more than 5-7 hops from a device, though 
there are pathological cases; this could be configured (and 
disableable).  There are also ways to do discovery on the local router; 
these might work better than discovery of UDP port binding time, which 
is known to not work.


-- 
Randell Jesup
randell-ietf@jesup.org

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Subject: Re: [rtcweb] NAT behavior heuristics
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On Aug 5, 2012 12:20 AM, "Randell Jesup" <randell-ietf@jesup.org> wrote:
>
> On 8/2/2012 7:15 PM, Dan Wing wrote:
>>>
>>> -----Original Message-----
>>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>>> Sent: Thursday, August 02, 2012 4:02 PM
>>> To: Dan Wing
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] NAT behavior heuristics
>>>
>>> I assume that this applies only to the NAT that doesn't exist yet and
>>> that we will have to live with status quo (and the current keep-alive
>>> recommendations) until PCP becomes bountiful.
>>
>>
>> Yes.  PCP is new, somewhat like RTCWEB.
>>
>> There is an incentive for the existing CGNs, deployed by almost all
>> 3G/LTE carriers around the world, to have their vendors add PCP
>> support to those NATs, as it saves battery lifetime for their
>> subscribers and reduces chatter on their network.  Incentives are
>> well aligned for that to happen.
>>
>> I agree that home NATs, enterprise NATs, and enterprise firewalls
>> do not have those same incentives.
>
>
> And that's a rub, since in many/most cases, the 3G/LTE people will likely
be talking to non-3G/LTE people, and if either side needs keepalives, then
radio will be kept active.  Note we're talking long-term inactive media
flows and an inactive (or rarely active) datachannel, such as a client
using PeerConnection and DataChannels to keep a registration or empty
conference alive, or various non-phone-like applications.
>
> An alternative mechanism for keepalives might help - you can use
short-TTL packets to prop the local router without letting the packet go
all the way to the other end.  If the fixed-station PC uses this TTL trick,
and the mobile unit uses PCP, the mobile unit can keep its radio off.
>

Fyi. I have not seen any traction for pcp anywhere in the mobile space.
Not on host and not on the CGN. I would not assume it will catch on. As a
mobile operator i have zero plans for ever supporting it. I have operated
CGN in mobile for years, most mobile operators have, and we don't see
inbound connections to mobiles via the cgn as a requirement (nat44 works
today, and applying polish to crap is not productive)

I would avoid the rtcweb layer going into too much effort to optimize
batteries of devices. The intention is good, but at some point it becomes a
layer violation and suboptimization.

Ipv6 is the solution to these issue.

CB

> Short-TTL can be handy for reducing loads on servers, especially where
the port needs to stay open with no real traffic for long periods (think
SIP).
>
> The local router is rarely more than 5-7 hops from a device, though there
are pathological cases; this could be configured (and disableable).  There
are also ways to do discovery on the local router; these might work better
than discovery of UDP port binding time, which is known to not work.
>
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<p><br>
On Aug 5, 2012 12:20 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 8/2/2012 7:15 PM, Dan Wing wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt; From: Martin Thomson [mailto:<a href=3D"mailto:martin.thomson@=
gmail.com">martin.thomson@gmail.com</a>]<br>
&gt;&gt;&gt; Sent: Thursday, August 02, 2012 4:02 PM<br>
&gt;&gt;&gt; To: Dan Wing<br>
&gt;&gt;&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt; Subject: Re: [rtcweb] NAT behavior heuristics<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I assume that this applies only to the NAT that doesn&#39;t ex=
ist yet and<br>
&gt;&gt;&gt; that we will have to live with status quo (and the current kee=
p-alive<br>
&gt;&gt;&gt; recommendations) until PCP becomes bountiful.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Yes. =A0PCP is new, somewhat like RTCWEB.<br>
&gt;&gt;<br>
&gt;&gt; There is an incentive for the existing CGNs, deployed by almost al=
l<br>
&gt;&gt; 3G/LTE carriers around the world, to have their vendors add PCP<br=
>
&gt;&gt; support to those NATs, as it saves battery lifetime for their<br>
&gt;&gt; subscribers and reduces chatter on their network. =A0Incentives ar=
e<br>
&gt;&gt; well aligned for that to happen.<br>
&gt;&gt;<br>
&gt;&gt; I agree that home NATs, enterprise NATs, and enterprise firewalls<=
br>
&gt;&gt; do not have those same incentives.<br>
&gt;<br>
&gt;<br>
&gt; And that&#39;s a rub, since in many/most cases, the 3G/LTE people will=
 likely be talking to non-3G/LTE people, and if either side needs keepalive=
s, then radio will be kept active. =A0Note we&#39;re talking long-term inac=
tive media flows and an inactive (or rarely active) datachannel, such as a =
client using PeerConnection and DataChannels to keep a registration or empt=
y conference alive, or various non-phone-like applications.<br>

&gt;<br>
&gt; An alternative mechanism for keepalives might help - you can use short=
-TTL packets to prop the local router without letting the packet go all the=
 way to the other end. =A0If the fixed-station PC uses this TTL trick, and =
the mobile unit uses PCP, the mobile unit can keep its radio off.<br>

&gt;</p>
<p>Fyi. I have not seen any traction for pcp anywhere in the mobile space.=
=A0 Not on host and not on the CGN. I would not assume it will catch on. As=
 a mobile operator i have zero plans for ever supporting it. I have operate=
d CGN in mobile for years, most mobile operators have, and we don&#39;t see=
 inbound connections to mobiles via the cgn as a requirement (nat44 works t=
oday, and applying polish to crap is not productive)</p>

<p>I would avoid the rtcweb layer going into too much effort to optimize ba=
tteries of devices. The intention is good, but at some point it becomes a l=
ayer violation and suboptimization.</p>
<p>Ipv6 is the solution to these issue.</p>
<p>CB<br><br></p>
<p>&gt; Short-TTL can be handy for reducing loads on servers, especially wh=
ere the port needs to stay open with no real traffic for long periods (thin=
k SIP).<br>
&gt;<br>
&gt; The local router is rarely more than 5-7 hops from a device, though th=
ere are pathological cases; this could be configured (and disableable). =A0=
There are also ways to do discovery on the local router; these might work b=
etter than discovery of UDP port binding time, which is known to not work.<=
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; 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>
</p>

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] NAT behavior heuristics
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> And that's a rub, since in many/most cases, the 3G/LTE people will
> likely be talking to non-3G/LTE people, and if either side needs
> keepalives, then radio will be kept active.  Note we're talking
> long-term inactive media flows and an inactive (or rarely active)
> datachannel, such as a client using PeerConnection and DataChannels to
> keep a registration or empty conference alive, or various non-phone-
> like applications.

If just 3G/LTE supports PCP and non-3G/LTE does not support PCP -> Any STUN=
 indications coming from the non-3G can be dropped by the Mobile Network it=
self to avoid reaching the Mobile Node, since it knows NAT binding is alive=
 using PCP.=20

--Tiru.

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Randell Jesup
> Sent: Sunday, August 05, 2012 1:18 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] NAT behavior heuristics
>=20
> On 8/2/2012 7:15 PM, Dan Wing wrote:
> >> -----Original Message-----
> >> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> >> Sent: Thursday, August 02, 2012 4:02 PM
> >> To: Dan Wing
> >> Cc: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] NAT behavior heuristics
> >>
> >> I assume that this applies only to the NAT that doesn't exist yet
> and
> >> that we will have to live with status quo (and the current keep-
> alive
> >> recommendations) until PCP becomes bountiful.
> >
> > Yes.  PCP is new, somewhat like RTCWEB.
> >
> > There is an incentive for the existing CGNs, deployed by almost all
> > 3G/LTE carriers around the world, to have their vendors add PCP
> > support to those NATs, as it saves battery lifetime for their
> > subscribers and reduces chatter on their network.  Incentives are
> > well aligned for that to happen.
> >
> > I agree that home NATs, enterprise NATs, and enterprise firewalls
> > do not have those same incentives.
>=20
> And that's a rub, since in many/most cases, the 3G/LTE people will
> likely be talking to non-3G/LTE people, and if either side needs
> keepalives, then radio will be kept active.  Note we're talking
> long-term inactive media flows and an inactive (or rarely active)
> datachannel, such as a client using PeerConnection and DataChannels to
> keep a registration or empty conference alive, or various non-phone-
> like
> applications.
>=20
> An alternative mechanism for keepalives might help - you can use
> short-TTL packets to prop the local router without letting the packet
> go
> all the way to the other end.  If the fixed-station PC uses this TTL
> trick, and the mobile unit uses PCP, the mobile unit can keep its radio
> off.
>=20
> Short-TTL can be handy for reducing loads on servers, especially where
> the port needs to stay open with no real traffic for long periods
> (think
> SIP).
>=20
> The local router is rarely more than 5-7 hops from a device, though
> there are pathological cases; this could be configured (and
> disableable).  There are also ways to do discovery on the local router;
> these might work better than discovery of UDP port binding time, which
> is known to not work.
>=20
>=20
> --
> Randell Jesup
> randell-ietf@jesup.org
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Cameron Byrne <cb.list6@gmail.com>, Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Date: Sun, 5 Aug 2012 22:05:51 +0000
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Subject: Re: [rtcweb] NAT behavior heuristics
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> Fyi. I have not seen any traction for pcp anywhere in the mobile space.

http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01 describes us=
age of PCP in Mobile Deployments.

--Tiru.

On Aug 5, 2012 12:20 AM, "Randell Jesup" <randell-ietf@jesup.org> wrote:
>
> On 8/2/2012 7:15 PM, Dan Wing wrote:
>>>
>>> -----Original Message-----
>>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>>> Sent: Thursday, August 02, 2012 4:02 PM
>>> To: Dan Wing
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] NAT behavior heuristics
>>>
>>> I assume that this applies only to the NAT that doesn't exist yet and
>>> that we will have to live with status quo (and the current keep-alive
>>> recommendations) until PCP becomes bountiful.
>>
>>
>> Yes. =A0PCP is new, somewhat like RTCWEB.
>>
>> There is an incentive for the existing CGNs, deployed by almost all
>> 3G/LTE carriers around the world, to have their vendors add PCP
>> support to those NATs, as it saves battery lifetime for their
>> subscribers and reduces chatter on their network. =A0Incentives are
>> well aligned for that to happen.
>>
>> I agree that home NATs, enterprise NATs, and enterprise firewalls
>> do not have those same incentives.
>
>
> And that's a rub, since in many/most cases, the 3G/LTE people will likely=
 be talking to non-3G/LTE people, and if either side needs keepalives, then=
 radio will be kept active. =A0Note we're talking long-term inactive media =
flows and an inactive (or rarely active) datachannel, such as a client usin=
g PeerConnection and DataChannels to keep a registration or empty conferenc=
e alive, or various non-phone-like applications.
>
> An alternative mechanism for keepalives might help - you can use short-TT=
L packets to prop the local router without letting the packet go all the wa=
y to the other end. =A0If the fixed-station PC uses this TTL trick, and the=
 mobile unit uses PCP, the mobile unit can keep its radio off.
>
Fyi. I have not seen any traction for pcp anywhere in the mobile space.=A0 =
Not on host and not on the CGN. I would not assume it will catch on. As a m=
obile operator i have zero plans for ever supporting it. I have operated CG=
N in mobile for years, most mobile operators have, and we don't see inbound=
 connections to mobiles via the cgn as a requirement (nat44 works today, an=
d applying polish to crap is not productive)
I would avoid the rtcweb layer going into too much effort to optimize batte=
ries of devices. The intention is good, but at some point it becomes a laye=
r violation and suboptimization.
Ipv6 is the solution to these issue.
CB
> Short-TTL can be handy for reducing loads on servers, especially where th=
e port needs to stay open with no real traffic for long periods (think SIP)=
.
>
> The local router is rarely more than 5-7 hops from a device, though there=
 are pathological cases; this could be configured (and disableable). =A0The=
re are also ways to do discovery on the local router; these might work bett=
er than discovery of UDP port binding time, which is known to not work.
>
>
> --=20
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] NAT behavior heuristics
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On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
<tireddy@cisco.com> wrote:
>> Fyi. I have not seen any traction for pcp anywhere in the mobile space.
>
> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01 describes usage of PCP in Mobile Deployments.
>
> --Tiru.
>

Yep, that's  an I-D.

And, it says things like "Without
   the particular considerations , PCP may not provide desirable
   outcomes.  Some default behaviours may even cause negative impacts or
   system failures in a mobile environment.  "

Got an operators with a timeline on supporting it?  How about a mobile
OS that plans to support PCP, because you will need both.  And, that
would be interesting information.

I asked around in Vancouver about PCP, and all the operators i spoke
to said no-plans, no motivation.  But, that was a small sample.

In any event, my only point is that RTCWEB should NOT be counting on
PCP.  If PCP is a tool you have, use it.  My point: it is not a tool
you have.

CB

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Cameron Byrne <cb.list6@gmail.com>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "phdgang@gmail.com" <phdgang@gmail.com>
Subject: Re: [rtcweb] NAT behavior heuristics
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> -----Original Message-----
> From: Cameron Byrne [mailto:cb.list6@gmail.com]
> Sent: Sunday, August 05, 2012 4:22 PM
> To: Tirumaleswar Reddy (tireddy)
> Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
> Subject: Re: [rtcweb] NAT behavior heuristics
>=20
> On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
> <tireddy@cisco.com> wrote:
> >> Fyi. I have not seen any traction for pcp anywhere in the mobile
> space.
> >
> > http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01
> describes usage of PCP in Mobile Deployments.
> >
> > --Tiru.
> >
>=20
> Yep, that's  an I-D.
>=20
> And, it says things like "Without
>    the particular considerations , PCP may not provide desirable
>    outcomes.  Some default behaviours may even cause negative impacts
> or
>    system failures in a mobile environment.  "
>=20
> Got an operators with a timeline on supporting it?  How about a mobile
> OS that plans to support PCP, because you will need both.  And, that
> would be interesting information.

In my conversation Nokia supports PCP and my understanding is it can be imp=
lemented as library in user space without any support from kernel.

>=20
> I asked around in Vancouver about PCP, and all the operators i spoke
> to said no-plans, no motivation.  But, that was a small sample.
>=20
> In any event, my only point is that RTCWEB should NOT be counting on
> PCP.  If PCP is a tool you have, use it.  My point: it is not a tool
> you have.

Yes, If the tool is gets deployed in future it will not just help RTCWEB an=
d various other applications. Even with IPv6 this draft refers to Firewalls=
 deployed by Mobile Networks and Firewalls usually timeout UDP sessions in =
30 seconds.

--Tiru.

>=20
> CB

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Subject: Re: [rtcweb] NAT behavior heuristics
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On 08/05/2012 11:49 PM, Tirumaleswar Reddy (tireddy) wrote:
>> And that's a rub, since in many/most cases, the 3G/LTE people will
>> likely be talking to non-3G/LTE people, and if either side needs
>> keepalives, then radio will be kept active.  Note we're talking
>> long-term inactive media flows and an inactive (or rarely active)
>> datachannel, such as a client using PeerConnection and DataChannels to
>> keep a registration or empty conference alive, or various non-phone-
>> like applications.
> If just 3G/LTE supports PCP and non-3G/LTE does not support PCP -> Any STUN indications coming from the non-3G can be dropped by the Mobile Network itself to avoid reaching the Mobile Node, since it knows NAT binding is alive using PCP.
>
1) the at-the-moment theory is that consent verification will use ICE 
Bind requests, not indications
2) if the mobile network drops these, the end node will assume that 
consent is withdrawn, and stop.

Agreed with those who argue that we shouldn't spend too much energy on 
optimizing for a moving target.


From harald@alvestrand.no  Mon Aug  6 06:58:58 2012
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Date: Mon, 06 Aug 2012 15:58:59 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Comparision between H.264 Baseline and VP8
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--------------080305030808040409060505
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Executing on an action item:

I enclose a presentation that was shown by Mohamad Raad at the recent 
ISO MPEG meeting in Stockholm, comparing VP8 with the "anchors" for the 
IVC project (H.264 Baseline, approximately).

The comparisions to "RFM 1.1" refer to the so-called "Chinese proposal" 
for a royalty-free codec.

Stefan Wenger was there; I'm sure he can give more details on the 
comments that were made, in particular why the "Class D" results were 
seen as suspicious.

                     Harald


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "public-webrtc@w3.org" <public-webrtc@w3.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability 
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Subject: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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I see that Microsoft decided to wait until the W3C and IETF were close to d=
one before putting together a proposal that, if accepted, would explode mos=
t the current works and create maximal delay on this work.=20

http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html




From ldecicco@gmail.com  Mon Aug  6 10:16:50 2012
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Date: Mon, 6 Aug 2012 19:16:49 +0200
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From: Luca De Cicco <ldecicco@gmail.com>
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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--047d7b33d6da0d35a604c69c0bda
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I think that the proposed "customizable response to changing network
quality" is a good idea.
Wiring the adaptation logic into webrtc would make it less flexible to
different needs in particular
applications. Of course a default adaptation logic should be implemented in
the browser.

Cheers,
--
Luca De Cicco, PhD
Politecnico di Bari
Dipartimento di Elettrotecnica ed Elettronica
Via Re David, 200 - Bari - ITALY
Office: +39 080 596 3851


On Mon, Aug 6, 2012 at 6:43 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> I see that Microsoft decided to wait until the W3C and IETF were close to
> done before putting together a proposal that, if accepted, would explode
> most the current works and create maximal delay on this work.
>
> http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b33d6da0d35a604c69c0bda
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<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">I think that the pr=
oposed &quot;customizable response to changing network quality&quot; is a g=
ood idea.=A0</span></font></div>
<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">Wiring the adaptati=
on logic into webrtc would make it less flexible to different needs in part=
icular</span></font></div>
<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">applications. Of co=
urse a default adaptation=A0</span></font><span style=3D"font-size:14px;lin=
e-height:18.983333587646484px;color:rgb(44,44,44);font-family:Helvetica,Ari=
al,sans-serif">logic should be implemented in the browser.</span></div>
<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px"><br></span></font><=
/div><div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><sp=
an style=3D"font-size:14px;line-height:18.983333587646484px">Cheers,</span>=
</font></div>
<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">--</span></font></d=
iv><div>Luca De Cicco, PhD</div>Politecnico di Bari<br>Dipartimento di Elet=
trotecnica ed Elettronica<br>
Via Re David, 200 - Bari - ITALY<br>Office: +39 080 596 3851<br>
<br><br><div class=3D"gmail_quote">On Mon, Aug 6, 2012 at 6:43 PM, Cullen J=
ennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" =
target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
<br>
I see that Microsoft decided to wait until the W3C and IETF were close to d=
one before putting together a proposal that, if accepted, would explode mos=
t the current works and create maximal delay on this work.<br>
<br>
<a href=3D"http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t=
.html" target=3D"_blank">http://blogs.skype.com/en/2012/08/customizable_ubi=
quitous_real_t.html</a><br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br>

--047d7b33d6da0d35a604c69c0bda--

From rohit.puri@tenhands.net  Mon Aug  6 12:27:33 2012
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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Based on our experiences at TenHands Inc. where we are trying to build a RT
video-centric collaboration service, the goal cited in MSFT proposal (
http://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:

"*Flexibility in its support of popular media formats and codecs as well as
openness to future innovation*=97A successful standard cannot be tied to
individual codecs, data formats or scenarios. They may soon be supplanted
by newer versions, which would make such a tightly coupled standard
obsolete just as quickly. The right approach is instead to to support
multiple media formats and to bring the bulk of the logic to the
application layer, enabling developers to innovate. "

sounds like a great idea.

I have not gone through the MSFT proposal yet and do not consider myself to
be a browser expert, but if it were possible to have flexibility in the
webRTC framework to the extent that a client developer could use their own
"algorithms" in the media pipeline, that would be awesome. Concretely, some
algorithms that come to mind include (a) pre-processing of captured media
(b) any choice of codecs (b) bandwidth adaptation (c) post-processing of
decoded media (d) CPU adaptation.  There is probably many more.



On Mon, Aug 6, 2012 at 10:16 AM, Luca De Cicco <ldecicco@gmail.com> wrote:

> I think that the proposed "customizable response to changing network
> quality" is a good idea.
> Wiring the adaptation logic into webrtc would make it less flexible to
> different needs in particular
> applications. Of course a default adaptation logic should be implemented
> in the browser.
>
> Cheers,
> --
> Luca De Cicco, PhD
> Politecnico di Bari
> Dipartimento di Elettrotecnica ed Elettronica
> Via Re David, 200 - Bari - ITALY
> Office: +39 080 596 3851
>
>
>
> On Mon, Aug 6, 2012 at 6:43 PM, Cullen Jennings (fluffy) <fluffy@cisco.co=
m
> > wrote:
>
>>
>> I see that Microsoft decided to wait until the W3C and IETF were close t=
o
>> done before putting together a proposal that, if accepted, would explode
>> most the current works and create maximal delay on this work.
>>
>> http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html
>>
>>
>>
>> _______________________________________________
>> 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
Thanks and best regards,

Rohit Puri (rohit.puri@tenhands.net)
Software Development, TenHands Inc. (www.tenhands.net)
My TenHands URL: http://tenhands.net/rohit.puri@tenhands.net

--e89a8fb1ff1482043704c69ddef6
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

Based on our experiences at TenHands Inc. where we are trying to build a RT=
 video-centric collaboration service, the goal cited in MSFT proposal (<a h=
ref=3D"http://html5labs.com/cu-rtc-web/cu-rtc-web.htm">http://html5labs.com=
/cu-rtc-web/cu-rtc-web.htm</a>), namely:<br>
<br>&quot;<b>Flexibility in its support of popular media formats and codecs=
 as=20
  well as openness to future innovation</b>=97A successful standard cannot =
be=20
  tied to individual codecs, data formats or scenarios. They may soon be=20
  supplanted by newer versions, which would make such a tightly coupled sta=
ndard=20
  obsolete just as quickly. The right approach is instead to to support mul=
tiple=20
  media formats and to bring the bulk of the logic to the application layer=
,=20
  enabling developers to innovate. &quot;<br><br>sounds like a great idea. =
<br><br>I have not gone through the MSFT proposal yet and do not consider m=
yself to be a browser expert, but if it were possible to have flexibility i=
n the webRTC framework to the extent that a client developer could use thei=
r own &quot;algorithms&quot; in the media pipeline, that would be awesome. =
Concretely, some algorithms that come to mind include (a) pre-processing of=
 captured media (b) any choice of codecs (b) bandwidth adaptation (c) post-=
processing of decoded media (d) CPU adaptation.=A0 There is probably many m=
ore. <br>

<br><br><br><div class=3D"gmail_quote">On Mon, Aug 6, 2012 at 10:16 AM, Luc=
a De Cicco <span dir=3D"ltr">&lt;<a href=3D"mailto:ldecicco@gmail.com" targ=
et=3D"_blank">ldecicco@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">

<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">I think that the pr=
oposed &quot;customizable response to changing network quality&quot; is a g=
ood idea.=A0</span></font></div>


<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">Wiring the adaptati=
on logic into webrtc would make it less flexible to different needs in part=
icular</span></font></div>


<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">applications. Of co=
urse a default adaptation=A0</span></font><span style=3D"font-size:14px;lin=
e-height:18.983333587646484px;color:rgb(44,44,44);font-family:Helvetica,Ari=
al,sans-serif">logic should be implemented in the browser.</span></div>


<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px"><br></span></font><=
/div><div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><sp=
an style=3D"font-size:14px;line-height:18.983333587646484px">Cheers,</span>=
</font></div>


<div><font color=3D"#2c2c2c" face=3D"Helvetica, Arial, sans-serif"><span st=
yle=3D"font-size:14px;line-height:18.983333587646484px">--</span></font></d=
iv><div>Luca De Cicco, PhD</div>Politecnico di Bari<br>Dipartimento di Elet=
trotecnica ed Elettronica<br>


Via Re David, 200 - Bari - ITALY<br>Office: <a href=3D"tel:%2B39%20080%2059=
6%203851" value=3D"+390805963851" target=3D"_blank">+39 080 596 3851</a><di=
v><div><br>
<br><br><div class=3D"gmail_quote">On Mon, Aug 6, 2012 at 6:43 PM, Cullen J=
ennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" =
target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">


<br>
I see that Microsoft decided to wait until the W3C and IETF were close to d=
one before putting together a proposal that, if accepted, would explode mos=
t the current works and create maximal delay on this work.<br>
<br>
<a href=3D"http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t=
.html" target=3D"_blank">http://blogs.skype.com/en/2012/08/customizable_ubi=
quitous_real_t.html</a><br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><br>-- <br><div>Thanks and bes=
t regards,</div><div><br></div>Rohit Puri=A0(<a href=3D"mailto:rohit.puri@t=
enhands.net" target=3D"_blank">rohit.puri@tenhands.net</a>)<div>Software De=
velopment, TenHands Inc. (<a href=3D"http://www.tenhands.net" target=3D"_bl=
ank">www.tenhands.net</a>)</div>

<div>My TenHands URL:=A0<a href=3D"http://tenhands.net/rohit.puri@tenhands.=
net" target=3D"_blank">http://tenhands.net/rohit.puri@tenhands.net</a></div=
><br>

--e89a8fb1ff1482043704c69ddef6--

From harald@alvestrand.no  Mon Aug  6 12:43:52 2012
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 08/06/2012 09:27 PM, Rohit Puri wrote:
> Based on our experiences at TenHands Inc. where we are trying to build 
> a RT video-centric collaboration service, the goal cited in MSFT 
> proposal (http://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:
>
> "*Flexibility in its support of popular media formats and codecs as 
> well as openness to future innovation*---A successful standard cannot 
> be tied to individual codecs, data formats or scenarios. They may soon 
> be supplanted by newer versions, which would make such a tightly 
> coupled standard obsolete just as quickly. The right approach is 
> instead to to support multiple media formats and to bring the bulk of 
> the logic to the application layer, enabling developers to innovate. "
>
> sounds like a great idea.
The WG agrees with you that multiple codecs should be supported, but the 
WG long ago came to consensus that there needs to be a 
mandatory-to-implement codec to prevent interoperability failure.
That consensus has been rough at times, but I believe it is still the 
consensus.

See section 2.3 of draft-ietf-rtcweb-overview.



--------------070203080905090407040901
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    <div class="moz-cite-prefix">On 08/06/2012 09:27 PM, Rohit Puri
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAPk5xQv_ZNqo66LfApshWtRrvXuBMscnp3+kY_GMiibgD1BCqw@mail.gmail.com"
      type="cite">Based on our experiences at TenHands Inc. where we are
      trying to build a RT video-centric collaboration service, the goal
      cited in MSFT proposal (<a moz-do-not-send="true"
        href="http://html5labs.com/cu-rtc-web/cu-rtc-web.htm">http://html5labs.com/cu-rtc-web/cu-rtc-web.htm</a>),
      namely:<br>
      <br>
      "<b>Flexibility in its support of popular media formats and codecs
        as well as openness to future innovation</b>&#8212;A successful
      standard cannot be tied to individual codecs, data formats or
      scenarios. They may soon be supplanted by newer versions, which
      would make such a tightly coupled standard obsolete just as
      quickly. The right approach is instead to to support multiple
      media formats and to bring the bulk of the logic to the
      application layer, enabling developers to innovate. "<br>
      <br>
      sounds like a great idea. <br>
    </blockquote>
    The WG agrees with you that multiple codecs should be supported, but
    the WG long ago came to consensus that there needs to be a
    mandatory-to-implement codec to prevent interoperability failure.<br>
    That consensus has been rough at times, but I believe it is still
    the consensus.<br>
    <br>
    See section 2.3 of draft-ietf-rtcweb-overview.<br>
    <br>
    <br>
  </body>
</html>

--------------070203080905090407040901--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Rohit Puri <rohit.puri@tenhands.net>
Thread-Topic: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On Aug 6, 2012, at 1:27 PM, Rohit Puri wrote:

> Based on our experiences at TenHands Inc. where we are trying to build a =
RT video-centric collaboration service, the goal cited in MSFT proposal (ht=
tp://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:
>=20
> "Flexibility in its support of popular media formats and codecs as well a=
s openness to future innovation=97A successful standard cannot be tied to i=
ndividual codecs, data formats or scenarios. They may soon be supplanted by=
 newer versions, which would make such a tightly coupled standard obsolete =
just as quickly. The right approach is instead to to support multiple media=
 formats and to bring the bulk of the logic to the application layer, enabl=
ing developers to innovate. "
>=20
> sounds like a great idea.=20

and every proposal from any company or individual that I can recall being s=
ent to theses WGs has had exactly that property. So it is sort of disappoin=
ting to see Microsoft present it as if their proposal was somehow different=
 in this regards. =

From juberti@google.com  Mon Aug  6 13:20:06 2012
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From: Justin Uberti <juberti@google.com>
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Note also that pre- and post-processing of media are not covered by this
proposal, those concepts are in the domain of MediaStream (and
post-processing is possible today through various methods).

In fact, I think the primary novel features of this proposal are:
- SessionDescriptions are true objects, instead of wrappers around SDP
- Additional control over the ICE Agent

However, no use cases have yet been outlined that require this
functionality.


On Mon, Aug 6, 2012 at 10:00 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Aug 6, 2012, at 1:27 PM, Rohit Puri wrote:
>
> > Based on our experiences at TenHands Inc. where we are trying to build =
a
> RT video-centric collaboration service, the goal cited in MSFT proposal (
> http://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:
> >
> > "Flexibility in its support of popular media formats and codecs as well
> as openness to future innovation=E2=80=94A successful standard cannot be =
tied to
> individual codecs, data formats or scenarios. They may soon be supplanted
> by newer versions, which would make such a tightly coupled standard
> obsolete just as quickly. The right approach is instead to to support
> multiple media formats and to bring the bulk of the logic to the
> application layer, enabling developers to innovate. "
> >
> > sounds like a great idea.
>
> and every proposal from any company or individual that I can recall being
> sent to theses WGs has had exactly that property. So it is sort of
> disappointing to see Microsoft present it as if their proposal was someho=
w
> different in this regards.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--485b397dcfcd6260fc04c69e9aed
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Note also that pre- and post-processing of media are not covered by this pr=
oposal, those concepts are in the domain of MediaStream (and post-processin=
g is possible today through various methods).<div><br></div><div>In fact, I=
 think the primary novel features of this proposal are:</div>

<div>- SessionDescriptions are true objects, instead of wrappers around SDP=
</div><div>- Additional control over the ICE Agent</div><div><br></div><div=
>However, no use cases have yet been outlined that require this functionali=
ty.</div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Aug 6=
, 2012 at 10:00 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=
=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Aug 6, 2012, at 1:27 PM, Rohit Puri wrote:<br>
<br>
&gt; Based on our experiences at TenHands Inc. where we are trying to build=
 a RT video-centric collaboration service, the goal cited in MSFT proposal =
(<a href=3D"http://html5labs.com/cu-rtc-web/cu-rtc-web.htm" target=3D"_blan=
k">http://html5labs.com/cu-rtc-web/cu-rtc-web.htm</a>), namely:<br>


&gt;<br>
&gt; &quot;Flexibility in its support of popular media formats and codecs a=
s well as openness to future innovation=E2=80=94A successful standard canno=
t be tied to individual codecs, data formats or scenarios. They may soon be=
 supplanted by newer versions, which would make such a tightly coupled stan=
dard obsolete just as quickly. The right approach is instead to to support =
multiple media formats and to bring the bulk of the logic to the applicatio=
n layer, enabling developers to innovate. &quot;<br>


&gt;<br>
&gt; sounds like a great idea.<br>
<br>
</div>and every proposal from any company or individual that I can recall b=
eing sent to theses WGs has had exactly that property. So it is sort of dis=
appointing to see Microsoft present it as if their proposal was somehow dif=
ferent in this regards.<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>

--485b397dcfcd6260fc04c69e9aed--

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Date: Mon, 6 Aug 2012 13:49:34 -0700
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From: Rohit Puri <rohit.puri@tenhands.net>
To: Justin Uberti <juberti@google.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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--e89a8f8396e3e82bb604c69f034e
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Having not been actively involved in the effort and realizing that i might
be late to the party, i do not mean to cause distraction. Regardless, I
want to voice out the following high-level opinion based on what we have
been learning at TenHands Inc. (this is one of the reasons for us to
continue with a plugin implementation for our service).

If from the media pipeline perspective, it were possible to have an
architecture like for example, gstreamer, wherein there is a concept of a
processing chain for media and it is possible for the client developer to
provide proprietary processing blocks that suit the use case for their
application better than the default implementation, that would be great.

Perhaps, the MSFT proposal does not talk about this, but two of the tenets
they lay out upfront, namely (a) *Customizable response to changing network
quality and (b) **Flexibility in its support of popular media formats and
codecs as well as openness to future innovation* certainly sound appealing.



On Mon, Aug 6, 2012 at 1:19 PM, Justin Uberti <juberti@google.com> wrote:

> Note also that pre- and post-processing of media are not covered by this
> proposal, those concepts are in the domain of MediaStream (and
> post-processing is possible today through various methods).
>
> In fact, I think the primary novel features of this proposal are:
> - SessionDescriptions are true objects, instead of wrappers around SDP
> - Additional control over the ICE Agent
>
> However, no use cases have yet been outlined that require this
> functionality.
>
>
> On Mon, Aug 6, 2012 at 10:00 PM, Cullen Jennings (fluffy) <
> fluffy@cisco.com> wrote:
>
>>
>> On Aug 6, 2012, at 1:27 PM, Rohit Puri wrote:
>>
>> > Based on our experiences at TenHands Inc. where we are trying to build
>> a RT video-centric collaboration service, the goal cited in MSFT proposa=
l (
>> http://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:
>> >
>> > "Flexibility in its support of popular media formats and codecs as wel=
l
>> as openness to future innovation=97A successful standard cannot be tied =
to
>> individual codecs, data formats or scenarios. They may soon be supplante=
d
>> by newer versions, which would make such a tightly coupled standard
>> obsolete just as quickly. The right approach is instead to to support
>> multiple media formats and to bring the bulk of the logic to the
>> application layer, enabling developers to innovate. "
>> >
>> > sounds like a great idea.
>>
>> and every proposal from any company or individual that I can recall bein=
g
>> sent to theses WGs has had exactly that property. So it is sort of
>> disappointing to see Microsoft present it as if their proposal was someh=
ow
>> different in this regards.
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


--=20
Thanks and best regards,

Rohit Puri (rohit.puri@tenhands.net)
Software Development, TenHands Inc. (www.tenhands.net)
My TenHands URL: http://tenhands.net/rohit.puri@tenhands.net

--e89a8f8396e3e82bb604c69f034e
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

Having not been actively involved in the effort and realizing that i might =
be late to the party, i do not mean to cause distraction. Regardless, I wan=
t to voice out the following high-level opinion based on what we have been =
learning at TenHands Inc. (this is one of the reasons for us to continue wi=
th a plugin implementation for our service).<br>
<br>If from the media pipeline perspective, it were possible to have an arc=
hitecture like for example, gstreamer, wherein there is a concept of a proc=
essing chain for media and it is possible for the client developer to provi=
de proprietary processing blocks that suit the use case for their applicati=
on better than the default implementation, that would be great. <br>
<br>Perhaps, the MSFT proposal does not talk about this, but two of the ten=
ets they lay out upfront, namely (a) <strong>Customizable response to chang=
ing network quality and (b) </strong><strong>Flexibility in its support of =
popular media formats and codecs as=20
  well as openness to future innovation</strong> certainly sound appealing.=
 <br><br><br><br><div class=3D"gmail_quote">On Mon, Aug 6, 2012 at 1:19 PM,=
 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">Note also that pre- and post-processing of m=
edia are not covered by this proposal, those concepts are in the domain of =
MediaStream (and post-processing is possible today through various methods)=
.<div>
<br></div><div>In fact, I think the primary novel features of this proposal=
 are:</div>

<div>- SessionDescriptions are true objects, instead of wrappers around SDP=
</div><div>- Additional control over the ICE Agent</div><div><br></div><div=
>However, no use cases have yet been outlined that require this functionali=
ty.</div>


<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote"><div><div cla=
ss=3D"h5">On Mon, Aug 6, 2012 at 10:00 PM, Cullen Jennings (fluffy) <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy=
@cisco.com</a>&gt;</span> wrote:<br>


</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5"><div><br>
On Aug 6, 2012, at 1:27 PM, Rohit Puri wrote:<br>
<br>
&gt; Based on our experiences at TenHands Inc. where we are trying to build=
 a RT video-centric collaboration service, the goal cited in MSFT proposal =
(<a href=3D"http://html5labs.com/cu-rtc-web/cu-rtc-web.htm" target=3D"_blan=
k">http://html5labs.com/cu-rtc-web/cu-rtc-web.htm</a>), namely:<br>



&gt;<br>
&gt; &quot;Flexibility in its support of popular media formats and codecs a=
s well as openness to future innovation=97A successful standard cannot be t=
ied to individual codecs, data formats or scenarios. They may soon be suppl=
anted by newer versions, which would make such a tightly coupled standard o=
bsolete just as quickly. The right approach is instead to to support multip=
le media formats and to bring the bulk of the logic to the application laye=
r, enabling developers to innovate. &quot;<br>



&gt;<br>
&gt; sounds like a great idea.<br>
<br>
</div>and every proposal from any company or individual that I can recall b=
eing sent to theses WGs has had exactly that property. So it is sort of dis=
appointing to see Microsoft present it as if their proposal was somehow dif=
ferent in this regards.<br>



</div></div><div class=3D"im"><div><div>___________________________________=
____________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></div></blockquote></div><br></div>
</blockquote></div><br><br clear=3D"all"><br>-- <br><div>Thanks and best re=
gards,</div><div><br></div>Rohit Puri=A0(<a href=3D"mailto:rohit.puri@tenha=
nds.net" target=3D"_blank">rohit.puri@tenhands.net</a>)<div>Software Develo=
pment, TenHands Inc. (<a href=3D"http://www.tenhands.net" target=3D"_blank"=
>www.tenhands.net</a>)</div>
<div>My TenHands URL:=A0<a href=3D"http://tenhands.net/rohit.puri@tenhands.=
net" target=3D"_blank">http://tenhands.net/rohit.puri@tenhands.net</a></div=
><br>

--e89a8f8396e3e82bb604c69f034e--

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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 08/06/2012 03:19 PM, Justin Uberti wrote:
> Note also that pre- and post-processing of media are not covered by this
> proposal, those concepts are in the domain of MediaStream (and
> post-processing is possible today through various methods).
>
> In fact, I think the primary novel features of this proposal are:
> - SessionDescriptions are true objects, instead of wrappers around SDP
> - Additional control over the ICE Agent
>
> However, no use cases have yet been outlined that require this
> functionality.

I doubt you'll ever see a use case that *requires* SessionDescription 
objects, since the identical behavior can be achieved (in most cases) by 
handling SDP directly. As has been previously discussed, though, using 
objects would make life easier for JS developers, and might lead to more 
natural solutions to problems that have been recently posted on this 
list (like inspection of offers to determine which parts should be 
accepted and which should not).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From Jim.Barnett@genesyslab.com  Mon Aug  6 14:12:20 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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As I recall, we recently discussed the idea of an API abstraction on top
of the SDP and rejected it.  I think that the reasoning was that it was
an unnecessary complication.

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Kevin P. Fleming
Sent: Monday, August 06, 2012 5:07 PM
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no
signs of offering real world interoperability

On 08/06/2012 03:19 PM, Justin Uberti wrote:
> Note also that pre- and post-processing of media are not covered by=20
> this proposal, those concepts are in the domain of MediaStream (and=20
> post-processing is possible today through various methods).
>
> In fact, I think the primary novel features of this proposal are:
> - SessionDescriptions are true objects, instead of wrappers around SDP
> - Additional control over the ICE Agent
>
> However, no use cases have yet been outlined that require this=20
> functionality.

I doubt you'll ever see a use case that *requires* SessionDescription
objects, since the identical behavior can be achieved (in most cases) by
handling SDP directly. As has been previously discussed, though, using
objects would make life easier for JS developers, and might lead to more
natural solutions to problems that have been recently posted on this
list (like inspection of offers to determine which parts should be
accepted and which should not).

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
www.digium.com & www.asterisk.org
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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On 08/06/2012 04:11 PM, Jim Barnett wrote:
> As I recall, we recently discussed the idea of an API abstraction on top
> of the SDP and rejected it.  I think that the reasoning was that it was
> an unnecessary complication.

Indeed... and yet if it turns out that the abstraction is useful 
somebody will produce one and publish it as a JS library :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From randell-ietf@jesup.org  Mon Aug  6 14:18:37 2012
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On 8/6/2012 1:16 PM, Luca De Cicco wrote:
> I think that the proposed "customizable response to changing network
> quality" is a good idea.
> Wiring the adaptation logic into webrtc would make it less flexible to
> different needs in particular
> applications. Of course a default adaptation logic should be implemented
> in the browser.

Already proposed by me (and we've said here that we will have a way to 
do this from the application).  Here's my proposed API, which I think 
covers exactly this and what you want:

https://www.w3.org/Bugs/Public/show_bug.cgi?id=15861

-- 
Randell Jesup
randell-ietf@jesup.org

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On 8/6/2012 4:49 PM, Rohit Puri wrote:
> If from the media pipeline perspective, it were possible to have an
> architecture like for example, gstreamer, wherein there is a concept of
> a processing chain for media and it is possible for the client developer
> to provide proprietary processing blocks that suit the use case for
> their application better than the default implementation, that would be
> great.

See the MediaStream Processing API proposal:
https://dvcs.w3.org/hg/audio/raw-file/tip/streams/StreamProcessing.html

Note that this was partly an alternative to the Web Audio API, which has 
been adopted, and they plan to integrate some aspects of MediaStream 
Processing into Web Audio, but MediaStream Processing is still relevant, 
especially for video (and will be more so once JS workers can access WebGL).

-- 
Randell Jesup
randell-ietf@jesup.org

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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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=0A=
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Justin=
 Uberti [juberti@google.com]:=0A=
=0A=
> In fact, I think the primary novel features of this proposal are:=0A=
>=0A=
> - SessionDescriptions are true objects, instead of wrappers around SDP=0A=
=0A=
More important, there is no SDP Offer/Answer state machine within the brows=
er. A developer who desires SDP Offer/Answer semantics may implement that i=
n Javascript or server-side. This is as opposed to JSEP, which includes a p=
articular SDP O/A state machine within the browser, thus requiring it to be=
 interoperable with other SDP O/A state machines both within other vendor b=
rowsers as well as existing SIP+SDP equipment and software.=0A=
=0A=
> - Additional control over the ICE Agent=0A=
>=0A=
> However, no use cases have yet been outlined that require this functional=
ity.=0A=
=0A=
I believe that there are several use cases, including interoperability use =
cases, where our proposed solution is more likely to be able to meet all ed=
ge cases.=0A=
=0A=
An example would be recovery from call setup in the face of a browser page =
reload... a case where the state of the browser must be reinitialized, lead=
ing to edge cases where it becomes impossible with JSEP for a developer to =
write Javascript that behaves properly in all cases (because without an off=
er one cannot generate an answer, and once an offer has been generated one =
must not generate another offer until the first offer has been answered, bu=
t in either case there is no longer sufficient information as to how to pro=
ceed). While SDP O/A is a useful abstraction for trunk interconnection of p=
eer endpoints, we do not believe it is a good choice for the Javascript API=
 surface.=0A=
=0A=
Likewise, baking in one ICE implementation is likely to lead to interoperab=
ility failures when it is found that it has subtle incompatibilities with o=
ther deployed applications, whereas our proposal would allow the developer =
of the web site or supporting Javascript library to code around these issue=
s once RTC-capable browsers are already fielded.=0A=
=0A=
In addition to these differences that fall out of not baking in the full IC=
E and SDP O/A state machines, our proposal provides several other capabilit=
ies, including hooks that allow a developer to customize their application'=
s response to changing network conditions... an area which is currently com=
pletely unaddressed in the current WebRTC draft.=0A=
=0A=
I wouldn't be surprised if the existing use-case document(s) is (are) inade=
quate to describe situations where, for instance, a developer might wish to=
 prioritize video quality over frame rate, or drop video in order to contin=
ue audio, but that just means that we all need to provide more input on tho=
se documents as well.=0A=
=0A=
=0A=
Matthew Kaufman=


From juberti@google.com  Mon Aug  6 17:11:49 2012
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From: Justin Uberti <juberti@google.com>
Date: Tue, 7 Aug 2012 02:11:27 +0200
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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--20cf30363ebb17a1fd04c6a1d781
Content-Type: text/plain; charset=UTF-8

On Mon, Aug 6, 2012 at 11:51 PM, Matthew Kaufman
<matthew.kaufman@skype.net>wrote:

>
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of
> Justin Uberti [juberti@google.com]:
>
> > In fact, I think the primary novel features of this proposal are:
> >
> > - SessionDescriptions are true objects, instead of wrappers around SDP
>
> More important, there is no SDP Offer/Answer state machine within the
> browser. A developer who desires SDP Offer/Answer semantics may implement
> that in Javascript or server-side. This is as opposed to JSEP, which
> includes a particular SDP O/A state machine within the browser, thus
> requiring it to be interoperable with other SDP O/A state machines both
> within other vendor browsers as well as existing SIP+SDP equipment and
> software.
>

While there is a simple state machine, with JSEP you can drive the state
machine to any state you want, easily. To say that it bakes in SDP
offer/answer is inaccurate; if you don't want to use offer/answer you can
just set the local or remote description at any time (followed by
reapplying the existing remote/local description to complete the sequence).

This could of course be streamlined, but this could be done as an evolution
of the existing API.


> > - Additional control over the ICE Agent
> >
> > However, no use cases have yet been outlined that require this
> functionality.
>
> I believe that there are several use cases, including interoperability use
> cases, where our proposed solution is more likely to be able to meet all
> edge cases.
>
> An example would be recovery from call setup in the face of a browser page
> reload... a case where the state of the browser must be reinitialized,
> leading to edge cases where it becomes impossible with JSEP for a developer
> to write Javascript that behaves properly in all cases (because without an
> offer one cannot generate an answer, and once an offer has been generated
> one must not generate another offer until the first offer has been
> answered, but in either case there is no longer sufficient information as
> to how to proceed). While SDP O/A is a useful abstraction for trunk
> interconnection of peer endpoints, we do not believe it is a good choice
> for the Javascript API surface.
>
> Likewise, baking in one ICE implementation is likely to lead to
> interoperability failures when it is found that it has subtle
> incompatibilities with other deployed applications, whereas our proposal
> would allow the developer of the web site or supporting Javascript library
> to code around these issues once RTC-capable browsers are already fielded.


> In addition to these differences that fall out of not baking in the full
> ICE and SDP O/A state machines, our proposal provides several other
> capabilities, including hooks that allow a developer to customize their
> application's response to changing network conditions... an area which is
> currently completely unaddressed in the current WebRTC draft.
>

As others have pointed out, these kinds of hooks have been discussed but
haven't yet made it into the current API draft.

>
> I wouldn't be surprised if the existing use-case document(s) is (are)
> inadequate to describe situations where, for instance, a developer might
> wish to prioritize video quality over frame rate, or drop video in order to
> continue audio, but that just means that we all need to provide more input
> on those documents as well.
>
>
> Matthew Kaufman
>

--20cf30363ebb17a1fd04c6a1d781
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, A=
ug 6, 2012 at 11:51 PM, Matthew Kaufman <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:matthew.kaufman@skype.net" target=3D"_blank">matthew.kaufman@skype.net=
</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a=
> [<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
on behalf of Justin Uberti [<a href=3D"mailto:juberti@google.com">juberti@g=
oogle.com</a>]:<br>


<div class=3D"im"><br>
&gt; In fact, I think the primary novel features of this proposal are:<br>
&gt;<br>
&gt; - SessionDescriptions are true objects, instead of wrappers around SDP=
<br>
<br>
</div>More important, there is no SDP Offer/Answer state machine within the=
 browser. A developer who desires SDP Offer/Answer semantics may implement =
that in Javascript or server-side. This is as opposed to JSEP, which includ=
es a particular SDP O/A state machine within the browser, thus requiring it=
 to be interoperable with other SDP O/A state machines both within other ve=
ndor browsers as well as existing SIP+SDP equipment and software.<br>

</blockquote><div><br></div><div>While there is a simple state machine, wit=
h JSEP you can drive the state machine to any state you want, easily. To sa=
y that it bakes in SDP offer/answer is inaccurate; if you don&#39;t want to=
 use offer/answer you can just set the local or remote description at any t=
ime (followed by reapplying the existing remote/local description to comple=
te the sequence).=C2=A0</div>

<div><br></div><div>This could of course be streamlined, but this could be =
done as an evolution of the existing API.</div><div><br></div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">


<div class=3D"im"><br>
&gt; - Additional control over the ICE Agent<br>
&gt;<br>
&gt; However, no use cases have yet been outlined that require this functio=
nality.<br>
<br>
</div>I believe that there are several use cases, including interoperabilit=
y use cases, where our proposed solution is more likely to be able to meet =
all edge cases.<br>
<br>
An example would be recovery from call setup in the face of a browser page =
reload... a case where the state of the browser must be reinitialized, lead=
ing to edge cases where it becomes impossible with JSEP for a developer to =
write Javascript that behaves properly in all cases (because without an off=
er one cannot generate an answer, and once an offer has been generated one =
must not generate another offer until the first offer has been answered, bu=
t in either case there is no longer sufficient information as to how to pro=
ceed). While SDP O/A is a useful abstraction for trunk interconnection of p=
eer endpoints, we do not believe it is a good choice for the Javascript API=
 surface.<br>


<br>
Likewise, baking in one ICE implementation is likely to lead to interoperab=
ility failures when it is found that it has subtle incompatibilities with o=
ther deployed applications, whereas our proposal would allow the developer =
of the web site or supporting Javascript library to code around these issue=
s once RTC-capable browsers are already fielded.</blockquote>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
In addition to these differences that fall out of not baking in the full IC=
E and SDP O/A state machines, our proposal provides several other capabilit=
ies, including hooks that allow a developer to customize their application&=
#39;s response to changing network conditions... an area which is currently=
 completely unaddressed in the current WebRTC draft.<br>

</blockquote><div><br></div><div>As others have pointed out, these kinds of=
 hooks have been discussed but haven&#39;t yet made it into the current API=
 draft.</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex">


<br>
I wouldn&#39;t be surprised if the existing use-case document(s) is (are) i=
nadequate to describe situations where, for instance, a developer might wis=
h to prioritize video quality over frame rate, or drop video in order to co=
ntinue audio, but that just means that we all need to provide more input on=
 those documents as well.<br>


<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
Matthew Kaufman<br>
</font></span></blockquote></div><br></div>

--20cf30363ebb17a1fd04c6a1d781--

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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 2012/08/07 4:27, Rohit Puri wrote:
> Based on our experiences at TenHands Inc. where we are trying to build a RT
> video-centric collaboration service, the goal cited in MSFT proposal (
> http://html5labs.com/cu-rtc-web/cu-rtc-web.htm), namely:

I'm surprised to see this called a W3C Submission when it is not (yet?) 
listed at http://www.w3.org/Submission/ (and it's not yet the 7th in the 
areas of the world where most of the involved people are :-).

Regards,   Martin.

From randell-ietf@jesup.org  Mon Aug  6 18:48:06 2012
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 8/6/2012 5:51 PM, Matthew Kaufman wrote:
> In addition to these differences that fall out of not baking in the
> full ICE and SDP O/A state machines, our proposal provides several
> other capabilities, including hooks that allow a developer to
> customize their application's response to changing network
> conditions... an area which is currently completely unaddressed in the
> current WebRTC draft.

We've said repeatedly that we need such capability, and I have a 
proposed JS API for just that:
https://www.w3.org/Bugs/Public/show_bug.cgi?id=15861

The other side of this is in the hands of the Media Capture Task Force, 
where we've been discussing the best way to modify a MediaStream/Track 
after they're created.  There are two rough proposals, one to re-use the 
constraints API for getUserMedia(), and the other I've proposed 
(verbally only so far, no draft API) uses events and lets them "bubble 
up" the chain of MediaStreams/Tracks from a consumer to a source (which 
could be a remote source, and the application could signal that request 
(such as a resolution or frame-rate change) across to the remote source 
of the stream.  This is relevant even without a remote connection.

> I wouldn't be surprised if the existing use-case document(s) is (are)
> inadequate to describe situations where, for instance, a developer
> might wish to prioritize video quality over frame rate, or drop video
> in order to continue audio, but that just means that we all need to
> provide more input on those documents as well. Matthew Kaufman

I think most of those concerns are covered in what I describe above. 
(For example, the application could override the automatic allocation of 
bits and close a video stream, change the resolution, etc.)

I'd love to your input on these proposals, API suggestions, 
alternatives, etc!  I'm glad you're willing to help flesh out them out 
and bring them to completion, and get them into the spec drafts ASAP.


-- 
Randell Jesup
randell-ietf@jesup.org

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From tim@phonefromhere.com  Tue Aug  7 05:11:43 2012
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Ok, the timing is unfortunate, but can you honestly say that we didn't know t=
hat this was skype/microsofts opinion? We just chose to ignore it because it=
 was inconvenient.=20

Now that it is out there, are we seriously going to ignore a document with _=
those_ authors from a major browser maker and a team with extensive experien=
ce in the field jus because it is late!?!?=20

Tim, speaking for himself.=20

Sent from my iPhone

On 6 Aug 2012, at 19:11, Alexey Aylarov <alexey@zingaya.com> wrote:

> It was easy to expect from company which tries to build its own standards a=
ll the time instead of team work on open technologies. If they want to live i=
n their own universe - it's their choice. We need to continue team work on t=
he future of open web communications.
>=20
> "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:
>=20
>>=20
>> I see that Microsoft decided to wait until the W3C and IETF were close to=
 done before putting together a proposal that, if accepted, would explode mo=
st the current works and create maximal delay on this work.=20
>>=20
>> http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From lorenzo@meetecho.com  Tue Aug  7 05:28:08 2012
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From: Lorenzo Miniero <lorenzo@meetecho.com>
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On Tue, 7 Aug 2012 11:57:39 +0200
Tim Panton <tim@phonefromhere.com> wrote:

> Ok, the timing is unfortunate, but can you honestly say that we didn't know that this was skype/microsofts opinion? We just chose to ignore it because it was inconvenient. 
> 
> Now that it is out there, are we seriously going to ignore a document with _those_ authors from a major browser maker and a team with extensive experience in the field jus because it is late!?!? 
> 


An "unfortunately major" browser maker, I'd say, especially considering its well known history of poor adherence to standards (go tell the poor web developers out there), and this latest untimely post of theirs doesn't seem to change that. RTCWEB wasn't born yesterday, they've had more than a year to contribute, and claiming that a team at Microsoft can do better than 100s people contributing here and in the W3C seems a bit pretentious. I agree that the post probably contains some useful feedback, and ignoring it is unwise, but not it if that means changing too much on what we already have.

L.


> Tim, speaking for himself. 
> 
> Sent from my iPhone
> 
> On 6 Aug 2012, at 19:11, Alexey Aylarov <alexey@zingaya.com> wrote:
> 
> > It was easy to expect from company which tries to build its own standards all the time instead of team work on open technologies. If they want to live in their own universe - it's their choice. We need to continue team work on the future of open web communications.
> > 
> > "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:
> > 
> >> 
> >> I see that Microsoft decided to wait until the W3C and IETF were close to done before putting together a proposal that, if accepted, would explode most the current works and create maximal delay on this work. 
> >> 
> >> http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html
> > _______________________________________________
> > 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


-- 
Lorenzo Miniero, COB

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

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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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It seems to me that this proposal is a continuation of a discussion that
the group has been having for a while over how much to bake into the
browser.  JSEP leaves more to the application than ROAP did, and now
this proposal suggests moving even more into the application.  It seems
to me that the trade-off is clear, and doesn't  involve "openness": the
less we bake into the browser, the more flexibility the app developer
has, and the more work he has to do.  (To link this back to another
discussion we've been having, it has been suggested that thoughtless
developers might miss-use constraints. It seems to me that a developer
who can't use constraints correctly is unlikely to code the ICE state
machine correctly. So, are we worried about that kind of developer or
not?) =20

Justin suggests that JSEP can provide the same flexibility that this
proposal does, at least to a sophisticated developer who understands how
to tweak it. Let's find out. I agree with Stefan that we should  draw up
a detailed list of differences and analyze them. At first glance, it
seems to me that many of the differences are independent of each other,
so that it is not a question of simply accepting or rejecting the entire
Microsoft proposal.

- Jim =20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Alexey Aylarov
Sent: Monday, August 06, 2012 1:11 PM
To: Cullen Jennings (fluffy)
Cc: rtcweb@ietf.org ; public-webrtc@w3.org
Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no
signs of offering real world interoperability

It was easy to expect from company which tries to build its own
standards all the time instead of team work on open technologies. If
they want to live in their own universe - it's their choice. We need to
continue team work on the future of open web communications.

"Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:

>
>I see that Microsoft decided to wait until the W3C and IETF were close
to done before putting together a proposal that, if accepted, would
explode most the current works and create maximal delay on this work.=20
>
>http://blogs.skype.com/en/2012/08/customizable_ubiquitous_real_t.html
>
>
>
>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From Markus.Isomaki@nokia.com  Tue Aug  7 07:13:28 2012
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From: <Markus.Isomaki@nokia.com>
To: <cb.list6@gmail.com>, <tireddy@cisco.com>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Subject: Re: [rtcweb] NAT behavior heuristics
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Hi Cameron,

Cameron Byrne wrote:
>
>I asked around in Vancouver about PCP, and all the operators i spoke to sa=
id
>no-plans, no motivation.  But, that was a small sample.

This is probably a fair assessment about the concrete plans. Most mobile op=
erators don't even follow IETF much, so I doubt that they have even heard a=
bout PCP so far. There are however a few operators, where at least the rese=
arch and standardization people have been interested in PCP (as witnessed b=
y http://datatracker.ietf.org/doc/draft-chen-pcp-mobile-deployment/). I thi=
nk some operators are also proposing PCP to be included in the 3GPP specs (=
release 12). It is hard to say how much that means in the real world, but a=
t least I'd expect we could find a few operators willing to do real-network=
 trials withing the next 12 months.=20

Personally I see PCP as a very useful tool in the cellular access environme=
nt, for probing the NAT/FW timers and in the best case even setting them. I=
t will be a major chicken-and-egg situation between apps, mobile OS's and t=
he networks, to get anything into use. I think we should still do our best =
to overcome it. I agree that RTCWeb or anyone else should not rely on PCP (=
or any similar mechanim), but have the ability to make use of it if it happ=
ens to be available.=20

I'm unconvinced IPv6 will help at all in this case. How do you know how lon=
g the FW on the path will keep its state up without keepalives? It will pro=
bably be the same as for NATs: In some networks timeout is 2 minutes, while=
 in others 60 minutes. Having something like PCP to check out that it indee=
d is 60 min. would be extremely helpful. While most app developers might no=
t be willing/able to optimize for battery/mobile, there are already some th=
at are doing a good job. So there is hope that the new mechanisms would get=
 some traction.

Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Cameron Byrne
>Sent: 06 August, 2012 01:22
>To: Tirumaleswar Reddy (tireddy)
>Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
>Subject: Re: [rtcweb] NAT behavior heuristics
>
>On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
><tireddy@cisco.com> wrote:
>>> Fyi. I have not seen any traction for pcp anywhere in the mobile space.
>>
>> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01 describes
>usage of PCP in Mobile Deployments.
>>
>> --Tiru.
>>
>
>Yep, that's  an I-D.
>
>And, it says things like "Without
>   the particular considerations , PCP may not provide desirable
>   outcomes.  Some default behaviours may even cause negative impacts or
>   system failures in a mobile environment.  "
>
>Got an operators with a timeline on supporting it?  How about a mobile OS
>that plans to support PCP, because you will need both.  And, that would be
>interesting information.
>
>I asked around in Vancouver about PCP, and all the operators i spoke to sa=
id
>no-plans, no motivation.  But, that was a small sample.
>
>In any event, my only point is that RTCWEB should NOT be counting on PCP. =
 If
>PCP is a tool you have, use it.  My point: it is not a tool you have.
>
>CB
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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From: Cameron Byrne <cb.list6@gmail.com>
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Subject: Re: [rtcweb] NAT behavior heuristics
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--bcaec554d23e0dbcc604c6ae6751
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On Aug 7, 2012 7:13 AM, <Markus.Isomaki@nokia.com> wrote:
>
> Hi Cameron,
>
> Cameron Byrne wrote:
> >
> >I asked around in Vancouver about PCP, and all the operators i spoke to
said
> >no-plans, no motivation.  But, that was a small sample.
>
> This is probably a fair assessment about the concrete plans. Most mobile
operators don't even follow IETF much, so I doubt that they have even heard
about PCP so far. There are however a few operators, where at least the
research and standardization people have been interested in PCP (as
witnessed by
http://datatracker.ietf.org/doc/draft-chen-pcp-mobile-deployment/). I think
some operators are also proposing PCP to be included in the 3GPP specs
(release 12). It is hard to say how much that means in the real world, but
at least I'd expect we could find a few operators willing to do
real-network trials withing the next 12 months.
>
> Personally I see PCP as a very useful tool in the cellular access
environment, for probing the NAT/FW timers and in the best case even
setting them. It will be a major chicken-and-egg situation between apps,
mobile OS's and the networks, to get anything into use. I think we should
still do our best to overcome it. I agree that RTCWeb or anyone else should
not rely on PCP (or any similar mechanim), but have the ability to make use
of it if it happens to be available.
>
> I'm unconvinced IPv6 will help at all in this case. How do you know how
long the FW on the path will keep its state up without keepalives? It will
probably be the same as for NATs: In some networks timeout is 2 minutes,
while in others 60 minutes. Having something like PCP to check out that it
indeed is 60 min. would be extremely helpful. While most app developers
might not be willing/able to optimize for battery/mobile, there are already
some that are doing a good job. So there is hope that the new mechanisms
would get some traction.
>

I know of one large USA mobile operator that is not SPI firewalling or
tracking state for ipv6 mobiels, so this is not a problem for them.

If operators choose to statefully inspect user traffic, those cost and
limitations are assumed as part of the cost and service definition from the
mobile operator.

IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks.

CB

> Markus
>
>
> >-----Original Message-----
> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> >Of ext Cameron Byrne
> >Sent: 06 August, 2012 01:22
> >To: Tirumaleswar Reddy (tireddy)
> >Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
> >Subject: Re: [rtcweb] NAT behavior heuristics
> >
> >On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
> ><tireddy@cisco.com> wrote:
> >>> Fyi. I have not seen any traction for pcp anywhere in the mobile
space.
> >>
> >> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01describes
> >usage of PCP in Mobile Deployments.
> >>
> >> --Tiru.
> >>
> >
> >Yep, that's  an I-D.
> >
> >And, it says things like "Without
> >   the particular considerations , PCP may not provide desirable
> >   outcomes.  Some default behaviours may even cause negative impacts or
> >   system failures in a mobile environment.  "
> >
> >Got an operators with a timeline on supporting it?  How about a mobile OS
> >that plans to support PCP, because you will need both.  And, that would
be
> >interesting information.
> >
> >I asked around in Vancouver about PCP, and all the operators i spoke to
said
> >no-plans, no motivation.  But, that was a small sample.
> >
> >In any event, my only point is that RTCWEB should NOT be counting on
PCP.  If
> >PCP is a tool you have, use it.  My point: it is not a tool you have.
> >
> >CB
> >_______________________________________________
> >rtcweb mailing list
> >rtcweb@ietf.org
> >https://www.ietf.org/mailman/listinfo/rtcweb

--bcaec554d23e0dbcc604c6ae6751
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<p><br>
On Aug 7, 2012 7:13 AM, &lt;<a href=3D"mailto:Markus.Isomaki@nokia.com">Mar=
kus.Isomaki@nokia.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi Cameron,<br>
&gt;<br>
&gt; Cameron Byrne wrote:<br>
&gt; &gt;<br>
&gt; &gt;I asked around in Vancouver about PCP, and all the operators i spo=
ke to said<br>
&gt; &gt;no-plans, no motivation. =A0But, that was a small sample.<br>
&gt;<br>
&gt; This is probably a fair assessment about the concrete plans. Most mobi=
le operators don&#39;t even follow IETF much, so I doubt that they have eve=
n heard about PCP so far. There are however a few operators, where at least=
 the research and standardization people have been interested in PCP (as wi=
tnessed by <a href=3D"http://datatracker.ietf.org/doc/draft-chen-pcp-mobile=
-deployment/">http://datatracker.ietf.org/doc/draft-chen-pcp-mobile-deploym=
ent/</a>). I think some operators are also proposing PCP to be included in =
the 3GPP specs (release 12). It is hard to say how much that means in the r=
eal world, but at least I&#39;d expect we could find a few operators willin=
g to do real-network trials withing the next 12 months.<br>

&gt;<br>
&gt; Personally I see PCP as a very useful tool in the cellular access envi=
ronment, for probing the NAT/FW timers and in the best case even setting th=
em. It will be a major chicken-and-egg situation between apps, mobile OS&#3=
9;s and the networks, to get anything into use. I think we should still do =
our best to overcome it. I agree that RTCWeb or anyone else should not rely=
 on PCP (or any similar mechanim), but have the ability to make use of it i=
f it happens to be available.<br>

&gt;<br>
&gt; I&#39;m unconvinced IPv6 will help at all in this case. How do you kno=
w how long the FW on the path will keep its state up without keepalives? It=
 will probably be the same as for NATs: In some networks timeout is 2 minut=
es, while in others 60 minutes. Having something like PCP to check out that=
 it indeed is 60 min. would be extremely helpful. While most app developers=
 might not be willing/able to optimize for battery/mobile, there are alread=
y some that are doing a good job. So there is hope that the new mechanisms =
would get some traction.<br>

&gt;</p>
<p>I know of one large USA mobile operator that is not SPI firewalling or t=
racking state for ipv6 mobiels, so this is not a problem for them.</p>
<p>If operators choose to statefully inspect user traffic, those cost and l=
imitations are assumed as part of the cost and service definition from the =
mobile operator.</p>
<p>IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks=
. </p>
<p>CB</p>
<p>&gt; Markus<br>
&gt;<br>
&gt;<br>
&gt; &gt;-----Original Message-----<br>
&gt; &gt;From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ie=
tf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounce=
s@ietf.org</a>] On Behalf<br>
&gt; &gt;Of ext Cameron Byrne<br>
&gt; &gt;Sent: 06 August, 2012 01:22<br>
&gt; &gt;To: Tirumaleswar Reddy (tireddy)<br>
&gt; &gt;Cc: Randell Jesup; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.=
org</a>; <a href=3D"mailto:phdgang@gmail.com">phdgang@gmail.com</a><br>
&gt; &gt;Subject: Re: [rtcweb] NAT behavior heuristics<br>
&gt; &gt;<br>
&gt; &gt;On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)<br>
&gt; &gt;&lt;<a href=3D"mailto:tireddy@cisco.com">tireddy@cisco.com</a>&gt;=
 wrote:<br>
&gt; &gt;&gt;&gt; Fyi. I have not seen any traction for pcp anywhere in the=
 mobile space.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; <a href=3D"http://tools.ietf.org/html/draft-chen-pcp-mobile-d=
eployment-01">http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-0=
1</a> describes<br>
&gt; &gt;usage of PCP in Mobile Deployments.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; --Tiru.<br>
&gt; &gt;&gt;<br>
&gt; &gt;<br>
&gt; &gt;Yep, that&#39;s =A0an I-D.<br>
&gt; &gt;<br>
&gt; &gt;And, it says things like &quot;Without<br>
&gt; &gt; =A0 the particular considerations , PCP may not provide desirable=
<br>
&gt; &gt; =A0 outcomes. =A0Some default behaviours may even cause negative =
impacts or<br>
&gt; &gt; =A0 system failures in a mobile environment. =A0&quot;<br>
&gt; &gt;<br>
&gt; &gt;Got an operators with a timeline on supporting it? =A0How about a =
mobile OS<br>
&gt; &gt;that plans to support PCP, because you will need both. =A0And, tha=
t would be<br>
&gt; &gt;interesting information.<br>
&gt; &gt;<br>
&gt; &gt;I asked around in Vancouver about PCP, and all the operators i spo=
ke to said<br>
&gt; &gt;no-plans, no motivation. =A0But, that was a small sample.<br>
&gt; &gt;<br>
&gt; &gt;In any event, my only point is that RTCWEB should NOT be counting =
on PCP. =A0If<br>
&gt; &gt;PCP is a tool you have, use it. =A0My point: it is not a tool you =
have.<br>
&gt; &gt;<br>
&gt; &gt;CB<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://w=
ww.ietf.org/mailman/listinfo/rtcweb</a><br>
</p>

--bcaec554d23e0dbcc604c6ae6751--

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Cameron Byrne <cb.list6@gmail.com>, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Subject: Re: [rtcweb] NAT behavior heuristics
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> I know of one large USA mobile operator that is not SPI firewalling or tr=
acking state for ipv6 mobiels, so this is not a problem for them.
> If operators choose to statefully inspect user traffic, those cost and li=
mitations are assumed as part of the cost and service definition from the m=
obile=20
> operator.
> IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks.

http://www.bluecoat.com/sites/default/files/documents/files/Content_Filteri=
ng_for_Mobile_Operators.pdf
offers URL filtering based on user profile, claims to block Skype if sent o=
ver port 80/443  (http://www.bluecoat.com/sites/default/files/documents/fil=
es/How_Mobile_Operators_Can_Control_Skype.1.pdf), block viruses/worm. This =
product is targeted for Mobile Operators.=20

--Tiru.


From: Cameron Byrne [mailto:cb.list6@gmail.com]=20
Sent: Tuesday, August 07, 2012 8:41 PM
To: Markus.Isomaki@nokia.com
Cc: rtcweb@ietf.org; Tirumaleswar Reddy (tireddy); phdgang@gmail.com; rande=
ll-ietf@jesup.org
Subject: RE: [rtcweb] NAT behavior heuristics


On Aug 7, 2012 7:13 AM, <Markus.Isomaki@nokia.com> wrote:
>
> Hi Cameron,
>
> Cameron Byrne wrote:
> >
> >I asked around in Vancouver about PCP, and all the operators i spoke to =
said
> >no-plans, no motivation. =A0But, that was a small sample.
>
> This is probably a fair assessment about the concrete plans. Most mobile =
operators don't even follow IETF much, so I doubt that they have even heard=
 about PCP so far. There are however a few operators, where at least the re=
search and standardization people have been interested in PCP (as witnessed=
 by http://datatracker.ietf.org/doc/draft-chen-pcp-mobile-deployment/). I t=
hink some operators are also proposing PCP to be included in the 3GPP specs=
 (release 12). It is hard to say how much that means in the real world, but=
 at least I'd expect we could find a few operators willing to do real-netwo=
rk trials withing the next 12 months.
>
> Personally I see PCP as a very useful tool in the cellular access environ=
ment, for probing the NAT/FW timers and in the best case even setting them.=
 It will be a major chicken-and-egg situation between apps, mobile OS's and=
 the networks, to get anything into use. I think we should still do our bes=
t to overcome it. I agree that RTCWeb or anyone else should not rely on PCP=
 (or any similar mechanim), but have the ability to make use of it if it ha=
ppens to be available.
>
> I'm unconvinced IPv6 will help at all in this case. How do you know how l=
ong the FW on the path will keep its state up without keepalives? It will p=
robably be the same as for NATs: In some networks timeout is 2 minutes, whi=
le in others 60 minutes. Having something like PCP to check out that it ind=
eed is 60 min. would be extremely helpful. While most app developers might =
not be willing/able to optimize for battery/mobile, there are already some =
that are doing a good job. So there is hope that the new mechanisms would g=
et some traction.
>
I know of one large USA mobile operator that is not SPI firewalling or trac=
king state for ipv6 mobiels, so this is not a problem for them.
If operators choose to statefully inspect user traffic, those cost and limi=
tations are assumed as part of the cost and service definition from the mob=
ile operator.
IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks.=20
CB
> Markus
>
>
> >-----Original Message-----
> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> >Of ext Cameron Byrne
> >Sent: 06 August, 2012 01:22
> >To: Tirumaleswar Reddy (tireddy)
> >Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
> >Subject: Re: [rtcweb] NAT behavior heuristics
> >
> >On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
> ><tireddy@cisco.com> wrote:
> >>> Fyi. I have not seen any traction for pcp anywhere in the mobile spac=
e.
> >>
> >> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01 describ=
es
> >usage of PCP in Mobile Deployments.
> >>
> >> --Tiru.
> >>
> >
> >Yep, that's =A0an I-D.
> >
> >And, it says things like "Without
> > =A0 the particular considerations , PCP may not provide desirable
> > =A0 outcomes. =A0Some default behaviours may even cause negative impact=
s or
> > =A0 system failures in a mobile environment. =A0"
> >
> >Got an operators with a timeline on supporting it? =A0How about a mobile=
 OS
> >that plans to support PCP, because you will need both. =A0And, that woul=
d be
> >interesting information.
> >
> >I asked around in Vancouver about PCP, and all the operators i spoke to =
said
> >no-plans, no motivation. =A0But, that was a small sample.
> >
> >In any event, my only point is that RTCWEB should NOT be counting on PCP=
. =A0If
> >PCP is a tool you have, use it. =A0My point: it is not a tool you have.
> >
> >CB
> >_______________________________________________
> >rtcweb mailing list
> >rtcweb@ietf.org
> >https://www.ietf.org/mailman/listinfo/rtcweb

From lorenzo@meetecho.com  Tue Aug  7 09:08:06 2012
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Date: Tue, 7 Aug 2012 18:01:56 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] HTTP Fallback draft
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Dear all,

during the meeting in Vancouver, there was some discussion about the
need we have for UDP-blocking firewall traversal in RTCWEB. HTTP
fallback was mentioned, and I remember it being a matter of discussion
at the latest interim meeting as well (at least looking at the minutes),
even though it hasn't apparently been addressed on the ML since.
Something in line with this was also discussed in Prague at the BOF,
when it was clearly quite too early to address.

Someone suggested that, rather than trying to address the issue right
away, it could be better to first try and summarize as a BCP draft what
people already do in that respect: it's quite obvious, in fact, that
this is already being done by many companies in their deployments, even
though probably in a non-standard way.

Since the subject interests me (during the abovementioned IETF in Prague
we brought a simple demo, which we used to informally showcase to a few
interested folks the basic prototype refrlecting our research in that
field) I chose to write a simple individual draft, instead, that rather
than doing the summary, basically documents the first steps we did with
our prototype. You can read the draft here:

   http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt

Please beware that it is not at all meant to be a ready-to-be-used
solution to the problem, quite the opposite. I wrote the draft in a few
hours and it simply addresses what we started to do to try and address
the problem, and, as it will be obvious when reading the draft, it is
all but complete with respect to what such a solution is supposed to
make available: nevertheless, it tries to explain the basics of what may
be needed, even if much more is still missing. Its purpose is rather
trying to "throw a stone in the lake", as we like to say in Italy, in
order to draw attention on the problem and gather potential interest by
other people to work on it, whether or not such work would be based on
this draft. I placed a few Editors Notes across the text on some of the
most controversial issues, like how this should mix with ICE and
negotiation, how easy it should be to detect it and so on. Other even
more controversial ones, like how to address the impact of congestion
control on real-time streams, is not in the document as of yet, but it's
clear this will have to be taken care of eventually. Feedback and
suggestions are more than welcome!

That said, I guess there's a different question I should be asking to
the chairs: since there seems to be no related item in the milestones,
is such a work actually in line with what is the expected outcome of the
WG? Considering the draft basically addresses a new transport for RTP
and something that probably needs to be negotiated as well, I guess this
could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
Nevertheless, my feeling is it belongs more here than somewhere else,
especially considering we're specifying a solution that will be deployed
in browsers and, as such, people will expect it to work wherever other
web applications do.

Thanks,
Lorenzo

-- 
Lorenzo Miniero, COB

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

From cb.list6@gmail.com  Tue Aug  7 09:08:42 2012
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On Tue, Aug 7, 2012 at 8:37 AM, Tirumaleswar Reddy (tireddy)
<tireddy@cisco.com> wrote:
>> I know of one large USA mobile operator that is not SPI firewalling or t=
racking state for ipv6 mobiels, so this is not a problem for them.
>> If operators choose to statefully inspect user traffic, those cost and l=
imitations are assumed as part of the cost and service definition from the =
mobile
>> operator.
>> IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks=
.
>
> http://www.bluecoat.com/sites/default/files/documents/files/Content_Filte=
ring_for_Mobile_Operators.pdf
> offers URL filtering based on user profile, claims to block Skype if sent=
 over port 80/443  (http://www.bluecoat.com/sites/default/files/documents/f=
iles/How_Mobile_Operators_Can_Control_Skype.1.pdf), block viruses/worm. Thi=
s product is targeted for Mobile Operators.
>

Since you just sent URLs without an explanation, i am not sure the
intent of your email.

If you are suggesting that mobile operators should buy this product to
block Skype, you are probably on the wrong mailing list.

If you are suggesting mobile operators should buy this product to
block Skype and then use PCP to allow Skype, ....  Not sure that will
get much traction either.

CB

> --Tiru.
>
>
> From: Cameron Byrne [mailto:cb.list6@gmail.com]
> Sent: Tuesday, August 07, 2012 8:41 PM
> To: Markus.Isomaki@nokia.com
> Cc: rtcweb@ietf.org; Tirumaleswar Reddy (tireddy); phdgang@gmail.com; ran=
dell-ietf@jesup.org
> Subject: RE: [rtcweb] NAT behavior heuristics
>
>
> On Aug 7, 2012 7:13 AM, <Markus.Isomaki@nokia.com> wrote:
>>
>> Hi Cameron,
>>
>> Cameron Byrne wrote:
>> >
>> >I asked around in Vancouver about PCP, and all the operators i spoke to=
 said
>> >no-plans, no motivation.  But, that was a small sample.
>>
>> This is probably a fair assessment about the concrete plans. Most mobile=
 operators don't even follow IETF much, so I doubt that they have even hear=
d about PCP so far. There are however a few operators, where at least the r=
esearch and standardization people have been interested in PCP (as witnesse=
d by http://datatracker.ietf.org/doc/draft-chen-pcp-mobile-deployment/). I =
think some operators are also proposing PCP to be included in the 3GPP spec=
s (release 12). It is hard to say how much that means in the real world, bu=
t at least I'd expect we could find a few operators willing to do real-netw=
ork trials withing the next 12 months.
>>
>> Personally I see PCP as a very useful tool in the cellular access enviro=
nment, for probing the NAT/FW timers and in the best case even setting them=
. It will be a major chicken-and-egg situation between apps, mobile OS's an=
d the networks, to get anything into use. I think we should still do our be=
st to overcome it. I agree that RTCWeb or anyone else should not rely on PC=
P (or any similar mechanim), but have the ability to make use of it if it h=
appens to be available.
>>
>> I'm unconvinced IPv6 will help at all in this case. How do you know how =
long the FW on the path will keep its state up without keepalives? It will =
probably be the same as for NATs: In some networks timeout is 2 minutes, wh=
ile in others 60 minutes. Having something like PCP to check out that it in=
deed is 60 min. would be extremely helpful. While most app developers might=
 not be willing/able to optimize for battery/mobile, there are already some=
 that are doing a good job. So there is hope that the new mechanisms would =
get some traction.
>>
> I know of one large USA mobile operator that is not SPI firewalling or tr=
acking state for ipv6 mobiels, so this is not a problem for them.
> If operators choose to statefully inspect user traffic, those cost and li=
mitations are assumed as part of the cost and service definition from the m=
obile operator.
> IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile networks.
> CB
>> Markus
>>
>>
>> >-----Original Message-----
>> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behal=
f
>> >Of ext Cameron Byrne
>> >Sent: 06 August, 2012 01:22
>> >To: Tirumaleswar Reddy (tireddy)
>> >Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
>> >Subject: Re: [rtcweb] NAT behavior heuristics
>> >
>> >On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
>> ><tireddy@cisco.com> wrote:
>> >>> Fyi. I have not seen any traction for pcp anywhere in the mobile spa=
ce.
>> >>
>> >> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01 descri=
bes
>> >usage of PCP in Mobile Deployments.
>> >>
>> >> --Tiru.
>> >>
>> >
>> >Yep, that's  an I-D.
>> >
>> >And, it says things like "Without
>> >   the particular considerations , PCP may not provide desirable
>> >   outcomes.  Some default behaviours may even cause negative impacts o=
r
>> >   system failures in a mobile environment.  "
>> >
>> >Got an operators with a timeline on supporting it?  How about a mobile =
OS
>> >that plans to support PCP, because you will need both.  And, that would=
 be
>> >interesting information.
>> >
>> >I asked around in Vancouver about PCP, and all the operators i spoke to=
 said
>> >no-plans, no motivation.  But, that was a small sample.
>> >
>> >In any event, my only point is that RTCWEB should NOT be counting on PC=
P.  If
>> >PCP is a tool you have, use it.  My point: it is not a tool you have.
>> >
>> >CB
>> >_______________________________________________
>> >rtcweb mailing list
>> >rtcweb@ietf.org
>> >https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Cameron Byrne <cb.list6@gmail.com>
Thread-Topic: [rtcweb] NAT behavior heuristics
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Subject: Re: [rtcweb] NAT behavior heuristics
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> If you are suggesting that mobile operators should buy this product to
> block Skype, you are probably on the wrong mailing list.

I am not selling this product. I am just saying that these products are off=
ered to Mobile Operators to detect Port Misuse/for other security purposes =
like detecting viruses and hence the Mobile Network will have a firewall/IP=
S that will do DPI. My argument is IPv6 Firewall/IPS in various modes like =
content filtering, Application Visibility and Control when deployed in Mobi=
le Network will have sessions just like NAT but a different purpose and wil=
l have timeout mechanism. PCP can also be used to avoid keep-alive for Fire=
wall sessions. It's up to the Mobile Operator policy to permit the flow or =
not, honor the PCP request or not to permit certain flow.

--Tiru.

> -----Original Message-----
> From: Cameron Byrne [mailto:cb.list6@gmail.com]
> Sent: Tuesday, August 07, 2012 9:39 PM
> To: Tirumaleswar Reddy (tireddy)
> Cc: Markus.Isomaki@nokia.com; rtcweb@ietf.org; phdgang@gmail.com;
> randell-ietf@jesup.org
> Subject: Re: [rtcweb] NAT behavior heuristics
>=20
> On Tue, Aug 7, 2012 at 8:37 AM, Tirumaleswar Reddy (tireddy)
> <tireddy@cisco.com> wrote:
> >> I know of one large USA mobile operator that is not SPI firewalling
> or tracking state for ipv6 mobiels, so this is not a problem for them.
> >> If operators choose to statefully inspect user traffic, those cost
> and limitations are assumed as part of the cost and service definition
> from the mobile
> >> operator.
> >> IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile
> networks.
> >
> >
> http://www.bluecoat.com/sites/default/files/documents/files/Content_Fil
> tering_for_Mobile_Operators.pdf
> > offers URL filtering based on user profile, claims to block Skype if
> sent over port 80/443
> (http://www.bluecoat.com/sites/default/files/documents/files/How_Mobile
> _Operators_Can_Control_Skype.1.pdf), block viruses/worm. This product
> is targeted for Mobile Operators.
> >
>=20
> Since you just sent URLs without an explanation, i am not sure the
> intent of your email.
>=20
> If you are suggesting that mobile operators should buy this product to
> block Skype, you are probably on the wrong mailing list.
>=20
> If you are suggesting mobile operators should buy this product to
> block Skype and then use PCP to allow Skype, ....  Not sure that will
> get much traction either.
>=20
> CB
>=20
> > --Tiru.
> >
> >
> > From: Cameron Byrne [mailto:cb.list6@gmail.com]
> > Sent: Tuesday, August 07, 2012 8:41 PM
> > To: Markus.Isomaki@nokia.com
> > Cc: rtcweb@ietf.org; Tirumaleswar Reddy (tireddy); phdgang@gmail.com;
> randell-ietf@jesup.org
> > Subject: RE: [rtcweb] NAT behavior heuristics
> >
> >
> > On Aug 7, 2012 7:13 AM, <Markus.Isomaki@nokia.com> wrote:
> >>
> >> Hi Cameron,
> >>
> >> Cameron Byrne wrote:
> >> >
> >> >I asked around in Vancouver about PCP, and all the operators i
> spoke to said
> >> >no-plans, no motivation.  But, that was a small sample.
> >>
> >> This is probably a fair assessment about the concrete plans. Most
> mobile operators don't even follow IETF much, so I doubt that they have
> even heard about PCP so far. There are however a few operators, where
> at least the research and standardization people have been interested
> in PCP (as witnessed by http://datatracker.ietf.org/doc/draft-chen-pcp-
> mobile-deployment/). I think some operators are also proposing PCP to
> be included in the 3GPP specs (release 12). It is hard to say how much
> that means in the real world, but at least I'd expect we could find a
> few operators willing to do real-network trials withing the next 12
> months.
> >>
> >> Personally I see PCP as a very useful tool in the cellular access
> environment, for probing the NAT/FW timers and in the best case even
> setting them. It will be a major chicken-and-egg situation between
> apps, mobile OS's and the networks, to get anything into use. I think
> we should still do our best to overcome it. I agree that RTCWeb or
> anyone else should not rely on PCP (or any similar mechanim), but have
> the ability to make use of it if it happens to be available.
> >>
> >> I'm unconvinced IPv6 will help at all in this case. How do you know
> how long the FW on the path will keep its state up without keepalives?
> It will probably be the same as for NATs: In some networks timeout is 2
> minutes, while in others 60 minutes. Having something like PCP to check
> out that it indeed is 60 min. would be extremely helpful. While most
> app developers might not be willing/able to optimize for
> battery/mobile, there are already some that are doing a good job. So
> there is hope that the new mechanisms would get some traction.
> >>
> > I know of one large USA mobile operator that is not SPI firewalling
> or tracking state for ipv6 mobiels, so this is not a problem for them.
> > If operators choose to statefully inspect user traffic, those cost
> and limitations are assumed as part of the cost and service definition
> from the mobile operator.
> > IMHO, SPI FW in ipv6 is not prudent or cost justified in mobile
> networks.
> > CB
> >> Markus
> >>
> >>
> >> >-----Original Message-----
> >> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf
> >> >Of ext Cameron Byrne
> >> >Sent: 06 August, 2012 01:22
> >> >To: Tirumaleswar Reddy (tireddy)
> >> >Cc: Randell Jesup; rtcweb@ietf.org; phdgang@gmail.com
> >> >Subject: Re: [rtcweb] NAT behavior heuristics
> >> >
> >> >On Sun, Aug 5, 2012 at 3:05 PM, Tirumaleswar Reddy (tireddy)
> >> ><tireddy@cisco.com> wrote:
> >> >>> Fyi. I have not seen any traction for pcp anywhere in the mobile
> space.
> >> >>
> >> >> http://tools.ietf.org/html/draft-chen-pcp-mobile-deployment-01
> describes
> >> >usage of PCP in Mobile Deployments.
> >> >>
> >> >> --Tiru.
> >> >>
> >> >
> >> >Yep, that's  an I-D.
> >> >
> >> >And, it says things like "Without
> >> >   the particular considerations , PCP may not provide desirable
> >> >   outcomes.  Some default behaviours may even cause negative
> impacts or
> >> >   system failures in a mobile environment.  "
> >> >
> >> >Got an operators with a timeline on supporting it?  How about a
> mobile OS
> >> >that plans to support PCP, because you will need both.  And, that
> would be
> >> >interesting information.
> >> >
> >> >I asked around in Vancouver about PCP, and all the operators i
> spoke to said
> >> >no-plans, no motivation.  But, that was a small sample.
> >> >
> >> >In any event, my only point is that RTCWEB should NOT be counting
> on PCP.  If
> >> >PCP is a tool you have, use it.  My point: it is not a tool you
> have.
> >> >
> >> >CB
> >> >_______________________________________________
> >> >rtcweb mailing list
> >> >rtcweb@ietf.org
> >> >https://www.ietf.org/mailman/listinfo/rtcweb

From hannes.tschofenig@gmx.net  Tue Aug  7 09:38:57 2012
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To: Lorenzo Miniero <lorenzo@meetecho.com>
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Subject: Re: [rtcweb] HTTP Fallback draft
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Hi Lorenzo,=20

working on http://tools.ietf.org/html/draft-tschofenig-hourglass-00 (and =
Marc on draft-blanchet-iab-internetoverport443) I am wondering about the =
following aspect:=20

1) Do you have experience with network elements filtering RTP traffic? =
If so, in what networks did that happen?

2) In case you answered question #1 with 'yes' I would also like to =
understand why you believe that HTTP traffic will be allowed to =
traverse? Wouldn't you have to switch to HTTPS instead?=20

Just to be clear, I don't have objections regarding your draft. I just =
would like to understand what motives people to run everything over HTTP =
(besides using HTTP features, which you don't really do).=20

Ciao
Hannes

On Aug 7, 2012, at 7:01 PM, Lorenzo Miniero wrote:

> Dear all,
>=20
> during the meeting in Vancouver, there was some discussion about the
> need we have for UDP-blocking firewall traversal in RTCWEB. HTTP
> fallback was mentioned, and I remember it being a matter of discussion
> at the latest interim meeting as well (at least looking at the =
minutes),
> even though it hasn't apparently been addressed on the ML since.
> Something in line with this was also discussed in Prague at the BOF,
> when it was clearly quite too early to address.
>=20
> Someone suggested that, rather than trying to address the issue right
> away, it could be better to first try and summarize as a BCP draft =
what
> people already do in that respect: it's quite obvious, in fact, that
> this is already being done by many companies in their deployments, =
even
> though probably in a non-standard way.
>=20
> Since the subject interests me (during the abovementioned IETF in =
Prague
> we brought a simple demo, which we used to informally showcase to a =
few
> interested folks the basic prototype refrlecting our research in that
> field) I chose to write a simple individual draft, instead, that =
rather
> than doing the summary, basically documents the first steps we did =
with
> our prototype. You can read the draft here:
>=20
>   http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt
>=20
> Please beware that it is not at all meant to be a ready-to-be-used
> solution to the problem, quite the opposite. I wrote the draft in a =
few
> hours and it simply addresses what we started to do to try and address
> the problem, and, as it will be obvious when reading the draft, it is
> all but complete with respect to what such a solution is supposed to
> make available: nevertheless, it tries to explain the basics of what =
may
> be needed, even if much more is still missing. Its purpose is rather
> trying to "throw a stone in the lake", as we like to say in Italy, in
> order to draw attention on the problem and gather potential interest =
by
> other people to work on it, whether or not such work would be based on
> this draft. I placed a few Editors Notes across the text on some of =
the
> most controversial issues, like how this should mix with ICE and
> negotiation, how easy it should be to detect it and so on. Other even
> more controversial ones, like how to address the impact of congestion
> control on real-time streams, is not in the document as of yet, but =
it's
> clear this will have to be taken care of eventually. Feedback and
> suggestions are more than welcome!
>=20
> That said, I guess there's a different question I should be asking to
> the chairs: since there seems to be no related item in the milestones,
> is such a work actually in line with what is the expected outcome of =
the
> WG? Considering the draft basically addresses a new transport for RTP
> and something that probably needs to be negotiated as well, I guess =
this
> could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> Nevertheless, my feeling is it belongs more here than somewhere else,
> especially considering we're specifying a solution that will be =
deployed
> in browsers and, as such, people will expect it to work wherever other
> web applications do.
>=20
> Thanks,
> Lorenzo
>=20
> --=20
> Lorenzo Miniero, COB
>=20
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From ekr@rtfm.com  Tue Aug  7 10:15:14 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 7 Aug 2012 10:14:31 -0700
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Subject: [rtcweb] Initial notes on MS proposal
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I've gone over MS's proposal and written up some initial notes:

http://www.educatedguesswork.org/2012/08/initial_notes_on_microsofts_cu.html

-Ekr

From ekr@rtfm.com  Tue Aug  7 10:18:36 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 7 Aug 2012 10:17:54 -0700
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FWIW, I'm finding that CU-RTC-Web is a bit of a mouthful. I hereby
propose the pronunciation "Curtsy-Web"

-Ekr


On Tue, Aug 7, 2012 at 10:14 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> I've gone over MS's proposal and written up some initial notes:
>
> http://www.educatedguesswork.org/2012/08/initial_notes_on_microsofts_cu.html
>
> -Ekr

From lorenzo@meetecho.com  Tue Aug  7 10:18:37 2012
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Date: Tue, 7 Aug 2012 19:12:26 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
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Hi Hannes,

answering inline,


On Tue, 7 Aug 2012 19:38:52 +0300
Hannes Tschofenig <hannes.tschofenig@gmx.net> wrote:

> Hi Lorenzo, 
> 
> working on http://tools.ietf.org/html/draft-tschofenig-hourglass-00 (and Marc on draft-blanchet-iab-internetoverport443) I am wondering about the following aspect: 
> 
> 1) Do you have experience with network elements filtering RTP traffic? If so, in what networks did that happen?
> 


We indeed met such network elements in our experience, even though most of the times it was just blind UDP filtering rather than RTP filtering per se. It sometimes was just blind port filtering but the effect was the same. This mostly happened within enterprises networks where we tried, no matter how small or large.


> 2) In case you answered question #1 with 'yes' I would also like to understand why you believe that HTTP traffic will be allowed to traverse? Wouldn't you have to switch to HTTPS instead? 
> 


You're right. A couple of years ago we wrote a paper addressing the potential approaches for tunneling attempts (if you're interested, I can send you the link offline). In this paper we basically described, as a diagram, which incremental steps could be carried out in order to attempt a successful tunneling: 1) the first attempt is to just try port 443, without encapulating anything (e.g., ssh using 443 instead of 22); 2) in case that doesn't work, the second attempt is to use HTTP CONNECT and then fall back to 1. for the connection that is established; 3) the third attempt (e.g., 443 is not available or the proxy acts as a MITM) is to actually encapsulate in HTTP messages, whether you do HTTP or HTTPS. In every case, the peer (either endpoint or server) must be configured accordingly of course.

What I describe in the draft is step 3, even though I guess some words to suggest steps 1 and 2 (where you'd still need to encapsulate RTP packets on top of a TCP-based protocol anyway) could be considered. As long as it looks like valid HTTP and it behaves accordingly, I think there's no reason why traversing should be impeded: there are cases where of course we might WANT the traversal to be controlled (as I said in the draft, it doesn't have to be sneaky, and could be made so that network administrators can control it) but that's a different issue. Probably far from being the optimal solution, but for extreme scenarios it may get things working whereas everything else might fail.


> Just to be clear, I don't have objections regarding your draft. I just would like to understand what motives people to run everything over HTTP (besides using HTTP features, which you don't really do). 
> 


I agree with you and I'm not really dying to do RTP over HTTP either, but if some scenarios make it impossible for use cases to work (and some firewall/NAT deployers are to blame here, probably) then a fallback mechanism is something that can be nice to have, especially if we're interested in something that "just works".

Thanks,
Lorenzo


> Ciao
> Hannes
> 
> On Aug 7, 2012, at 7:01 PM, Lorenzo Miniero wrote:
> 
> > Dear all,
> > 
> > during the meeting in Vancouver, there was some discussion about the
> > need we have for UDP-blocking firewall traversal in RTCWEB. HTTP
> > fallback was mentioned, and I remember it being a matter of discussion
> > at the latest interim meeting as well (at least looking at the minutes),
> > even though it hasn't apparently been addressed on the ML since.
> > Something in line with this was also discussed in Prague at the BOF,
> > when it was clearly quite too early to address.
> > 
> > Someone suggested that, rather than trying to address the issue right
> > away, it could be better to first try and summarize as a BCP draft what
> > people already do in that respect: it's quite obvious, in fact, that
> > this is already being done by many companies in their deployments, even
> > though probably in a non-standard way.
> > 
> > Since the subject interests me (during the abovementioned IETF in Prague
> > we brought a simple demo, which we used to informally showcase to a few
> > interested folks the basic prototype refrlecting our research in that
> > field) I chose to write a simple individual draft, instead, that rather
> > than doing the summary, basically documents the first steps we did with
> > our prototype. You can read the draft here:
> > 
> >   http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt
> > 
> > Please beware that it is not at all meant to be a ready-to-be-used
> > solution to the problem, quite the opposite. I wrote the draft in a few
> > hours and it simply addresses what we started to do to try and address
> > the problem, and, as it will be obvious when reading the draft, it is
> > all but complete with respect to what such a solution is supposed to
> > make available: nevertheless, it tries to explain the basics of what may
> > be needed, even if much more is still missing. Its purpose is rather
> > trying to "throw a stone in the lake", as we like to say in Italy, in
> > order to draw attention on the problem and gather potential interest by
> > other people to work on it, whether or not such work would be based on
> > this draft. I placed a few Editors Notes across the text on some of the
> > most controversial issues, like how this should mix with ICE and
> > negotiation, how easy it should be to detect it and so on. Other even
> > more controversial ones, like how to address the impact of congestion
> > control on real-time streams, is not in the document as of yet, but it's
> > clear this will have to be taken care of eventually. Feedback and
> > suggestions are more than welcome!
> > 
> > That said, I guess there's a different question I should be asking to
> > the chairs: since there seems to be no related item in the milestones,
> > is such a work actually in line with what is the expected outcome of the
> > WG? Considering the draft basically addresses a new transport for RTP
> > and something that probably needs to be negotiated as well, I guess this
> > could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> > Nevertheless, my feeling is it belongs more here than somewhere else,
> > especially considering we're specifying a solution that will be deployed
> > in browsers and, as such, people will expect it to work wherever other
> > web applications do.
> > 
> > Thanks,
> > Lorenzo
> > 
> > -- 
> > Lorenzo Miniero, COB
> > 
> > Meetecho s.r.l.
> > Web Conferencing and Collaboration Tools
> > http://www.meetecho.com
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 
Lorenzo Miniero, COB

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

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On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
> That said, I guess there's a different question I should be asking to
> the chairs: since there seems to be no related item in the milestones,
> is such a work actually in line with what is the expected outcome of the
> WG? Considering the draft basically addresses a new transport for RTP
> and something that probably needs to be negotiated as well, I guess this
> could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> Nevertheless, my feeling is it belongs more here than somewhere else,
> especially considering we're specifying a solution that will be deployed
> in browsers and, as such, people will expect it to work wherever other
> web applications do.

My take on this is that the actual work on developing the alternate
transport for RTP would have to occur elsewhere and, frankly, I think
it is a large enough task that it would likely require its own working
group (much as the RTP congestion control topic ended up as a BoF and
hopefully will become its own working group).  That doesn't mean that
the work couldn't be informed by the RTCWEB use cases, but I think it
would have to be done elsewhere.

I'd personally suggest starting with a discussion with the ADs on
whether a BoF on this topic would be something they might consider.
(Note, however, that I have not talked to Cullen about this and Magnus
is on vacation, so this is not a "Chairs' response"; just my own
thoughts).

regards,

Ted Hardie

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Subject: [rtcweb] Why http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt ?
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Hi, I'm reading
http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt:

   This draft describes a way to implement an HTTP Fallback for RTP
   media streams, that is, a way to effectively encapsulate RTP packets
   in HTTP messages in order to traverse proxies and firewalls.

Well, why does WebRTC need it at all? don't we have TURN over TCP/TLS?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

On 08/07/2012 11:04 AM, IÃ±aki Baz Castillo wrote:
> Hi, I'm reading 
> http://www.ietf.org/id/draft-miniero-rtcweb-http-fallback-00.txt:
> 
> This draft describes a way to implement an HTTP Fallback for RTP media
> streams, that is, a way to effectively encapsulate RTP packets in HTTP
> messages in order to traverse proxies and firewalls.
> 
> Well, why does WebRTC need it at all? don't we have TURN over TCP/TLS?
> 

Or TURN over Websocket.

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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2012/8/7 Marc Petit-Huguenin <petithug@acm.org>:
> Or TURN over Websocket.

Why do we need that? By using TURN over TLS we have all we need to
avoid stupid hotel's firewalls (that provide "Internet free access"
but just for HTTP port 80 and HTTPS port 443). It just about setting a
TURN server listening TLS on port 443, am I right?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Tue, Aug 7, 2012 at 2:18 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/8/7 Marc Petit-Huguenin <petithug@acm.org>:
> > Or TURN over Websocket.
>
> Why do we need that? By using TURN over TLS we have all we need to
> avoid stupid hotel's firewalls (that provide "Internet free access"
> but just for HTTP port 80 and HTTPS port 443). It just about setting a
> TURN server listening TLS on port 443, am I right?
>
>
You still got the locations where the only way to connect to anything is
via the man-in-the-middle accept HTTP request and resend it proxies. Such
proxies will install their own certificate in the client certificate chain
and will decode every request. TURN over websocket will work over such
connection, but regular TURN will not. I am not sure how much effort we
want to spend supporting this, since we are talking about prisons,
military, and other similar nice places which will generally would try to
avoid supporting WebRTC due to its own security.
_____________
Roman Shpount

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<div class=3D"gmail_quote">On Tue, Aug 7, 2012 at 2:18 PM, I=F1aki Baz Cast=
illo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blan=
k">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
2012/8/7 Marc Petit-Huguenin &lt;<a href=3D"mailto:petithug@acm.org">petith=
ug@acm.org</a>&gt;:<br>
&gt; Or TURN over Websocket.<br>
<br>
Why do we need that? By using TURN over TLS we have all we need to<br>
avoid stupid hotel&#39;s firewalls (that provide &quot;Internet free access=
&quot;<br>
but just for HTTP port 80 and HTTPS port 443). It just about setting a<br>
TURN server listening TLS on port 443, am I right?<br><font color=3D"#88888=
8"><br></font></blockquote><div><br></div><div>You still got the locations =
where the only way to connect to anything is via the man-in-the-middle acce=
pt HTTP request and resend it proxies. Such proxies will install their own =
certificate in the client certificate chain and will decode every request. =
TURN over websocket will work over such connection, but regular TURN will n=
ot. I am not sure how much effort we want to spend supporting this, since w=
e are talking about prisons, military, and other similar nice places which =
will generally would try to avoid supporting WebRTC due to its own security=
.</div>
_____________<br>Roman Shpount<br><div>=A0</div></div>

--f46d042dfdcdcb385804c6b12e67--

From lorenzo@meetecho.com  Tue Aug  7 12:01:10 2012
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Hello Ted,


On Tue, 7 Aug 2012 10:50:07 -0700
Ted Hardie <ted.ietf@gmail.com> wrote:

> On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
> > That said, I guess there's a different question I should be asking to
> > the chairs: since there seems to be no related item in the milestones,
> > is such a work actually in line with what is the expected outcome of the
> > WG? Considering the draft basically addresses a new transport for RTP
> > and something that probably needs to be negotiated as well, I guess this
> > could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> > Nevertheless, my feeling is it belongs more here than somewhere else,
> > especially considering we're specifying a solution that will be deployed
> > in browsers and, as such, people will expect it to work wherever other
> > web applications do.
> 
> My take on this is that the actual work on developing the alternate
> transport for RTP would have to occur elsewhere and, frankly, I think
> it is a large enough task that it would likely require its own working
> group (much as the RTP congestion control topic ended up as a BoF and
> hopefully will become its own working group).  That doesn't mean that
> the work couldn't be informed by the RTCWEB use cases, but I think it
> would have to be done elsewhere.
> 
> I'd personally suggest starting with a discussion with the ADs on
> whether a BoF on this topic would be something they might consider.
> (Note, however, that I have not talked to Cullen about this and Magnus
> is on vacation, so this is not a "Chairs' response"; just my own
> thoughts).
> 


This makes sense. I'm a bit concerned about the additional time that may be needed by going through the process of forming a new WG (compared to just adding a milestone to an existing WG, that is), as the final result may end up being available much after the original WG completed its works, but I see your point.

I'll wait for more feedback about this and, if enough people seem interested about doing something like this, contacting the ADs and consider the next steps may be a good idea.

Lorenzo


> regards,
> 
> Ted Hardie


-- 
Lorenzo Miniero, COB

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

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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: =?UTF-8?B?ScOxYWtp?= Baz Castillo <ibc@aliax.net>
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I=C3=B1aki,

discussions would be much easier to follow if you just answered to the orig=
inal thread, rather than opening a new one... about your question, please s=
ee inline.


On Tue, 7 Aug 2012 20:18:58 +0200
I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/8/7 Marc Petit-Huguenin <petithug@acm.org>:
> > Or TURN over Websocket.
>=20
> Why do we need that? By using TURN over TLS we have all we need to
> avoid stupid hotel's firewalls (that provide "Internet free access"
> but just for HTTP port 80 and HTTPS port 443). It just about setting a
> TURN server listening TLS on port 443, am I right?
>=20


"Do we need that" is exactly what I wanted to ask by publishing the draft. =
There was already some discussions in the past, someone even suggested just=
 deploying TURN over HTTP (a bit like Marc's point on WebSockets), and TURN=
 over TLS (or just anything on 443, for what matters) is another viable sol=
ution.

The problem IMHO is that, just as Roman pointed out in his reply, that's no=
t always a way out. In my experience such a scenario is not only limited to=
 those ultra-closed environments he mentioned, but that's probably debatabl=
e (who closes his network that way, probably doesn't want anything else to =
go through anyway).

About the "just pipe everything on 443", please refer to my reply to Hannes=
 in the original thread, as I've answered more extensively there.

Lorenzo

> --=20
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Lorenzo Miniero, COB

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

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Hi Lorenzo, sorry for opening another thread about this (I didn't
check my unread emails before reading the draft). Please see comments
inline:


> Hannes Tschofenig <hannes.tschofenig@gmx.net> wrote:

>> Just to be clear, I don't have objections regarding your draft. I just w=
ould like to understand what motives people to run everything over HTTP (be=
sides using HTTP features, which you don't really do).

Honestly I don't like the "all-over-http" vision. Admins filtering
traffic other than HTTP or HTTPS will also do their best efforts for
filtering specific traffic over HTTP, and IMHO making specifications
to run protocolos over HTTP legitimizes those hateful filtering
techniques.





2012/8/7 Lorenzo Miniero <lorenzo@meetecho.com>:

> I agree with you and I'm not really dying to do RTP over HTTP either, but=
 if some scenarios make it impossible for use cases to work (and some firew=
all/NAT deployers are to blame here, probably) then a fallback mechanism is=
 something that can be nice to have, especially if we're interested in some=
thing that "just works".

That's the problem (IMHO): if we make all to work over HTTP then some
day telcos providers will just offer HTTP (as "Internet") and we'll
need to study a new OSI model in which the lowest layer is called
"HTTP".

BTW: I sitll don't understand which is the advantage of using RTP over
HTTPS in comparison to TURN over TLS:443. Does is exist?


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Hi I=C3=B1aki,


On Tue, 7 Aug 2012 21:34:09 +0200
I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> Hi Lorenzo, sorry for opening another thread about this (I didn't
> check my unread emails before reading the draft). Please see comments
> inline:
>=20


No harm done, don't worry!


>=20
> > Hannes Tschofenig <hannes.tschofenig@gmx.net> wrote:
>=20
> >> Just to be clear, I don't have objections regarding your draft. I just=
 would like to understand what motives people to run everything over HTTP (=
besides using HTTP features, which you don't really do).
>=20
> Honestly I don't like the "all-over-http" vision. Admins filtering
> traffic other than HTTP or HTTPS will also do their best efforts for
> filtering specific traffic over HTTP, and IMHO making specifications
> to run protocolos over HTTP legitimizes those hateful filtering
> techniques.
>=20


I'm not a fan either, believe me :)

That said, as I said in another post I still think that, if RTCWEB/WebRTC i=
s going to be deployed in browsers, they'll expect it to work in all the pl=
aces where all other web applications do. Many proprietary protocols that d=
o real-time communication implement tunneling somehow, or pretend to do so,=
 which means something in line with that is needed for RTCWEB as well.


>=20
>=20
>=20
>=20
> 2012/8/7 Lorenzo Miniero <lorenzo@meetecho.com>:
>=20
> > I agree with you and I'm not really dying to do RTP over HTTP either, b=
ut if some scenarios make it impossible for use cases to work (and some fir=
ewall/NAT deployers are to blame here, probably) then a fallback mechanism =
is something that can be nice to have, especially if we're interested in so=
mething that "just works".
>=20
> That's the problem (IMHO): if we make all to work over HTTP then some
> day telcos providers will just offer HTTP (as "Internet") and we'll
> need to study a new OSI model in which the lowest layer is called
> "HTTP".
>=20


Which is close enough to the hourglass draft Hannes mentioned in his post, =
I'm afraid. I can't but agree with you, but, again, if we have use cases an=
d we want to have them working in as many topologies as possible, something=
 must be done.


> BTW: I sitll don't understand which is the advantage of using RTP over
> HTTPS in comparison to TURN over TLS:443. Does is exist?
>=20


If we talk about a simple comparison, I'm sure TURN:TLS:443 performs much b=
etter than RTP over HTTPS, or at least, over the simple approach I document=
ed in the draft at least (even though I tried to keep the overhead as littl=
e as possible). Comparing them is probably not fair, though, as they addres=
s two different scenarios: that's why HTTP would be just a fallback, and no=
t something you'd use when you have alternatives.

Referring to the steps I described in my previous answer to Hannes, TURN:TL=
S:443 is probably the most useful option for scenarios 1 and 2, that is, wh=
ere you can just pipe what you want on port 443 and be sure it will be igno=
red by the proxy. If the proxy checks what's going on, though, that may not=
 be as easy. The proxy may be acting as a MITM for HTTPS, as Roman describe=
d, or just doing some traffic monitoring with anomaly detection or somethin=
g like that: faking to be HTTPS is not going to work in that case, since th=
e proxy may just see it's not HTTP at all, or may just understand it's not =
what it claims to be because it doesn't behave as it should; HTTP connectio=
ns have a usually quite simple pattern (e.g., much more traffic going in on=
e direction than the other and never at the same time), something that woul=
d be very different in a TURN connection instead.

Lorenzo

>=20
> Regards.
>=20
>=20
> --=20
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>


--=20
Lorenzo Miniero, COB

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

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To: Lorenzo Miniero <lorenzo@meetecho.com>, =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] HTTP Fallback draft
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On 08/07/2012 12:54 PM, Lorenzo Miniero wrote:
> Hi IÃ±aki,
> 
> 
> On Tue, 7 Aug 2012 21:34:09 +0200 IÃ±aki Baz Castillo <ibc@aliax.net>
> wrote:
> 
>> Hi Lorenzo, sorry for opening another thread about this (I didn't check
>> my unread emails before reading the draft). Please see comments inline:
>> 
> 
> 
> No harm done, don't worry!
> 
> 
>> 
>>> Hannes Tschofenig <hannes.tschofenig@gmx.net> wrote:
>> 
>>>> Just to be clear, I don't have objections regarding your draft. I
>>>> just would like to understand what motives people to run everything
>>>> over HTTP (besides using HTTP features, which you don't really do).
>> 
>> Honestly I don't like the "all-over-http" vision. Admins filtering 
>> traffic other than HTTP or HTTPS will also do their best efforts for 
>> filtering specific traffic over HTTP, and IMHO making specifications to
>> run protocolos over HTTP legitimizes those hateful filtering techniques.
>> 
> 
> 
> I'm not a fan either, believe me :)
> 
> That said, as I said in another post I still think that, if RTCWEB/WebRTC
> is going to be deployed in browsers, they'll expect it to work in all the
> places where all other web applications do. Many proprietary protocols that
> do real-time communication implement tunneling somehow, or pretend to do
> so, which means something in line with that is needed for RTCWEB as well.
> 
> 
>> 
>> 
>> 
>> 
>> 2012/8/7 Lorenzo Miniero <lorenzo@meetecho.com>:
>> 
>>> I agree with you and I'm not really dying to do RTP over HTTP either,
>>> but if some scenarios make it impossible for use cases to work (and
>>> some firewall/NAT deployers are to blame here, probably) then a
>>> fallback mechanism is something that can be nice to have, especially if
>>> we're interested in something that "just works".
>> 
>> That's the problem (IMHO): if we make all to work over HTTP then some day
>> telcos providers will just offer HTTP (as "Internet") and we'll need to
>> study a new OSI model in which the lowest layer is called "HTTP".
>> 
> 
> 
> Which is close enough to the hourglass draft Hannes mentioned in his post,
> I'm afraid. I can't but agree with you, but, again, if we have use cases
> and we want to have them working in as many topologies as possible,
> something must be done.
> 
> 
>> BTW: I sitll don't understand which is the advantage of using RTP over 
>> HTTPS in comparison to TURN over TLS:443. Does is exist?
>> 
> 
> 
> If we talk about a simple comparison, I'm sure TURN:TLS:443 performs much
> better than RTP over HTTPS, or at least, over the simple approach I
> documented in the draft at least (even though I tried to keep the overhead
> as little as possible). Comparing them is probably not fair, though, as
> they address two different scenarios: that's why HTTP would be just a
> fallback, and not something you'd use when you have alternatives.
> 
> Referring to the steps I described in my previous answer to Hannes,
> TURN:TLS:443 is probably the most useful option for scenarios 1 and 2, that
> is, where you can just pipe what you want on port 443 and be sure it will
> be ignored by the proxy. If the proxy checks what's going on, though, that
> may not be as easy. The proxy may be acting as a MITM for HTTPS, as Roman
> described, or just doing some traffic monitoring with anomaly detection or
> something like that: faking to be HTTPS is not going to work in that case,
> since the proxy may just see it's not HTTP at all, or may just understand
> it's not what it claims to be because it doesn't behave as it should; HTTP
> connections have a usually quite simple pattern (e.g., much more traffic
> going in one direction than the other and never at the same time),
> something that would be very different in a TURN connection instead.
> 

I think that this problem should be looked from another point of view.  One of
the basic assumption of ICE is that a TURN address is always added with the
lowest priority to the gathered candidate addresses, because it guarantees
that ICE will always works.  Now we all now that this assumption is not true
in 100% of the cases, and this is the reason why years ago I added a TURN over
HTTP(S) transport in the product I was developing at the time and worked on an
I-D (never published).  So instead of seeing TURN over Websocket as a way to
work in difficult environments, we can just see it as a way to fulfill the
promise of ICE, which is to find a path in 100% of the cases, by adding TURN
over UDP then TURN over TCP/TLS, then TURN over Websockets as the lowest
priority in the gathered candidate addresses (that will probably require a
modification in RFC 5245, as it already recommends a priority of 0 for TURN
over UDP).

After Taipei I restarted working on this idea, but using Websocket this time.
 I gave up when I discovered that there was no Websocket dissector in
Wireshark, but now that this issue seems to be fixed, perhaps it is a good
idea to finish this work.

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 8/7/12 2:57 AM, Tim Panton wrote:
> Ok, the timing is unfortunate, but can you honestly say that we didn't know that this was skype/microsofts opinion? We just chose to ignore it because it was inconvenient.
>
> Now that it is out there, are we seriously going to ignore a document with _those_ authors from a major browser maker and a team with extensive experience in the field jus because it is late!?!?

I absolutely don't want to discount the extensive collective experience 
and knowledge of how RTC works that went behind the document - but I 
have a hard time treating this proposal as coming from a major browser 
vendor. A proposal from the team behind Skype or Lync is different from 
a proposal that comes from the IE team.

In particular, I strongly disagree with the statement that the existing 
proposal does not honor "key web tenets". As ekr wrote in his analysis, 
the web has moved away from a completely stateless model for a while 
now. In fact, PeerConnection is already more low-level than I would have 
preferred, especially when compared to the original proposal from Ian 
Hickson.

I don't fully grasp how the significantly more complex Javascript API 
proposed in the document is actually beneficial to the modern web as a 
whole.

-Anant


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Subject: Re: [rtcweb] HTTP Fallback draft
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> We indeed met such network elements in our experience, even though most of=
 the times it was just blind UDP filtering rather than RTP filtering per se.=
 It sometimes was just blind port filtering but the effect was the same. Thi=
s mostly happened within enterprises networks where we tried, no matter how s=
mall or large.


[BA] this fits with my experience as well.

> You're right. A couple of years ago we wrote a paper addressing the potent=
ial approaches for tunneling attempts (if you're interested, I can send you t=
he link offline). In this paper we basically described, as a diagram, which i=
ncremental steps could be carried out in order to attempt a successful tunne=
ling: 1) the first attempt is to just try port 443, without encapulating any=
thing (e.g., ssh using 443 instead of 22);

[BA] There are DPI boxes that will compare traffic against TLS and catch thi=
s. So if it doesn't work you can't assume that 443 is blocked by the firewal=
l. Same with non-HTTP on port 80.

> 2) in case that doesn't work, the second attempt is to use HTTP CONNECT an=
d then fall back to 1. for the connection that is established;

[BA] Trying HTTP on port 443 isn't likely to work if the original non-TLS te=
st on 443 failed.=20

> 3) the third attempt (e.g., 443 is not available or the proxy acts as a MI=
TM) is to actually encapsulate in HTTP messages, whether you do HTTP or HTTP=
S. In every case, the peer (either endpoint or server) must be configured ac=
cordingly of course.

[BA] If HTTP failed earlier, HTTP encapsulation also will probably fail. It m=
akes more sense to me to try TLS on 443.

> What I describe in the draft is step 3, even though I guess some words to s=
uggest steps 1 and 2 (where you'd still need to encapsulate RTP packets on t=
op of a TCP-based protocol anyway) could be considered. As long as it looks l=
ike valid HTTP and it behaves accordingly, I think there's no reason why tra=
versing should be impeded:

[BA] DPI boxes aren't always up to date. For example don't expect them to un=
derstand websockets.

> I agree with you and I'm not really dying to do RTP over HTTP either, but i=
f some scenarios make it impossible for use cases to work (and some firewall=
/NAT deployers are to blame here, probably) then a fallback mechanism is som=
ething that can be nice to have, especially if we're interested in something=
 that "just works".

[BA] If all the other avenues are tried first, then this would really be a l=
ast resort. Any idea how frequently it should be expected to be used?=20=

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My perspective is that we can make a huge impact simply by codifying the
rules for doing basic things like TURN over 443 (with or without TLS), HTTP
CONNECT, etc. Running RTP over HTTP could help us get closer to 100%
coverage, but I think we start to enter the area of diminishing returns.

As such, I'd like to see us work towards a best practices document for
establishing sessions in these restricted environments; I think such a
document would be in scope for RTCWEB.


On Tue, Aug 7, 2012 at 11:54 AM, Lorenzo Miniero <lorenzo@meetecho.com>wrote:

> Hello Ted,
>
>
> On Tue, 7 Aug 2012 10:50:07 -0700
> Ted Hardie <ted.ietf@gmail.com> wrote:
>
> > On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero <lorenzo@meetecho.com>
> wrote:
> > > That said, I guess there's a different question I should be asking to
> > > the chairs: since there seems to be no related item in the milestones,
> > > is such a work actually in line with what is the expected outcome of
> the
> > > WG? Considering the draft basically addresses a new transport for RTP
> > > and something that probably needs to be negotiated as well, I guess
> this
> > > could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> > > Nevertheless, my feeling is it belongs more here than somewhere else,
> > > especially considering we're specifying a solution that will be
> deployed
> > > in browsers and, as such, people will expect it to work wherever other
> > > web applications do.
> >
> > My take on this is that the actual work on developing the alternate
> > transport for RTP would have to occur elsewhere and, frankly, I think
> > it is a large enough task that it would likely require its own working
> > group (much as the RTP congestion control topic ended up as a BoF and
> > hopefully will become its own working group).  That doesn't mean that
> > the work couldn't be informed by the RTCWEB use cases, but I think it
> > would have to be done elsewhere.
> >
> > I'd personally suggest starting with a discussion with the ADs on
> > whether a BoF on this topic would be something they might consider.
> > (Note, however, that I have not talked to Cullen about this and Magnus
> > is on vacation, so this is not a "Chairs' response"; just my own
> > thoughts).
> >
>
>
> This makes sense. I'm a bit concerned about the additional time that may
> be needed by going through the process of forming a new WG (compared to
> just adding a milestone to an existing WG, that is), as the final result
> may end up being available much after the original WG completed its works,
> but I see your point.
>
> I'll wait for more feedback about this and, if enough people seem
> interested about doing something like this, contacting the ADs and consider
> the next steps may be a good idea.
>
> Lorenzo
>
>
> > regards,
> >
> > Ted Hardie
>
>
> --
> Lorenzo Miniero, COB
>
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf30363ebbe181b104c6b8a2b4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

My perspective is that we can make a huge impact simply by codifying the ru=
les for doing basic things like TURN over 443 (with or without TLS), HTTP C=
ONNECT, etc. Running RTP over HTTP could help us get closer to 100% coverag=
e, but I think we start to enter the area of diminishing returns.<div>

<br></div><div>As such, I&#39;d like to see us work towards a best practice=
s document for establishing sessions in these restricted environments; I th=
ink such a document would be in scope for RTCWEB.</div><div class=3D"gmail_=
extra">

<br><br><div class=3D"gmail_quote">On Tue, Aug 7, 2012 at 11:54 AM, Lorenzo=
 Miniero <span dir=3D"ltr">&lt;<a href=3D"mailto:lorenzo@meetecho.com" targ=
et=3D"_blank">lorenzo@meetecho.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">

Hello Ted,<br>
<div><div class=3D"h5"><br>
<br>
On Tue, 7 Aug 2012 10:50:07 -0700<br>
Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>=
&gt; wrote:<br>
<br>
&gt; On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero &lt;<a href=3D"mailto:=
lorenzo@meetecho.com">lorenzo@meetecho.com</a>&gt; wrote:<br>
&gt; &gt; That said, I guess there&#39;s a different question I should be a=
sking to<br>
&gt; &gt; the chairs: since there seems to be no related item in the milest=
ones,<br>
&gt; &gt; is such a work actually in line with what is the expected outcome=
 of the<br>
&gt; &gt; WG? Considering the draft basically addresses a new transport for=
 RTP<br>
&gt; &gt; and something that probably needs to be negotiated as well, I gue=
ss this<br>
&gt; &gt; could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).<br=
>
&gt; &gt; Nevertheless, my feeling is it belongs more here than somewhere e=
lse,<br>
&gt; &gt; especially considering we&#39;re specifying a solution that will =
be deployed<br>
&gt; &gt; in browsers and, as such, people will expect it to work wherever =
other<br>
&gt; &gt; web applications do.<br>
&gt;<br>
&gt; My take on this is that the actual work on developing the alternate<br=
>
&gt; transport for RTP would have to occur elsewhere and, frankly, I think<=
br>
&gt; it is a large enough task that it would likely require its own working=
<br>
&gt; group (much as the RTP congestion control topic ended up as a BoF and<=
br>
&gt; hopefully will become its own working group). =C2=A0That doesn&#39;t m=
ean that<br>
&gt; the work couldn&#39;t be informed by the RTCWEB use cases, but I think=
 it<br>
&gt; would have to be done elsewhere.<br>
&gt;<br>
&gt; I&#39;d personally suggest starting with a discussion with the ADs on<=
br>
&gt; whether a BoF on this topic would be something they might consider.<br=
>
&gt; (Note, however, that I have not talked to Cullen about this and Magnus=
<br>
&gt; is on vacation, so this is not a &quot;Chairs&#39; response&quot;; jus=
t my own<br>
&gt; thoughts).<br>
&gt;<br>
<br>
<br>
</div></div>This makes sense. I&#39;m a bit concerned about the additional =
time that may be needed by going through the process of forming a new WG (c=
ompared to just adding a milestone to an existing WG, that is), as the fina=
l result may end up being available much after the original WG completed it=
s works, but I see your point.<br>


<br>
I&#39;ll wait for more feedback about this and, if enough people seem inter=
ested about doing something like this, contacting the ADs and consider the =
next steps may be a good idea.<br>
<br>
Lorenzo<br>
<br>
<br>
&gt; regards,<br>
&gt;<br>
&gt; Ted Hardie<br>
<div class=3D"im HOEnZb"><br>
<br>
--<br>
Lorenzo Miniero, COB<br>
<br>
Meetecho s.r.l.<br>
Web Conferencing and Collaboration Tools<br>
<a href=3D"http://www.meetecho.com" target=3D"_blank">http://www.meetecho.c=
om</a><br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--20cf30363ebbe181b104c6b8a2b4--

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, Lorenzo Miniero <lorenzo@meetecho.com>
Thread-Topic: [rtcweb] HTTP Fallback draft
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bernard Aboba
> Sent: Wednesday, August 08, 2012 7:22 AM
> To: Lorenzo Miniero
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] HTTP Fallback draft
>=20
> > We indeed met such network elements in our experience, even though
> most of the times it was just blind UDP filtering rather than RTP
> filtering per se. It sometimes was just blind port filtering but the
> effect was the same. This mostly happened within enterprises networks
> where we tried, no matter how small or large.
>=20
>=20
> [BA] this fits with my experience as well.
>=20
> > You're right. A couple of years ago we wrote a paper addressing the
> potential approaches for tunneling attempts (if you're interested, I
> can send you the link offline). In this paper we basically described,
> as a diagram, which incremental steps could be carried out in order to
> attempt a successful tunneling: 1) the first attempt is to just try
> port 443, without encapulating anything (e.g., ssh using 443 instead of
> 22);
>=20
> [BA] There are DPI boxes that will compare traffic against TLS and
> catch this. So if it doesn't work you can't assume that 443 is blocked
> by the firewall. Same with non-HTTP on port 80.
>=20
> > 2) in case that doesn't work, the second attempt is to use HTTP
> CONNECT and then fall back to 1. for the connection that is
> established;
>=20
> [BA] Trying HTTP on port 443 isn't likely to work if the original non-
> TLS test on 443 failed.
>=20
> > 3) the third attempt (e.g., 443 is not available or the proxy acts as
> a MITM) is to actually encapsulate in HTTP messages, whether you do
> HTTP or HTTPS. In every case, the peer (either endpoint or server) must
> be configured accordingly of course.
>=20
> [BA] If HTTP failed earlier, HTTP encapsulation also will probably
> fail. It makes more sense to me to try TLS on 443.

[Tiru] Firewalls with TLS Proxy capability can detect such misuse and block=
. we have an alternate proposal to permit UDP flows across firewall
http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00=20

>=20
> > What I describe in the draft is step 3, even though I guess some
> words to suggest steps 1 and 2 (where you'd still need to encapsulate
> RTP packets on top of a TCP-based protocol anyway) could be considered.
> As long as it looks like valid HTTP and it behaves accordingly, I think
> there's no reason why traversing should be impeded:
>=20
> [BA] DPI boxes aren't always up to date. For example don't expect them
> to understand websockets.
>=20
> > I agree with you and I'm not really dying to do RTP over HTTP either,
> but if some scenarios make it impossible for use cases to work (and
> some firewall/NAT deployers are to blame here, probably) then a
> fallback mechanism is something that can be nice to have, especially if
> we're interested in something that "just works".
>=20
> [BA] If all the other avenues are tried first, then this would really
> be a last resort. Any idea how frequently it should be expected to be
> used?
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From lorenzo@meetecho.com  Wed Aug  8 03:09:31 2012
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Date: Wed, 8 Aug 2012 12:03:18 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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Hi Bernard,

answering inline,


On Tue, 7 Aug 2012 18:51:39 -0700
Bernard Aboba <bernard_aboba@hotmail.com> wrote:

> > We indeed met such network elements in our experience, even though most of the times it was just blind UDP filtering rather than RTP filtering per se. It sometimes was just blind port filtering but the effect was the same. This mostly happened within enterprises networks where we tried, no matter how small or large.
> 
> 
> [BA] this fits with my experience as well.
> 
> > You're right. A couple of years ago we wrote a paper addressing the potential approaches for tunneling attempts (if you're interested, I can send you the link offline). In this paper we basically described, as a diagram, which incremental steps could be carried out in order to attempt a successful tunneling: 1) the first attempt is to just try port 443, without encapulating anything (e.g., ssh using 443 instead of 22);
> 
> [BA] There are DPI boxes that will compare traffic against TLS and catch this. So if it doesn't work you can't assume that 443 is blocked by the firewall. Same with non-HTTP on port 80.
> 


You're right, actually I didn't fully describe the approach we presented in that paper. In that approach, infact, we did made use of TLS to encapsulate the traffic to tunnel (in the paper we tunneled everything we needed, not just RTP) on 443, with or without the CONNECT. So it was not a blind piping stuff on 443 or 80 (which may still work in a few cases, though), which means that other approaches like the TURN/TLS:443 could work fine there.


> > 2) in case that doesn't work, the second attempt is to use HTTP CONNECT and then fall back to 1. for the connection that is established;
> 
> [BA] Trying HTTP on port 443 isn't likely to work if the original non-TLS test on 443 failed. 
> 


The CONNECT approach as we conceived it in the paper at the time actually sends an HTTP CONNECT on port 80 to request an encrypted connection towards a 443 destination. If the request is successfull, we have the same TLS stuff as in the direct approach. Or are you talking about something different?


> > 3) the third attempt (e.g., 443 is not available or the proxy acts as a MITM) is to actually encapsulate in HTTP messages, whether you do HTTP or HTTPS. In every case, the peer (either endpoint or server) must be configured accordingly of course.
> 
> [BA] If HTTP failed earlier, HTTP encapsulation also will probably fail. It makes more sense to me to try TLS on 443.
> 
> > What I describe in the draft is step 3, even though I guess some words to suggest steps 1 and 2 (where you'd still need to encapsulate RTP packets on top of a TCP-based protocol anyway) could be considered. As long as it looks like valid HTTP and it behaves accordingly, I think there's no reason why traversing should be impeded:
> 
> [BA] DPI boxes aren't always up to date. For example don't expect them to understand websockets.
> 


That's true, and that's why in the draft I proposed a more "basic" approach: that is, relying on a simpler behaviour for HTTP, chunked or not GET and POST messages, in order to look as much as possible as normal HTTP. Probably not very efficient, but more likely to be accepted by intermediaries.


> > I agree with you and I'm not really dying to do RTP over HTTP either, but if some scenarios make it impossible for use cases to work (and some firewall/NAT deployers are to blame here, probably) then a fallback mechanism is something that can be nice to have, especially if we're interested in something that "just works".
> 
> [BA] If all the other avenues are tried first, then this would really be a last resort. Any idea how frequently it should be expected to be used? 


The ideal scenario would be "never" :)

Unfortunately, there are still cases where such an approach may be needed. I'm not sure how often this could happen, though, as it probably depends on how many deployments that enforce such limitations are out there: I'm not sure it's just prisons or military as Roman said, as you just need to properly set up a proxy like Squid to do something like that cheaply, so there could be more.

Thanks,
Lorenzo

-- 
Lorenzo Miniero, COB

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

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Hi Justin,


On Tue, 7 Aug 2012 20:23:15 -0700
Justin Uberti <juberti@google.com> wrote:

> My perspective is that we can make a huge impact simply by codifying the
> rules for doing basic things like TURN over 443 (with or without TLS), HTTP
> CONNECT, etc. Running RTP over HTTP could help us get closer to 100%
> coverage, but I think we start to enter the area of diminishing returns.
> 
> As such, I'd like to see us work towards a best practices document for
> establishing sessions in these restricted environments; I think such a
> document would be in scope for RTCWEB.
> 


You're right, according to the feedback gathered on the draft so far this looks like a reasonable way to move on. I'm not sure a pure BCP would be enough, though, considering that too many documented different approaches, no matter how effective each, could eventually lead to a few interoperability issues (some browsers/relays may end up implementing some and not others). Maybe something in line with a BCP that at least mandates a certain sequence of or logic in attempts?

Lorenzo


> 
> On Tue, Aug 7, 2012 at 11:54 AM, Lorenzo Miniero <lorenzo@meetecho.com>wrote:
> 
> > Hello Ted,
> >
> >
> > On Tue, 7 Aug 2012 10:50:07 -0700
> > Ted Hardie <ted.ietf@gmail.com> wrote:
> >
> > > On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero <lorenzo@meetecho.com>
> > wrote:
> > > > That said, I guess there's a different question I should be asking to
> > > > the chairs: since there seems to be no related item in the milestones,
> > > > is such a work actually in line with what is the expected outcome of
> > the
> > > > WG? Considering the draft basically addresses a new transport for RTP
> > > > and something that probably needs to be negotiated as well, I guess
> > this
> > > > could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> > > > Nevertheless, my feeling is it belongs more here than somewhere else,
> > > > especially considering we're specifying a solution that will be
> > deployed
> > > > in browsers and, as such, people will expect it to work wherever other
> > > > web applications do.
> > >
> > > My take on this is that the actual work on developing the alternate
> > > transport for RTP would have to occur elsewhere and, frankly, I think
> > > it is a large enough task that it would likely require its own working
> > > group (much as the RTP congestion control topic ended up as a BoF and
> > > hopefully will become its own working group).  That doesn't mean that
> > > the work couldn't be informed by the RTCWEB use cases, but I think it
> > > would have to be done elsewhere.
> > >
> > > I'd personally suggest starting with a discussion with the ADs on
> > > whether a BoF on this topic would be something they might consider.
> > > (Note, however, that I have not talked to Cullen about this and Magnus
> > > is on vacation, so this is not a "Chairs' response"; just my own
> > > thoughts).
> > >
> >
> >
> > This makes sense. I'm a bit concerned about the additional time that may
> > be needed by going through the process of forming a new WG (compared to
> > just adding a milestone to an existing WG, that is), as the final result
> > may end up being available much after the original WG completed its works,
> > but I see your point.
> >
> > I'll wait for more feedback about this and, if enough people seem
> > interested about doing something like this, contacting the ADs and consider
> > the next steps may be a good idea.
> >
> > Lorenzo
> >
> >
> > > regards,
> > >
> > > Ted Hardie
> >
> >
> > --
> > Lorenzo Miniero, COB
> >
> > Meetecho s.r.l.
> > Web Conferencing and Collaboration Tools
> > http://www.meetecho.com
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >


-- 
Lorenzo Miniero, COB

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

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Date: Wed, 8 Aug 2012 08:05:28 -0700
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From: Cameron Byrne <cb.list6@gmail.com>
To: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
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Subject: Re: [rtcweb] HTTP Fallback draft
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On Aug 7, 2012 11:18 PM, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
wrote:
>
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Bernard Aboba
> > Sent: Wednesday, August 08, 2012 7:22 AM
> > To: Lorenzo Miniero
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] HTTP Fallback draft
> >
> > > We indeed met such network elements in our experience, even though
> > most of the times it was just blind UDP filtering rather than RTP
> > filtering per se. It sometimes was just blind port filtering but the
> > effect was the same. This mostly happened within enterprises networks
> > where we tried, no matter how small or large.
> >
> >
> > [BA] this fits with my experience as well.
> >
> > > You're right. A couple of years ago we wrote a paper addressing the
> > potential approaches for tunneling attempts (if you're interested, I
> > can send you the link offline). In this paper we basically described,
> > as a diagram, which incremental steps could be carried out in order to
> > attempt a successful tunneling: 1) the first attempt is to just try
> > port 443, without encapulating anything (e.g., ssh using 443 instead of
> > 22);
> >
> > [BA] There are DPI boxes that will compare traffic against TLS and
> > catch this. So if it doesn't work you can't assume that 443 is blocked
> > by the firewall. Same with non-HTTP on port 80.
> >
> > > 2) in case that doesn't work, the second attempt is to use HTTP
> > CONNECT and then fall back to 1. for the connection that is
> > established;
> >
> > [BA] Trying HTTP on port 443 isn't likely to work if the original non-
> > TLS test on 443 failed.
> >
> > > 3) the third attempt (e.g., 443 is not available or the proxy acts as
> > a MITM) is to actually encapsulate in HTTP messages, whether you do
> > HTTP or HTTPS. In every case, the peer (either endpoint or server) must
> > be configured accordingly of course.
> >
> > [BA] If HTTP failed earlier, HTTP encapsulation also will probably
> > fail. It makes more sense to me to try TLS on 443.
>
> [Tiru] Firewalls with TLS Proxy capability can detect such misuse and
block. we have an alternate proposal to permit UDP flows across firewall
> http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00
>

Is this thread  really about the ietf engineering a way to by-pass network
policy set by network operators?

I do not believe that is acceptable.

CB

> >
> > > What I describe in the draft is step 3, even though I guess some
> > words to suggest steps 1 and 2 (where you'd still need to encapsulate
> > RTP packets on top of a TCP-based protocol anyway) could be considered.
> > As long as it looks like valid HTTP and it behaves accordingly, I think
> > there's no reason why traversing should be impeded:
> >
> > [BA] DPI boxes aren't always up to date. For example don't expect them
> > to understand websockets.
> >
> > > I agree with you and I'm not really dying to do RTP over HTTP either,
> > but if some scenarios make it impossible for use cases to work (and
> > some firewall/NAT deployers are to blame here, probably) then a
> > fallback mechanism is something that can be nice to have, especially if
> > we're interested in something that "just works".
> >
> > [BA] If all the other avenues are tried first, then this would really
> > be a last resort. Any idea how frequently it should be expected to be
> > used?
> > _______________________________________________
> > 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

--f46d040169b50904a804c6c2712b
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<p><br>
On Aug 7, 2012 11:18 PM, &quot;Tirumaleswar Reddy (tireddy)&quot; &lt;<a hr=
ef=3D"mailto:tireddy@cisco.com">tireddy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of Bernard Aboba<br>
&gt; &gt; Sent: Wednesday, August 08, 2012 7:22 AM<br>
&gt; &gt; To: Lorenzo Miniero<br>
&gt; &gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; Subject: Re: [rtcweb] HTTP Fallback draft<br>
&gt; &gt;<br>
&gt; &gt; &gt; We indeed met such network elements in our experience, even =
though<br>
&gt; &gt; most of the times it was just blind UDP filtering rather than RTP=
<br>
&gt; &gt; filtering per se. It sometimes was just blind port filtering but =
the<br>
&gt; &gt; effect was the same. This mostly happened within enterprises netw=
orks<br>
&gt; &gt; where we tried, no matter how small or large.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; [BA] this fits with my experience as well.<br>
&gt; &gt;<br>
&gt; &gt; &gt; You&#39;re right. A couple of years ago we wrote a paper add=
ressing the<br>
&gt; &gt; potential approaches for tunneling attempts (if you&#39;re intere=
sted, I<br>
&gt; &gt; can send you the link offline). In this paper we basically descri=
bed,<br>
&gt; &gt; as a diagram, which incremental steps could be carried out in ord=
er to<br>
&gt; &gt; attempt a successful tunneling: 1) the first attempt is to just t=
ry<br>
&gt; &gt; port 443, without encapulating anything (e.g., ssh using 443 inst=
ead of<br>
&gt; &gt; 22);<br>
&gt; &gt;<br>
&gt; &gt; [BA] There are DPI boxes that will compare traffic against TLS an=
d<br>
&gt; &gt; catch this. So if it doesn&#39;t work you can&#39;t assume that 4=
43 is blocked<br>
&gt; &gt; by the firewall. Same with non-HTTP on port 80.<br>
&gt; &gt;<br>
&gt; &gt; &gt; 2) in case that doesn&#39;t work, the second attempt is to u=
se HTTP<br>
&gt; &gt; CONNECT and then fall back to 1. for the connection that is<br>
&gt; &gt; established;<br>
&gt; &gt;<br>
&gt; &gt; [BA] Trying HTTP on port 443 isn&#39;t likely to work if the orig=
inal non-<br>
&gt; &gt; TLS test on 443 failed.<br>
&gt; &gt;<br>
&gt; &gt; &gt; 3) the third attempt (e.g., 443 is not available or the prox=
y acts as<br>
&gt; &gt; a MITM) is to actually encapsulate in HTTP messages, whether you =
do<br>
&gt; &gt; HTTP or HTTPS. In every case, the peer (either endpoint or server=
) must<br>
&gt; &gt; be configured accordingly of course.<br>
&gt; &gt;<br>
&gt; &gt; [BA] If HTTP failed earlier, HTTP encapsulation also will probabl=
y<br>
&gt; &gt; fail. It makes more sense to me to try TLS on 443.<br>
&gt;<br>
&gt; [Tiru] Firewalls with TLS Proxy capability can detect such misuse and =
block. we have an alternate proposal to permit UDP flows across firewall<br=
>
&gt; <a href=3D"http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-=
traversal-00">http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-tr=
aversal-00</a><br>
&gt;</p>
<p>Is this thread=A0 really about the ietf engineering a way to by-pass net=
work policy set by network operators?</p>
<p>I do not believe that is acceptable.</p>
<p>CB</p>
<p>&gt; &gt;<br>
&gt; &gt; &gt; What I describe in the draft is step 3, even though I guess =
some<br>
&gt; &gt; words to suggest steps 1 and 2 (where you&#39;d still need to enc=
apsulate<br>
&gt; &gt; RTP packets on top of a TCP-based protocol anyway) could be consi=
dered.<br>
&gt; &gt; As long as it looks like valid HTTP and it behaves accordingly, I=
 think<br>
&gt; &gt; there&#39;s no reason why traversing should be impeded:<br>
&gt; &gt;<br>
&gt; &gt; [BA] DPI boxes aren&#39;t always up to date. For example don&#39;=
t expect them<br>
&gt; &gt; to understand websockets.<br>
&gt; &gt;<br>
&gt; &gt; &gt; I agree with you and I&#39;m not really dying to do RTP over=
 HTTP either,<br>
&gt; &gt; but if some scenarios make it impossible for use cases to work (a=
nd<br>
&gt; &gt; some firewall/NAT deployers are to blame here, probably) then a<b=
r>
&gt; &gt; fallback mechanism is something that can be nice to have, especia=
lly if<br>
&gt; &gt; we&#39;re interested in something that &quot;just works&quot;.<br=
>
&gt; &gt;<br>
&gt; &gt; [BA] If all the other avenues are tried first, then this would re=
ally<br>
&gt; &gt; be a last resort. Any idea how frequently it should be expected t=
o be<br>
&gt; &gt; used?<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; 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>
</p>

--f46d040169b50904a804c6c2712b--

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2012/8/8 Cameron Byrne <cb.list6@gmail.com>:
> Is this thread  really about the ietf engineering a way to by-pass networ=
k
> policy set by network operators?
>
> I do not believe that is acceptable.

Not exactly. Imagine the classic example of hotels providing free WiFi
(but just for HTTP and HTTPS). They do allow their clients to visit
web pages and, therefore, I could expect that they would also allow
any kind of communication that takes place within the web page.

WebRTC introduces UDP, RTP, ICE and more stuf into the WWW world.
These protocols could be blocked in some scenarios (hotel WiFi) but
that does not necessarily mean that the network admins want to block
WebRTC (since it occurs "within" the WWW they already allow).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Cameron Byrne <cb.list6@gmail.com>
Thread-Topic: [rtcweb] HTTP Fallback draft
Thread-Index: AQHNdLbePFtYWCPhFk+Ws7zgJPkdgJdO4KUAgAAJYQCAAJERgP//9WQggADoZgD//63vwA==
Date: Wed, 8 Aug 2012 15:21:09 +0000
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References: <20120807180156.286e74d2@rainpc> <D5BDA7BE-FE55-47FA-99FD-1645084370B0@gmx.net> <20120807191226.5b8e7f32@rainpc> <BLU401-EAS2566333A7F4DEA0D3BFDBE593CD0@phx.gbl> <913383AAA69FF945B8F946018B75898A1477EDB9@xmb-rcd-x10.cisco.com> <CAD6AjGSU8mzbcdbOkgGtAms1tdHhjiuQn_NFXELwO2kjegJkCQ@mail.gmail.com>
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Subject: Re: [rtcweb] HTTP Fallback draft
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> [Tiru] Firewalls with TLS Proxy capability can detect such misuse and blo=
ck. we have an alternate proposal to permit UDP flows across firewall
> http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00
>

Is this thread  really about the ietf engineering a way to by-pass network =
policy set by network operators?

I do not believe that is acceptable.

CB
[Tiru] The above technique is not to by bass Enterprise Firewall policy, bu=
t have Firewall policy to permit nominated UDP flows only. For example Ente=
rprise can permit calls to selected WebRTC servers that it trusts (let's sa=
y Dr.Good) and block others (like Dr. Evil). This draft is  solving the pro=
blem that in such cases permit call to be successful with WebRTC server Dr.=
Good and permit UDP flows for those calls without the need to have a ALG on=
 Firewall.


From: Cameron Byrne [mailto:cb.list6@gmail.com]
Sent: Wednesday, August 08, 2012 8:35 PM
To: Tirumaleswar Reddy (tireddy)
Cc: rtcweb@ietf.org; Lorenzo Miniero; Bernard Aboba
Subject: Re: [rtcweb] HTTP Fallback draft


On Aug 7, 2012 11:18 PM, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com<=
mailto:tireddy@cisco.com>> wrote:
>
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:r=
tcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On
> > Behalf Of Bernard Aboba
> > Sent: Wednesday, August 08, 2012 7:22 AM
> > To: Lorenzo Miniero
> > Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > Subject: Re: [rtcweb] HTTP Fallback draft
> >
> > > We indeed met such network elements in our experience, even though
> > most of the times it was just blind UDP filtering rather than RTP
> > filtering per se. It sometimes was just blind port filtering but the
> > effect was the same. This mostly happened within enterprises networks
> > where we tried, no matter how small or large.
> >
> >
> > [BA] this fits with my experience as well.
> >
> > > You're right. A couple of years ago we wrote a paper addressing the
> > potential approaches for tunneling attempts (if you're interested, I
> > can send you the link offline). In this paper we basically described,
> > as a diagram, which incremental steps could be carried out in order to
> > attempt a successful tunneling: 1) the first attempt is to just try
> > port 443, without encapulating anything (e.g., ssh using 443 instead of
> > 22);
> >
> > [BA] There are DPI boxes that will compare traffic against TLS and
> > catch this. So if it doesn't work you can't assume that 443 is blocked
> > by the firewall. Same with non-HTTP on port 80.
> >
> > > 2) in case that doesn't work, the second attempt is to use HTTP
> > CONNECT and then fall back to 1. for the connection that is
> > established;
> >
> > [BA] Trying HTTP on port 443 isn't likely to work if the original non-
> > TLS test on 443 failed.
> >
> > > 3) the third attempt (e.g., 443 is not available or the proxy acts as
> > a MITM) is to actually encapsulate in HTTP messages, whether you do
> > HTTP or HTTPS. In every case, the peer (either endpoint or server) must
> > be configured accordingly of course.
> >
> > [BA] If HTTP failed earlier, HTTP encapsulation also will probably
> > fail. It makes more sense to me to try TLS on 443.
>
> [Tiru] Firewalls with TLS Proxy capability can detect such misuse and blo=
ck. we have an alternate proposal to permit UDP flows across firewall
> http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00
>

Is this thread  really about the ietf engineering a way to by-pass network =
policy set by network operators?

I do not believe that is acceptable.

CB

> >
> > > What I describe in the draft is step 3, even though I guess some
> > words to suggest steps 1 and 2 (where you'd still need to encapsulate
> > RTP packets on top of a TCP-based protocol anyway) could be considered.
> > As long as it looks like valid HTTP and it behaves accordingly, I think
> > there's no reason why traversing should be impeded:
> >
> > [BA] DPI boxes aren't always up to date. For example don't expect them
> > to understand websockets.
> >
> > > I agree with you and I'm not really dying to do RTP over HTTP either,
> > but if some scenarios make it impossible for use cases to work (and
> > some firewall/NAT deployers are to blame here, probably) then a
> > fallback mechanism is something that can be nice to have, especially if
> > we're interested in something that "just works".
> >
> > [BA] If all the other avenues are tried first, then this would really
> > be a last resort. Any idea how frequently it should be expected to be
> > used?
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

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<div class=3D"Section1">
<p>&gt; [Tiru] Firewalls with TLS Proxy capability can detect such misuse a=
nd block. we have an alternate proposal to permit UDP flows across firewall=
<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-=
traversal-00">
http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00</a>=
<br>
&gt;<o:p></o:p></p>
<p>Is this thread&nbsp; really about the ietf engineering a way to by-pass =
network policy set by network operators?<o:p></o:p></p>
<p>I do not believe that is acceptable.<o:p></o:p></p>
<p>CB<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">[Tiru] The above technique is not to by bass Enterprise Fire=
wall policy, but have Firewall policy to permit nominated UDP flows only. F=
or example Enterprise
 can permit calls to selected WebRTC servers that it trusts (let&#8217;s sa=
y Dr.Good) and block others (like Dr. Evil). This draft is&nbsp; solving th=
e problem that in such cases permit call to be successful with WebRTC serve=
r Dr.Good and permit UDP flows for those calls
 without the need to have a ALG on Firewall.<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>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Cameron =
Byrne [mailto:cb.list6@gmail.com]
<br>
<b>Sent:</b> Wednesday, August 08, 2012 8:35 PM<br>
<b>To:</b> Tirumaleswar Reddy (tireddy)<br>
<b>Cc:</b> rtcweb@ietf.org; Lorenzo Miniero; Bernard Aboba<br>
<b>Subject:</b> Re: [rtcweb] HTTP Fallback draft<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p><br>
On Aug 7, 2012 11:18 PM, &quot;Tirumaleswar Reddy (tireddy)&quot; &lt;<a hr=
ef=3D"mailto:tireddy@cisco.com">tireddy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of Bernard Aboba<br>
&gt; &gt; Sent: Wednesday, August 08, 2012 7:22 AM<br>
&gt; &gt; To: Lorenzo Miniero<br>
&gt; &gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; Subject: Re: [rtcweb] HTTP Fallback draft<br>
&gt; &gt;<br>
&gt; &gt; &gt; We indeed met such network elements in our experience, even =
though<br>
&gt; &gt; most of the times it was just blind UDP filtering rather than RTP=
<br>
&gt; &gt; filtering per se. It sometimes was just blind port filtering but =
the<br>
&gt; &gt; effect was the same. This mostly happened within enterprises netw=
orks<br>
&gt; &gt; where we tried, no matter how small or large.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; [BA] this fits with my experience as well.<br>
&gt; &gt;<br>
&gt; &gt; &gt; You're right. A couple of years ago we wrote a paper address=
ing the<br>
&gt; &gt; potential approaches for tunneling attempts (if you're interested=
, I<br>
&gt; &gt; can send you the link offline). In this paper we basically descri=
bed,<br>
&gt; &gt; as a diagram, which incremental steps could be carried out in ord=
er to<br>
&gt; &gt; attempt a successful tunneling: 1) the first attempt is to just t=
ry<br>
&gt; &gt; port 443, without encapulating anything (e.g., ssh using 443 inst=
ead of<br>
&gt; &gt; 22);<br>
&gt; &gt;<br>
&gt; &gt; [BA] There are DPI boxes that will compare traffic against TLS an=
d<br>
&gt; &gt; catch this. So if it doesn't work you can't assume that 443 is bl=
ocked<br>
&gt; &gt; by the firewall. Same with non-HTTP on port 80.<br>
&gt; &gt;<br>
&gt; &gt; &gt; 2) in case that doesn't work, the second attempt is to use H=
TTP<br>
&gt; &gt; CONNECT and then fall back to 1. for the connection that is<br>
&gt; &gt; established;<br>
&gt; &gt;<br>
&gt; &gt; [BA] Trying HTTP on port 443 isn't likely to work if the original=
 non-<br>
&gt; &gt; TLS test on 443 failed.<br>
&gt; &gt;<br>
&gt; &gt; &gt; 3) the third attempt (e.g., 443 is not available or the prox=
y acts as<br>
&gt; &gt; a MITM) is to actually encapsulate in HTTP messages, whether you =
do<br>
&gt; &gt; HTTP or HTTPS. In every case, the peer (either endpoint or server=
) must<br>
&gt; &gt; be configured accordingly of course.<br>
&gt; &gt;<br>
&gt; &gt; [BA] If HTTP failed earlier, HTTP encapsulation also will probabl=
y<br>
&gt; &gt; fail. It makes more sense to me to try TLS on 443.<br>
&gt;<br>
&gt; [Tiru] Firewalls with TLS Proxy capability can detect such misuse and =
block. we have an alternate proposal to permit UDP flows across firewall<br=
>
&gt; <a href=3D"http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-=
traversal-00">
http://tools.ietf.org/html/draft-reddy-rtcweb-stun-auth-fw-traversal-00</a>=
<br>
&gt;<o:p></o:p></p>
<p>Is this thread&nbsp; really about the ietf engineering a way to by-pass =
network policy set by network operators?<o:p></o:p></p>
<p>I do not believe that is acceptable.<o:p></o:p></p>
<p>CB<o:p></o:p></p>
<p>&gt; &gt;<br>
&gt; &gt; &gt; What I describe in the draft is step 3, even though I guess =
some<br>
&gt; &gt; words to suggest steps 1 and 2 (where you'd still need to encapsu=
late<br>
&gt; &gt; RTP packets on top of a TCP-based protocol anyway) could be consi=
dered.<br>
&gt; &gt; As long as it looks like valid HTTP and it behaves accordingly, I=
 think<br>
&gt; &gt; there's no reason why traversing should be impeded:<br>
&gt; &gt;<br>
&gt; &gt; [BA] DPI boxes aren't always up to date. For example don't expect=
 them<br>
&gt; &gt; to understand websockets.<br>
&gt; &gt;<br>
&gt; &gt; &gt; I agree with you and I'm not really dying to do RTP over HTT=
P either,<br>
&gt; &gt; but if some scenarios make it impossible for use cases to work (a=
nd<br>
&gt; &gt; some firewall/NAT deployers are to blame here, probably) then a<b=
r>
&gt; &gt; fallback mechanism is something that can be nice to have, especia=
lly if<br>
&gt; &gt; we're interested in something that &quot;just works&quot;.<br>
&gt; &gt;<br>
&gt; &gt; [BA] If all the other avenues are tried first, then this would re=
ally<br>
&gt; &gt; be a last resort. Any idea how frequently it should be expected t=
o be<br>
&gt; &gt; used?<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; 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><o:p></o:p></p>
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Subject: [rtcweb] Appropriateness of bypass mechanisms (Re: HTTP Fallback draft)
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Forking thread....

On 08/08/2012 05:05 PM, Cameron Byrne wrote:
>
>
> Is this thread  really about the ietf engineering a way to by-pass 
> network policy set by network operators?
>
It's about getting stuff done in the presence of rules that hinder the 
"simple way".
Those rules may be set that way deliberately, by ignorance, because of 
implementation limitations, or in error; there's no way to know.

> I do not believe that is acceptable.
>
We've already done TURN, including TURN over TLS.

Old quote: "This is not about starting down the slippery slope; this is 
about how far we slide into the muck at the bottom".



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Cc: Alexey Aylarov <alexey@zingaya.com>, "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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On 08/07/2012 11:57 AM, Tim Panton wrote:
> Ok, the timing is unfortunate, but can you honestly say that we didn't know that this was skype/microsofts opinion? We just chose to ignore it because it was inconvenient.
>
> Now that it is out there, are we seriously going to ignore a document with _those_ authors from a major browser maker and a team with extensive experience in the field jus because it is late!?!?
Speaking with WG chair hat on:

At the time of previous "what is the appropriate level" discussion, the 
WG chairs concluded that there was a rough WG consensus to stay with a 
higher-level API rather than trying to move the level of the API 
downwards towards a "low level API".

Since that time and until this week, there has been no specific proposal 
to be evaluated, and the main proponents of such an  API have been 
silent. Normal behaviour is to assume that a previous consensus 
declaration stands until there is new technical evidence to be 
evaluated, and proceed with further work on that basis.

So I can't agree that "because it was inconvenient" is an appropriate 
characterization.

               Harald


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From: Justin Uberti <juberti@google.com>
Date: Wed, 8 Aug 2012 12:02:15 -0700
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Subject: Re: [rtcweb] HTTP Fallback draft
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That makes sense to me. The overall point I was driving at was that we
wouldn't have to invent new protocols, we could just propose the
sequence/logic that you describe, perhaps accompanied by real-world data
that explains the approach.


On Wed, Aug 8, 2012 at 3:08 AM, Lorenzo Miniero <lorenzo@meetecho.com>wrote:

> Hi Justin,
>
>
> On Tue, 7 Aug 2012 20:23:15 -0700
> Justin Uberti <juberti@google.com> wrote:
>
> > My perspective is that we can make a huge impact simply by codifying the
> > rules for doing basic things like TURN over 443 (with or without TLS),
> HTTP
> > CONNECT, etc. Running RTP over HTTP could help us get closer to 100%
> > coverage, but I think we start to enter the area of diminishing returns.
> >
> > As such, I'd like to see us work towards a best practices document for
> > establishing sessions in these restricted environments; I think such a
> > document would be in scope for RTCWEB.
> >
>
>
> You're right, according to the feedback gathered on the draft so far this
> looks like a reasonable way to move on. I'm not sure a pure BCP would be
> enough, though, considering that too many documented different approaches,
> no matter how effective each, could eventually lead to a few
> interoperability issues (some browsers/relays may end up implementing some
> and not others). Maybe something in line with a BCP that at least mandates
> a certain sequence of or logic in attempts?
>
> Lorenzo
>
>
> >
> > On Tue, Aug 7, 2012 at 11:54 AM, Lorenzo Miniero <lorenzo@meetecho.com
> >wrote:
> >
> > > Hello Ted,
> > >
> > >
> > > On Tue, 7 Aug 2012 10:50:07 -0700
> > > Ted Hardie <ted.ietf@gmail.com> wrote:
> > >
> > > > On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero <
> lorenzo@meetecho.com>
> > > wrote:
> > > > > That said, I guess there's a different question I should be asking
> to
> > > > > the chairs: since there seems to be no related item in the
> milestones,
> > > > > is such a work actually in line with what is the expected outcome
> of
> > > the
> > > > > WG? Considering the draft basically addresses a new transport for
> RTP
> > > > > and something that probably needs to be negotiated as well, I guess
> > > this
> > > > > could be seen as belonging elsewhere (AVTCORE and/or MMUSIC?).
> > > > > Nevertheless, my feeling is it belongs more here than somewhere
> else,
> > > > > especially considering we're specifying a solution that will be
> > > deployed
> > > > > in browsers and, as such, people will expect it to work wherever
> other
> > > > > web applications do.
> > > >
> > > > My take on this is that the actual work on developing the alternate
> > > > transport for RTP would have to occur elsewhere and, frankly, I think
> > > > it is a large enough task that it would likely require its own
> working
> > > > group (much as the RTP congestion control topic ended up as a BoF and
> > > > hopefully will become its own working group).  That doesn't mean that
> > > > the work couldn't be informed by the RTCWEB use cases, but I think it
> > > > would have to be done elsewhere.
> > > >
> > > > I'd personally suggest starting with a discussion with the ADs on
> > > > whether a BoF on this topic would be something they might consider.
> > > > (Note, however, that I have not talked to Cullen about this and
> Magnus
> > > > is on vacation, so this is not a "Chairs' response"; just my own
> > > > thoughts).
> > > >
> > >
> > >
> > > This makes sense. I'm a bit concerned about the additional time that
> may
> > > be needed by going through the process of forming a new WG (compared to
> > > just adding a milestone to an existing WG, that is), as the final
> result
> > > may end up being available much after the original WG completed its
> works,
> > > but I see your point.
> > >
> > > I'll wait for more feedback about this and, if enough people seem
> > > interested about doing something like this, contacting the ADs and
> consider
> > > the next steps may be a good idea.
> > >
> > > Lorenzo
> > >
> > >
> > > > regards,
> > > >
> > > > Ted Hardie
> > >
> > >
> > > --
> > > Lorenzo Miniero, COB
> > >
> > > Meetecho s.r.l.
> > > Web Conferencing and Collaboration Tools
> > > http://www.meetecho.com
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > >
>
>
> --
> Lorenzo Miniero, COB
>
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
>

--20cf3074b2d20894ee04c6c5c1da
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That makes sense to me. The overall point I was driving at was that we woul=
dn&#39;t have to invent new protocols, we could just propose the sequence/l=
ogic that you describe, perhaps accompanied by real-world data that explain=
s the approach.<div class=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Wed, Aug 8, 2012 at 3:08 AM, Lorenzo =
Miniero <span dir=3D"ltr">&lt;<a href=3D"mailto:lorenzo@meetecho.com" targe=
t=3D"_blank">lorenzo@meetecho.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">

Hi Justin,<br>
<div class=3D"im"><br>
<br>
On Tue, 7 Aug 2012 20:23:15 -0700<br>
Justin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com<=
/a>&gt; wrote:<br>
<br>
&gt; My perspective is that we can make a huge impact simply by codifying t=
he<br>
&gt; rules for doing basic things like TURN over 443 (with or without TLS),=
 HTTP<br>
&gt; CONNECT, etc. Running RTP over HTTP could help us get closer to 100%<b=
r>
&gt; coverage, but I think we start to enter the area of diminishing return=
s.<br>
&gt;<br>
&gt; As such, I&#39;d like to see us work towards a best practices document=
 for<br>
&gt; establishing sessions in these restricted environments; I think such a=
<br>
&gt; document would be in scope for RTCWEB.<br>
&gt;<br>
<br>
<br>
</div>You&#39;re right, according to the feedback gathered on the draft so =
far this looks like a reasonable way to move on. I&#39;m not sure a pure BC=
P would be enough, though, considering that too many documented different a=
pproaches, no matter how effective each, could eventually lead to a few int=
eroperability issues (some browsers/relays may end up implementing some and=
 not others). Maybe something in line with a BCP that at least mandates a c=
ertain sequence of or logic in attempts?<br>


<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Lorenzo<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt;<br>
&gt; On Tue, Aug 7, 2012 at 11:54 AM, Lorenzo Miniero &lt;<a href=3D"mailto=
:lorenzo@meetecho.com">lorenzo@meetecho.com</a>&gt;wrote:<br>
&gt;<br>
&gt; &gt; Hello Ted,<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On Tue, 7 Aug 2012 10:50:07 -0700<br>
&gt; &gt; Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gma=
il.com</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; &gt; On Tue, Aug 7, 2012 at 9:01 AM, Lorenzo Miniero &lt;<a href=
=3D"mailto:lorenzo@meetecho.com">lorenzo@meetecho.com</a>&gt;<br>
&gt; &gt; wrote:<br>
&gt; &gt; &gt; &gt; That said, I guess there&#39;s a different question I s=
hould be asking to<br>
&gt; &gt; &gt; &gt; the chairs: since there seems to be no related item in =
the milestones,<br>
&gt; &gt; &gt; &gt; is such a work actually in line with what is the expect=
ed outcome of<br>
&gt; &gt; the<br>
&gt; &gt; &gt; &gt; WG? Considering the draft basically addresses a new tra=
nsport for RTP<br>
&gt; &gt; &gt; &gt; and something that probably needs to be negotiated as w=
ell, I guess<br>
&gt; &gt; this<br>
&gt; &gt; &gt; &gt; could be seen as belonging elsewhere (AVTCORE and/or MM=
USIC?).<br>
&gt; &gt; &gt; &gt; Nevertheless, my feeling is it belongs more here than s=
omewhere else,<br>
&gt; &gt; &gt; &gt; especially considering we&#39;re specifying a solution =
that will be<br>
&gt; &gt; deployed<br>
&gt; &gt; &gt; &gt; in browsers and, as such, people will expect it to work=
 wherever other<br>
&gt; &gt; &gt; &gt; web applications do.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; My take on this is that the actual work on developing the al=
ternate<br>
&gt; &gt; &gt; transport for RTP would have to occur elsewhere and, frankly=
, I think<br>
&gt; &gt; &gt; it is a large enough task that it would likely require its o=
wn working<br>
&gt; &gt; &gt; group (much as the RTP congestion control topic ended up as =
a BoF and<br>
&gt; &gt; &gt; hopefully will become its own working group). =C2=A0That doe=
sn&#39;t mean that<br>
&gt; &gt; &gt; the work couldn&#39;t be informed by the RTCWEB use cases, b=
ut I think it<br>
&gt; &gt; &gt; would have to be done elsewhere.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; I&#39;d personally suggest starting with a discussion with t=
he ADs on<br>
&gt; &gt; &gt; whether a BoF on this topic would be something they might co=
nsider.<br>
&gt; &gt; &gt; (Note, however, that I have not talked to Cullen about this =
and Magnus<br>
&gt; &gt; &gt; is on vacation, so this is not a &quot;Chairs&#39; response&=
quot;; just my own<br>
&gt; &gt; &gt; thoughts).<br>
&gt; &gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; This makes sense. I&#39;m a bit concerned about the additional ti=
me that may<br>
&gt; &gt; be needed by going through the process of forming a new WG (compa=
red to<br>
&gt; &gt; just adding a milestone to an existing WG, that is), as the final=
 result<br>
&gt; &gt; may end up being available much after the original WG completed i=
ts works,<br>
&gt; &gt; but I see your point.<br>
&gt; &gt;<br>
&gt; &gt; I&#39;ll wait for more feedback about this and, if enough people =
seem<br>
&gt; &gt; interested about doing something like this, contacting the ADs an=
d consider<br>
&gt; &gt; the next steps may be a good idea.<br>
&gt; &gt;<br>
&gt; &gt; Lorenzo<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; &gt; regards,<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Ted Hardie<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; Lorenzo Miniero, COB<br>
&gt; &gt;<br>
&gt; &gt; Meetecho s.r.l.<br>
&gt; &gt; Web Conferencing and Collaboration Tools<br>
&gt; &gt; <a href=3D"http://www.meetecho.com" target=3D"_blank">http://www.=
meetecho.com</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt;<br>
<br>
<br>
--<br>
Lorenzo Miniero, COB<br>
<br>
Meetecho s.r.l.<br>
Web Conferencing and Collaboration Tools<br>
<a href=3D"http://www.meetecho.com" target=3D"_blank">http://www.meetecho.c=
om</a><br>
</div></div></blockquote></div><br></div>

--20cf3074b2d20894ee04c6c5c1da--

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Howdy,

We'd like to thank Christer and Brian for taking notes for the
session.  If there are folks who have additional notes from the two
sessions which they can share with the chairs or the group, we'd
appreciate it; that would help a lot in making more complete minutes.

thanks,

Ted

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Colin, Magnus,
fyi, some editorial nits,
regards,
Albrecht

Line numbers according
http://tools.ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-rtcw=
eb-rtp-usage-04.txt

Editorial change proposals:

232        RTP Media Stream:  A sequence of RTP packets, and associated RTC=
P
233           packets, using a single synchronisation source (SSRC) that

=3D>      synchronization (according RFC 3550)
=3D>      ... multiple times in doc

295           participants in the session; support for randomised RTCP
=3D> randomized

407        session also effects the possibility for differentiated treatmen=
t of
=3D> affects

528        implemented using a centralised server, multi-unicast, or using =
IP
=3D> centralized
... multiple times

588        Request above as there there may be multiple methods to fulfill =
the
=3D> delete 'there'

694        optimisations of non critical functions, and is hence OPTIONAL t=
o
=3D> optimizations

715        will support negative acknowlegdements (NACKs) for RTP data pack=
ets
=3D> acknowledgements

1247       own traffic.  While is is clearly considered a bug, it is import=
ant
=3D> is is
1248       that the end-point is able to recognise and handle the case when=
 it
=3D> recognize


1670       This in an attempt to highlight the differencies and the in many=
 case
=3D> differences

1718       An RTP session have good support for simultanously transport mul=
tiple
=3D> simultaneously

1723       video cameras can potentially transmitt video from both to its
=3D> transmit

1730       Thus we can introduce a couple of different notiations in the be=
low
=3D> notations

1731       two alternate figures of a single peer connection in a a point t=
o
=3D> a a

1838       different media bit-rates to the differnt peers, thus not forcin=
g B
=3D> different

1839       to endure the same quality reductions if there are limiations in=
 the
=3D> limitations

1874       encoder instances each beeing associated with two different
=3D> being

1883       resource costrained will use a different implementation strategy
=3D> constrained

1918       limited to congestion control, also prefered resolution to recei=
ve
=3D> preferred

1919       based on dispaly area available is another aspect requiring
=3D> display

1920       consideration.  The need for this type of descion logic does ari=
se in
=3D> decision

1949       its operation and form new RTP packets, encrypts and integegrity
=3D> integrity

1970       in a media domain mix of the incomming RTP media streams.  Then
=3D> incoming

1980       produce a Mix of all N streams for the group that are curently n=
ot
=3D> currently

2046       particpants.  As each peer receives a different version produced=
 by
=3D> participants

2070       There exist an possible implementation choice to have the RTP
=3D> "a possible" ?

2074       about the contributing sources.  This removes both the functiona=
lity
=3D> functionality

2078       they can be avoide to be resolved and instead remapped between t=
he
=3D> avoided

2080       requiresthat SSRC/CSRC are never exposed to the WebRTC javascrip=
t
=3D> space

2185       arguments and conisderations as discussed in Appendix A.3.1.1 ap=
plies
=3D> considerations

2194       mixer in another RTP session recieves media from that end-point.
=3D> receives

2257       sequence number will need to be consequitvely incremented based =
on
=3D> consequ... ?

2263       handled indepdentently also thus working around any SSRC collisi=
ons
=3D> independently

2265       related WebRTC MediaStream signalling must be correspondlingly
=3D> correspondingly

2272       requests comming from an end-point and decide if it can act on i=
t
=3D> coming

2327       end-point.  For example receving a 2.5 mbps video stream and the=
n
=3D> receiving

2328       send out a 250 kbps video stream after transcoding is a vaste of
=3D> waste

2332       increasing media bit-rate futher than what is needed to represen=
t the
=3D> further

2368       in the two different PeerConnections that are represtented by ha=
ving
=3D> represented

2372       likely needed to be soruced by the translator in response to act=
ions
=3D> sourced

2397       keymanagement.  On RTP level the main functions that may be miss=
ing
=3D> key management

2398       in a legacy implementation that otherswise support RTP are RTCP =
in
=3D> otherwise

2433       need to change is higly dependent on what functions it need to p=
roxy
=3D> highly

2451       packet content or media.  In fact one of the more likley scenari=
o is
=3D> likely

2455       encryption and integirty protection operation to resolve missmat=
ch in
=3D> integrity
=3D> mismatch

2464       and the second one which is to to provide a group communication
=3D> to to

2480       forwards all the RTP/RTCP traffic and keymanagment to the end-po=
int
=3D> key management

2551       the client A must be capable of handlilng that when determining
=3D> handling

2560       reporting in that case becomes incosistent and without explicit
=3D> inconsistent

2634       A relay approache will result in that the WebRTC end-points will=
 have
=3D> approach


2671       to be most easily accomplished by establishing mutliple
=3D> multiple



























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<font face=3D"Courier New, monospace" size=3D"2">
<div>Colin, Magnus,</div>
<div>fyi, some editorial nits,</div>
<div>regards,</div>
<div>Albrecht</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>Line numbers according</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2"><a href=3D"http://tools.=
ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-rtcweb-rtp-usage-=
04.txt"><font face=3D"Courier New, monospace" size=3D"2" color=3D"#0000FF">=
<u>http://tools.ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-r=
tcweb-rtp-usage-04.txt</u></font></a><font face=3D"Courier New, monospace" =
size=3D"2">
</font></font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>Editorial change proposals:</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>232&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RTP Media Stream:&nbsp; =
A sequence of RTP packets, and associated RTCP</div>
<ol start=3D"233" style=3D"margin-top: 0pt; margin-bottom: 0pt; margin-left=
: 36pt; ">
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; packets, using a single synchronisation =
source (SSRC) that</li></ol>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<ul style=3D"margin-top: 0pt; margin-bottom: 0pt; margin-left: 36pt; ">
<li>synchronization (according RFC 3550)</li><li>&#8230; multiple times in =
doc</li></ul>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>295&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; partic=
ipants in the session; support for randomised RTCP</div>
<div>=3D&gt; randomized</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>407&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; session also effects the=
 possibility for differentiated treatment of</div>
<div>=3D&gt; affects</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>528&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; implemented using a cent=
ralised server, multi-unicast, or using IP</div>
<div>=3D&gt; centralized</div>
<div>&#8230; multiple times</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>588&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Request above as there t=
here may be multiple methods to fulfill the</div>
<div>=3D&gt; delete &#8216;there&#8217;</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>694&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; optimisations of non cri=
tical functions, and is hence OPTIONAL to</div>
<div>=3D&gt; optimizations</div>
<div>&nbsp;</div>
<div>715&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; will support negative ac=
knowlegdements (NACKs) for RTP data packets</div>
<div>=3D&gt; acknowledgements </div>
<div>&nbsp;</div>
<div>1247&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; own traffic.&nbsp; While is i=
s clearly considered a bug, it is important<br>

=3D&gt; is is</div>
<div>1248&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; that the end-point is able to=
 recognise and handle the case when it</div>
<div>=3D&gt; recognize</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1670&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; This in an attempt to highlig=
ht the differencies and the in many case</div>
<div>=3D&gt; differences</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1718&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; An RTP session have good supp=
ort for simultanously transport multiple<br>

=3D&gt; simultaneously </div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1723&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; video cameras can potentially=
 transmitt video from both to its</div>
<div>=3D&gt; transmit</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1730&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thus we can introduce a coupl=
e of different notiations in the below</div>
<div>=3D&gt; notations</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1731&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; two alternate figures of a si=
ngle peer connection in a a point to</div>
<div>=3D&gt; a a</div>
<div>&nbsp;</div>
<div>1838&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; different media bit-rates to =
the differnt peers, thus not forcing B</div>
<div>=3D&gt; different</div>
<div>&nbsp;</div>
<div>1839&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; to endure the same quality re=
ductions if there are limiations in the</div>
<div>=3D&gt; limitations<br>

</div>
<div>1874&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; encoder instances each beeing=
 associated with two different</div>
<div>=3D&gt; being</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1883&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; resource costrained will use =
a different implementation strategy</div>
<div>=3D&gt; constrained</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1918&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; limited to congestion control=
, also prefered resolution to receive</div>
<div>=3D&gt; preferred</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1919&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; based on dispaly area availab=
le is another aspect requiring</div>
<div>=3D&gt; display</div>
<div>&nbsp;</div>
<div>1920&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; consideration.&nbsp; The need=
 for this type of descion logic does arise in</div>
<div>=3D&gt; decision</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1949&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; its operation and form new RT=
P packets, encrypts and integegrity</div>
<div>=3D&gt; integrity</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1970&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in a media domain mix of the =
incomming RTP media streams.&nbsp; Then</div>
<div>=3D&gt; incoming</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>1980&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; produce a Mix of all N stream=
s for the group that are curently not</div>
<div>=3D&gt; currently</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2046&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; particpants.&nbsp; As each pe=
er receives a different version produced by</div>
<div>=3D&gt; participants </div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2070&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; There exist an possible imple=
mentation choice to have the RTP</div>
<div>=3D&gt; &#8220;a possible&#8221; ?</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2074&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; about the contributing source=
s.&nbsp; This removes both the functionality<br>

=3D&gt; functionality</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2078&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; they can be avoide to be reso=
lved and instead remapped between the<br>

=3D&gt; avoided </div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2080&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; requiresthat SSRC/CSRC are ne=
ver exposed to the WebRTC javascript</div>
<div>=3D&gt; space</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2185&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; arguments and conisderations =
as discussed in Appendix A.3.1.1 applies</div>
<div>=3D&gt; considerations </div>
<div>&nbsp;</div>
<div>2194&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mixer in another RTP session =
recieves media from that end-point.</div>
<div>=3D&gt; receives</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2257&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sequence number will need to =
be consequitvely incremented based on</div>
<div>=3D&gt; consequ&#8230; ?</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2263&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; handled indepdentently also t=
hus working around any SSRC collisions<br>

=3D&gt; independently </div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2265&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; related WebRTC MediaStream si=
gnalling must be correspondlingly<br>

=3D&gt; correspondingly </div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2272&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; requests comming from an end-=
point and decide if it can act on it<br>

=3D&gt; coming</div>
<div>&nbsp;</div>
<div>2327&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; end-point.&nbsp; For example =
receving a 2.5 mbps video stream and then</div>
<div>=3D&gt; receiving </div>
<div>&nbsp;</div>
<div>2328&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; send out a 250 kbps video str=
eam after transcoding is a vaste of</div>
<div>=3D&gt; waste</div>
<div>&nbsp;</div>
<div>2332&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; increasing media bit-rate fut=
her than what is needed to represent the<br>

=3D&gt; further</div>
<div>&nbsp;</div>
<div>2368&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in the two different PeerConn=
ections that are represtented by having</div>
<div>=3D&gt; represented </div>
<div>&nbsp;</div>
<div>2372&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; likely needed to be soruced b=
y the translator in response to actions</div>
<div>=3D&gt; sourced</div>
<div>&nbsp;</div>
<div>2397&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; keymanagement.&nbsp; On RTP l=
evel the main functions that may be missing</div>
<div>=3D&gt; key management</div>
<div>&nbsp;</div>
<div>2398&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in a legacy implementation th=
at otherswise support RTP are RTCP in</div>
<div>=3D&gt; otherwise </div>
<div>&nbsp;</div>
<div>2433&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; need to change is higly depen=
dent on what functions it need to proxy</div>
<div>=3D&gt; highly <br>

</div>
<div>2451&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; packet content or media.&nbsp=
; In fact one of the more likley scenario is</div>
<div>=3D&gt; likely </div>
<div>&nbsp;</div>
<div>2455&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; encryption and integirty prot=
ection operation to resolve missmatch in</div>
<div>=3D&gt; integrity </div>
<div>=3D&gt; mismatch </div>
<div>&nbsp;</div>
<div>2464&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and the second one which is t=
o to provide a group communication</div>
<div>=3D&gt; to to<br>

</div>
<div>2480&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; forwards all the RTP/RTCP tra=
ffic and keymanagment to the end-point</div>
<div>=3D&gt; key management</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2551&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the client A must be capable =
of handlilng that when determining</div>
<div>=3D&gt; handling</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div>2560&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; reporting in that case become=
s incosistent and without explicit<br>

=3D&gt; inconsistent</div>
<div>&nbsp;</div>
<div>2634&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A relay approache will result=
 in that the WebRTC end-points will have</div>
<div>=3D&gt; approach</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>2671&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; to be most easily accomplishe=
d by establishing mutliple</div>
<div>=3D&gt; multiple</div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
<div><font face=3D"Calibri, sans-serif" size=3D"2">&nbsp;</font></div>
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
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Thread-Topic: Qualcomm's Position on Mandatory-to-Implement Audio Codecs for RTCWeb
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Subject: [rtcweb] Qualcomm's Position on Mandatory-to-Implement Audio Codecs for RTCWeb
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--_000_CAC8DBE4E9704C41BCB290C2F3CC921A162A76ABnasanexd01hnaqu_
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Hello All,
Qualcomm is in favor of G.711 being the only mandatory-to-implement codec f=
or RTCWeb.  We do not support having both Opus and G.711 as mandatory-to-im=
plement codecs.  We state the following reasons:


a)      There is no technical need to have an audio codec besides G.711 be =
designated as mandatory to implement.  Because RTCWeb uses SDP to negotiate=
 the codecs, the only reason to have anything as mandatory is to make sure =
there is always at least one codec that both sides can use, a lowest common=
 denominator.  This allows the market to decide which codecs to support, ra=
ther than edict from a standards body.  Implementations can base their choi=
ce of which codec to support based on what they are trying to optimize.  Fo=
r example, better performance for the environment at the time of the sessio=
n, such as cell phone or desktop.

b)      G.711 is universal, unencumbered, and widely implemented.    In add=
ition, note that codec implementation and testing is quite costly for chip =
and device manufacturers, and that cost is ultimately reflected in the cost=
 of the end-user device.  The computational simplicity of G.711 and its lon=
g-time availability on a broad range of platforms means that the implementa=
tion is available on low/entry-tier devices and testing costs have already =
been amortized over many years, thereby enabling its use in low cost end-us=
er devices.  If we want to see widespread use of RTCWeb for voice, than we =
will reach a much broader population if the minimum requirements do not pro=
hibit low-cost devices.

c)       A mandate for Opus will limit initial RTCWeb clients to use softwa=
re-based  codecs (on general purpose processors) where Opus can be implemen=
ted and tested easily until it is available on a variety of DSPs.  Even the=
n, it will likely start on high-cost platforms.  This may in turn mean that=
 RTCWeb clients could consume significantly more battery power than DSP-bas=
ed codecs used in many traditional (circuit-switched) voice services, furth=
er inhibiting the end-user from choosing RTCWeb over those services.

d)      We do not believe Opus is more versatile than other standardized co=
decs, because any measure of versatility should take into account quality a=
t a given bitrate.  If Opus is not able to deliver superior quality at all =
supported bitrates when compared to other codecs, then it cannot be deemed =
as being more versatile simply because it supports more bitrates when compa=
red to other standardized codecs.



In conclusion - the IETF has a strong tradition of starting work with the l=
east amount of complexity and specification possible, gaining operational e=
xperience, and then refining things with revisions or extensions.  So, the =
least amount of complexity would be G.711 as the sole audio codec.

--_000_CAC8DBE4E9704C41BCB290C2F3CC921A162A76ABnasanexd01hnaqu_
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<p class=3D"MsoNormal">Hello All,<o:p></o:p></p>
<p class=3D"MsoNormal">Qualcomm is in favor of G.711 being the only mandato=
ry-to-implement codec for RTCWeb.&nbsp; We do not support having both Opus =
and G.711 as mandatory-to-implement codecs.&nbsp; We state the following re=
asons:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText" style=3D"margin-left:.5in;text-indent:-.25in;mso-=
list:l0 level1 lfo2">
<![if !supportLists]><span style=3D"mso-list:Ignore">a)<span style=3D"font:=
7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>There is no technical need to have an audio codec b=
esides G.711 be designated as mandatory to implement.&nbsp; Because RTCWeb =
uses SDP to negotiate the codecs, the only reason to have anything as manda=
tory is to make sure there is always
 at least one codec that both sides can use, a lowest common denominator.&n=
bsp; This allows the market to decide which codecs to support, rather than =
edict from a standards body.&nbsp; Implementations can base their choice of=
 which codec to support based on what they
 are trying to optimize.&nbsp; For example, better performance for the envi=
ronment at the time of the session, such as cell phone or desktop.<o:p></o:=
p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">b)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>G.711 is universal, unencumbered, and widely implem=
ented.&nbsp;&nbsp;&nbsp; In addition, note that codec implementation and te=
sting is quite costly for chip and device manufacturers, and that cost is u=
ltimately reflected in the cost of the end-user
 device.&nbsp; The computational simplicity of G.711 and its long-time avai=
lability on a broad range of platforms means that the implementation is ava=
ilable on low/entry-tier devices and testing costs have already been amorti=
zed over many years, thereby enabling
 its use in low cost end-user devices.&nbsp; If we want to see widespread u=
se of RTCWeb for voice, than we will reach a much broader population if the=
 minimum requirements do not prohibit low-cost devices.&nbsp;
<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">c)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
</span></span><![endif]>A mandate for Opus will limit initial RTCWeb client=
s to use software-based&nbsp; codecs (on general purpose processors) where =
Opus can be implemented and tested easily until it is available on a variet=
y of DSPs.&nbsp; Even then, it will likely
 start on high-cost platforms.&nbsp; This may in turn mean that RTCWeb clie=
nts could consume significantly more battery power than DSP-based codecs us=
ed in many traditional (circuit-switched) voice services, further inhibitin=
g the end-user from choosing RTCWeb over
 those services.<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">d)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>We do not believe Opus is more versatile than other=
 standardized codecs, because any measure of versatility should take into a=
ccount quality at a given bitrate.&nbsp; If Opus is not able to deliver sup=
erior quality at all supported bitrates
 when compared to other codecs, then it cannot be deemed as being more vers=
atile simply because it supports more bitrates when compared to other stand=
ardized codecs.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">In conclusion - the IETF has a strong tradition o=
f starting work with the least amount of complexity and specification possi=
ble, gaining operational experience, and then refining things with revision=
s or extensions.&nbsp; So, the least amount
 of complexity would be G.711 as the sole audio codec.<o:p></o:p></p>
</div>
</body>
</html>

--_000_CAC8DBE4E9704C41BCB290C2F3CC921A162A76ABnasanexd01hnaqu_--

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On 15 Aug 2012, at 14:48, Mandyam, Giridhar wrote:

> Hello All,
> Qualcomm is in favor of G.711 being the only mandatory-to-implement =
codec for RTCWeb.  We do not support having both Opus and G.711 as =
mandatory-to-implement codecs.  We state the following reasons:
> =20
> a)      There is no technical need to have an audio codec besides =
G.711 be designated as mandatory to implement.  Because RTCWeb uses SDP =
to negotiate the codecs, the only reason to have anything as mandatory =
is to make sure there is always at least one codec that both sides can =
use, a lowest common denominator.  This allows the market to decide =
which codecs to support, rather than edict from a standards body.  =
Implementations can base their choice of which codec to support based on =
what they are trying to optimize.  For example, better performance for =
the environment at the time of the session, such as cell phone or =
desktop.

I made exactly the same argument about H261 and was told (correctly) =
that it was unacceptable to mandate a codec that would give users a
poorer than necessary experience. I think what applies for video applies =
for audio.

Specifically g711 is a poor codec choice in a wide variety of network =
situations - e.g. 3g and edge of wifi. In my experience a lower =
bandwidth
codec which supports wideband, dynamically variable bitrates and error =
correction / packet loss concealment is required to give these users =
audio
that is acceptable to use in the  coffee-shop / airport scenario.=20

> b)      G.711 is universal, unencumbered, and widely implemented.    =
In addition, note that codec implementation and testing is quite costly =
for chip and device manufacturers, and that cost is ultimately reflected =
in the cost of the end-user device.  The computational simplicity of =
G.711 and its long-time availability on a broad range of platforms means =
that the implementation is available on low/entry-tier devices and =
testing costs have already been amortized over many years, thereby =
enabling its use in low cost end-user devices.  If we want to see =
widespread use of RTCWeb for voice, than we will reach a much broader =
population if the minimum requirements do not prohibit low-cost devices.=20=


There are many codecs which that can be said for. We happily run speex =
on low end smartphones. - Heck we even use the java implementation in
applets and get acceptable performance. g722 has recently dropped out of =
patent, uses the same bandwidth as 711 and gives significantly better =
audio.
Any device capable of running SRTP is capable of running either 722 or =
speex in my experience.=20


> c)       A mandate for Opus will limit initial RTCWeb clients to use =
software-based  codecs (on general purpose processors) where Opus can be =
implemented and tested easily until it is available on a variety of =
DSPs.  Even then, it will likely start on high-cost platforms.  This may =
in turn mean that RTCWeb clients could consume significantly more =
battery power than DSP-based codecs used in many traditional =
(circuit-switched) voice services, further inhibiting the end-user from =
choosing RTCWeb over those services.

You are confusing mandatory-to-implement and mandatory-to-use. Just =
because Opus must be available , you don't have to use it. The devices =
you=20
mention almost all have AMR in DSP - but unfortunately that is =
encumbered so isn't ideal as a mandatory-to-implememnt codec but may =
well be=20
used when it is available.=20

> d)      We do not believe Opus is more versatile than other =
standardized codecs, because any measure of versatility should take into =
account quality at a given bitrate.  If Opus is not able to deliver =
superior quality at all supported bitrates when compared to other =
codecs, then it cannot be deemed as being more versatile simply because =
it supports more bitrates when compared to other standardized codecs.


Again - making Opus mandatory to implement does not preclude supporting =
or using any other codecs.=20

By the way point d) applies to any codec you choose to name - so you are =
probably arguing for no mandatory-to-implement codec - which would be
a mistake I think. It certainly applies to g711 - in spades!

> =20
> In conclusion - the IETF has a strong tradition of starting work with =
the least amount of complexity and specification possible, gaining =
operational experience, and then refining things with revisions or =
extensions.  So, the least amount of complexity would be G.711 as the =
sole audio codec.

RTCweb is never going to be a _least_possible_complexity_ project. The =
requirements mean we need  SRTP (DTLS?), a video modern codec, Echo =
cancellation, AGC, etc. Adding a good audio codec will not significantly =
extend the complexity.



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


--Apple-Mail=_D7AAB7AC-70E0-41A8-B4C2-3781AA1F0459
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head><base href=3D"x-msg://378/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><br><div><div>On 15 Aug 2012, at 14:48, Mandyam, =
Giridhar wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; ">Hello =
All,<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif; ">Qualcomm is in favor of G.711 being the only =
mandatory-to-implement codec for RTCWeb.&nbsp; We do not support having =
both Opus and G.711 as mandatory-to-implement codecs.&nbsp; We state the =
following reasons:<o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0.5in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; text-indent: -0.25in; "><span>a)<span style=3D"font: normal =
normal normal 7pt/normal 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>There is no =
technical need to have an audio codec besides G.711 be designated as =
mandatory to implement.&nbsp; Because RTCWeb uses SDP to negotiate the =
codecs, the only reason to have anything as mandatory is to make sure =
there is always at least one codec that both sides can use, a lowest =
common denominator.&nbsp; This allows the market to decide which codecs =
to support, rather than edict from a standards body.&nbsp; =
Implementations can base their choice of which codec to support based on =
what they are trying to optimize.&nbsp; For example, better performance =
for the environment at the time of the session, such as cell phone or =
desktop.</div></div></div></span></blockquote><div><br></div><div>I made =
exactly the same argument about H261 and was told (correctly) that it =
was unacceptable to mandate a codec that would give users =
a</div><div>poorer than necessary experience. I think what applies for =
video applies for audio.</div><div><br></div><div>Specifically g711 is a =
poor codec choice in a wide variety of network situations - e.g. 3g and =
edge of wifi. In my experience a lower bandwidth</div><div>codec which =
supports&nbsp;wideband, dynamically variable bitrates and error =
correction / packet loss concealment is required to give these users =
audio</div><div>that is acceptable to use in the &nbsp;coffee-shop / =
airport scenario.&nbsp;</div><br><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0.5in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -0.25in; =
"><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0.5in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; text-indent: -0.25in; "><span>b)<span =
style=3D"font: normal normal normal 7pt/normal 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>G.711 is =
universal, unencumbered, and widely implemented.&nbsp;&nbsp;&nbsp; In =
addition, note that codec implementation and testing is quite costly for =
chip and device manufacturers, and that cost is ultimately reflected in =
the cost of the end-user device.&nbsp; The computational simplicity of =
G.711 and its long-time availability on a broad range of platforms means =
that the implementation is available on low/entry-tier devices and =
testing costs have already been amortized over many years, thereby =
enabling its use in low cost end-user devices.&nbsp; If we want to see =
widespread use of RTCWeb for voice, than we will reach a much broader =
population if the minimum requirements do not prohibit low-cost =
devices.&nbsp;</div></div></div></span></blockquote><div><br></div><div>Th=
ere are many codecs which that can be said for. We happily run speex on =
low end smartphones. - Heck we even use the java implementation =
in</div><div>applets and get acceptable performance. g722 has recently =
dropped out of patent, uses the same bandwidth as 711 and gives =
significantly better audio.</div><div>Any device capable of running SRTP =
is capable of running either 722 or speex in my =
experience.&nbsp;</div><div><br></div><br><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0.5in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -0.25in; =
"><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0.5in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; text-indent: -0.25in; "><span>c)<span =
style=3D"font: normal normal normal 7pt/normal 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>A mandate for =
Opus will limit initial RTCWeb clients to use software-based&nbsp; =
codecs (on general purpose processors) where Opus can be implemented and =
tested easily until it is available on a variety of DSPs.&nbsp; Even =
then, it will likely start on high-cost platforms.&nbsp; This may in =
turn mean that RTCWeb clients could consume significantly more battery =
power than DSP-based codecs used in many traditional (circuit-switched) =
voice services, further inhibiting the end-user from choosing RTCWeb =
over those =
services.</div></div></div></span></blockquote><div><br></div><div>You =
are confusing mandatory-to-implement and mandatory-to-use. Just because =
Opus must be available , you don't have to use it. The devices =
you&nbsp;</div><div>mention almost all have AMR in DSP - but =
unfortunately that is encumbered so isn't ideal as a =
mandatory-to-implememnt codec but may well be&nbsp;</div><div>used when =
it is available.&nbsp;</div><br><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0.5in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -0.25in; =
"><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0.5in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; text-indent: -0.25in; "><span>d)<span =
style=3D"font: normal normal normal 7pt/normal 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>We do not =
believe Opus is more versatile than other standardized codecs, because =
any measure of versatility should take into account quality at a given =
bitrate.&nbsp; If Opus is not able to deliver superior quality at all =
supported bitrates when compared to other codecs, then it cannot be =
deemed as being more versatile simply because it supports more bitrates =
when compared to other standardized =
codecs.</div></div></div></span></blockquote><div><br></div><div><br></div=
><div>Again - making Opus mandatory to implement does not preclude =
supporting or using any other codecs.&nbsp;</div><div><br></div><div>By =
the way point d) applies to any codec you choose to name - so you are =
probably arguing for no mandatory-to-implement codec - which would =
be</div><div>a mistake I think.&nbsp;It certainly applies to g711 - in =
spades!</div><br><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0.5in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -0.25in; =
"><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; ">In conclusion - the =
IETF has a strong tradition of starting work with the least amount of =
complexity and specification possible, gaining operational experience, =
and then refining things with revisions or extensions.&nbsp; So, the =
least amount of complexity would be G.711 as the sole audio =
codec.</div></div></div></span></blockquote><div><br></div><div>RTCweb =
is never going to be a _least_possible_complexity_ project. The =
requirements mean we need &nbsp;SRTP (DTLS?), a video modern codec, Echo =
cancellation, AGC, etc. Adding a good audio codec will not significantly =
extend the =
complexity.</div><div><br></div><div><br></div><br><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-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: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; =
"><o:p></o:p></div></div>_______________________________________________<b=
r>rtcweb mailing list<br><a href=3D"mailto:rtcweb@ietf.org" =
style=3D"color: blue; text-decoration: underline; =
">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/rtcweb</a><br></div></span></block=
quote></div><br></body></html>=

--Apple-Mail=_D7AAB7AC-70E0-41A8-B4C2-3781AA1F0459--

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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] Confirmation of consensus to adopt draft-jennings-rtcweb-qos
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During the recent meeting, the sense of the room was that the working
group should adopt draft-jennings-rtcweb-qos as a work item.  If there
are new opinions, data, or proposals please send them to the list by
August 30th, after which the chairs will make a determination of
consensus.

regards,

Ted

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Subject: [rtcweb] Video codec proposals due October 15th, 2012
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As agreed at the recent meeting, the chairs solicit internet-drafts
naming proposed mandatory-to-implement video codecs or codec sets by
October 15th, 2012.  Please include any technical data you believe
supports the choice you are proposing.  A non-exhaustive list of data
you may wish to include is found in slides 3-6 of
http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-4.pptx .

Discussion based on these will commence immediately upon receipt,
rather than waiting until the October 15th deadline, so early
submission is encouraged.

regards,

Ted Hardie

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Confirmation of consensus on audio codecs
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At the last meeting we took a hum on selecting Opus and G.711 as the mediat=
ory to implement audio codecs. If there is any new opinions please send the=
m to the list by August 30th, after which the chairs will make a determinat=
ion of consensus.

Thanks,
Cullen

Please note that the following IPR disclosure have been made on these codec=
s. They can be found at=20

http://datatracker.ietf.org/ipr/


2010-11-07=09
=95 ID # 1445
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (1)"
2010-11-07=09
=95 ID # 1446
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-00"
2010-11-12=09
=95 ID # 1447
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (2)"
2011-03-23=09
=95 ID # 1520
"Qualcomm Incorporated's Statement about IPR related to draft-ietf-codec-op=
us-05"
2011-03-27=09
=95 ID # 1524
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-05"
2011-03-29=09
=95 ID # 1526
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-05"
2011-03-29=09
=95 ID # 1525
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
2011-07-23=09
=95 ID # 1602
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
2012-01-25=09
=95 ID # 1670
"Microsoft Corporation's Statement about IPR related to draft-ietf-codec-op=
us-10"
2012-03-12=09
=95 ID # 1712
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (1)"
2012-04-02=09
=95 ID # 1741
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (2)"




From roman@telurix.com  Thu Aug 16 10:35:14 2012
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Date: Thu, 16 Aug 2012 13:35:01 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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--047d7bdc992c931c6e04c76576cf
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As I have mentioned before I would like to see G.722 as a mandatory to
implement codec in addition to G.711 and OPUS. G.722 is license free and is
widely supported. It operates at the same bandwidths as G.711 with
virtually the same CPU requirements, but offers HD audio
quality superior to G.711.

One question I wanted to ask about G.711, do we require any of the
appendixes to be implemented, ie would PLC and DTX be required? I think
that PLC should be at least recommended, since it will significantly
improve resulting communication experience in most of the real deployment
scenarios.
_____________
Roman Shpount


On Thu, Aug 16, 2012 at 1:15 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> At the last meeting we took a hum on selecting Opus and G.711 as the
> mediatory to implement audio codecs. If there is any new opinions please
> send them to the list by August 30th, after which the chairs will make a
> determination of consensus.
>
> Thanks,
> Cullen
>
> Please note that the following IPR disclosure have been made on these
> codecs. They can be found at
>
> http://datatracker.ietf.org/ipr/
>
>
> 2010-11-07
> =95 ID # 1445
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> 2010-11-07
> =95 ID # 1446
> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-00"
> 2010-11-12
> =95 ID # 1447
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
> 2011-03-23
> =95 ID # 1520
> "Qualcomm Incorporated's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-27
> =95 ID # 1524
> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-29
> =95 ID # 1526
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-29
> =95 ID # 1525
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-07-23
> =95 ID # 1602
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
> 2012-01-25
> =95 ID # 1670
> "Microsoft Corporation's Statement about IPR related to
> draft-ietf-codec-opus-10"
> 2012-03-12
> =95 ID # 1712
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (1)"
> 2012-04-02
> =95 ID # 1741
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (2)"
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7bdc992c931c6e04c76576cf
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

As I have mentioned before I would like to see G.722 as a mandatory to impl=
ement codec in addition to G.711 and OPUS. G.722 is license free=A0and is w=
idely supported. It operates at the same bandwidths as G.711 with virtually=
 the same CPU requirements, but offers HD audio quality=A0superior=A0to G.7=
11.<div>
<br></div><div>One question I wanted to ask about G.711, do we require any =
of the appendixes to be implemented, ie would PLC and DTX be required? I th=
ink that PLC should be at least recommended, since it will=A0significantly =
improve resulting communication experience in most of the real deployment s=
cenarios.=A0</div>
<div>_____________</div><div><div>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Thu, Aug 16, 2012 at 1:15 PM, Cullen =
Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com"=
 target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex">
<br>
At the last meeting we took a hum on selecting Opus and G.711 as the mediat=
ory to implement audio codecs. If there is any new opinions please send the=
m to the list by August 30th, after which the chairs will make a determinat=
ion of consensus.<br>

<br>
Thanks,<br>
Cullen<br>
<br>
Please note that the following IPR disclosure have been made on these codec=
s. They can be found at<br>
<br>
<a href=3D"http://datatracker.ietf.org/ipr/" target=3D"_blank">http://datat=
racker.ietf.org/ipr/</a><br>
<br>
<br>
2010-11-07<br>
=95 ID # 1445<br>
&quot;Broadcom Corporation&#39;s Statement about IPR related to draft-ietf-=
codec-opus-00 and draft-ietf-codec-description-00 (1)&quot;<br>
2010-11-07<br>
=95 ID # 1446<br>
&quot;Xiph.Org Foundation&#39;s Statement about IPR related to draft-ietf-c=
odec-opus-00&quot;<br>
2010-11-12<br>
=95 ID # 1447<br>
&quot;Broadcom Corporation&#39;s Statement about IPR related to draft-ietf-=
codec-opus-00 and draft-ietf-codec-description-00 (2)&quot;<br>
2011-03-23<br>
=95 ID # 1520<br>
&quot;Qualcomm Incorporated&#39;s Statement about IPR related to draft-ietf=
-codec-opus-05&quot;<br>
2011-03-27<br>
=95 ID # 1524<br>
&quot;Xiph.Org Foundation&#39;s Statement about IPR related to draft-ietf-c=
odec-opus-05&quot;<br>
2011-03-29<br>
=95 ID # 1526<br>
&quot;Broadcom Corporation&#39;s Statement about IPR related to draft-ietf-=
codec-opus-05&quot;<br>
2011-03-29<br>
=95 ID # 1525<br>
&quot;Skype Limited&#39;s Statement about IPR related to draft-ietf-codec-o=
pus-05&quot;<br>
2011-07-23<br>
=95 ID # 1602<br>
&quot;Skype Limited&#39;s Statement about IPR related to draft-ietf-codec-o=
pus-07&quot;<br>
2012-01-25<br>
=95 ID # 1670<br>
&quot;Microsoft Corporation&#39;s Statement about IPR related to draft-ietf=
-codec-opus-10&quot;<br>
2012-03-12<br>
=95 ID # 1712<br>
&quot;Huawei Technologies Co.,Ltd&#39;s Statement about IPR related to draf=
t-ietf-codec-opus-11 (1)&quot;<br>
2012-04-02<br>
=95 ID # 1741<br>
&quot;Huawei Technologies Co.,Ltd&#39;s Statement about IPR related to draf=
t-ietf-codec-opus-11 (2)&quot;<br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--047d7bdc992c931c6e04c76576cf--

From richard@shockey.us  Thu Aug 16 16:24:13 2012
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From: "Richard Shockey" <richard@shockey.us>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Reading this list is occasionally an act of torture banned by several
conventions ..but since you ask.

I completely support the selection of Opus and G.711 as mandatory to
implement audio codec's ..however I'm very very open minded about supporting
G.722. It has it merits.  It should be totally obvious to most that if you
even think about interconnecting to public E.164 networks the default option
for VoLTE and Enterprise Voice networks is going to be G.722. 

If it is your goal to create globally useful stove pipes fine,  but
interconnection with existing carrier real time networks is IMHO a rational
goal. 

As for Video .. for goodness sakes just get over it people. H.264 is totally
implemented everywhere on the planet Earth. So what about the intellectual
property problems. It's not like VP8 doesn't have problems either. 

I'll save my comments about the SDP offer/answer issue for another day. 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Cullen Jennings (fluffy)
Sent: Thursday, August 16, 2012 1:16 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Confirmation of consensus on audio codecs


At the last meeting we took a hum on selecting Opus and G.711 as the
mediatory to implement audio codecs. If there is any new opinions please
send them to the list by August 30th, after which the chairs will make a
determination of consensus.

Thanks,
Cullen

Please note that the following IPR disclosure have been made on these
codecs. They can be found at 

http://datatracker.ietf.org/ipr/


2010-11-07	
. ID # 1445
"Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
2010-11-07	
. ID # 1446
"Xiph.Org Foundation's Statement about IPR related to
draft-ietf-codec-opus-00"
2010-11-12	
. ID # 1447
"Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
2011-03-23	
. ID # 1520
"Qualcomm Incorporated's Statement about IPR related to
draft-ietf-codec-opus-05"
2011-03-27	
. ID # 1524
"Xiph.Org Foundation's Statement about IPR related to
draft-ietf-codec-opus-05"
2011-03-29	
. ID # 1526
"Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-05"
2011-03-29	
. ID # 1525
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
2011-07-23	
. ID # 1602
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
2012-01-25	
. ID # 1670
"Microsoft Corporation's Statement about IPR related to
draft-ietf-codec-opus-10"
2012-03-12	
. ID # 1712
"Huawei Technologies Co.,Ltd's Statement about IPR related to
draft-ietf-codec-opus-11 (1)"
2012-04-02	
. ID # 1741
"Huawei Technologies Co.,Ltd's Statement about IPR related to
draft-ietf-codec-opus-11 (2)"



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


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From: Monty Montgomery <xiphmont@gmail.com>
To: Richard Shockey <richard@shockey.us>
Content-Type: text/plain; charset=ISO-8859-1
Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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> It's not like VP8 doesn't have problems either.

Actually, no, it doesn't have problems.

Monty
Xiph.Org

From bernard_aboba@hotmail.com  Thu Aug 16 17:41:58 2012
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I also support the selection of OPUS and G.711 as MTI. G.722 can be a SHOULD=
; it is widely supported.



On Aug 16, 2012, at 4:24 PM, "Richard Shockey" <richard@shockey.us> wrote:

> Reading this list is occasionally an act of torture banned by several
> conventions ..but since you ask.
>=20
> I completely support the selection of Opus and G.711 as mandatory to
> implement audio codec's ..however I'm very very open minded about supporti=
ng
> G.722. It has it merits.  It should be totally obvious to most that if you=

> even think about interconnecting to public E.164 networks the default opti=
on
> for VoLTE and Enterprise Voice networks is going to be G.722.=20
>=20
> If it is your goal to create globally useful stove pipes fine,  but
> interconnection with existing carrier real time networks is IMHO a rationa=
l
> goal.=20
>=20
> As for Video .. for goodness sakes just get over it people. H.264 is total=
ly
> implemented everywhere on the planet Earth. So what about the intellectual=

> property problems. It's not like VP8 doesn't have problems either.=20
>=20
> I'll save my comments about the SDP offer/answer issue for another day.=20=

>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f
> Cullen Jennings (fluffy)
> Sent: Thursday, August 16, 2012 1:16 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Confirmation of consensus on audio codecs
>=20
>=20
> At the last meeting we took a hum on selecting Opus and G.711 as the
> mediatory to implement audio codecs. If there is any new opinions please
> send them to the list by August 30th, after which the chairs will make a
> determination of consensus.
>=20
> Thanks,
> Cullen
>=20
> Please note that the following IPR disclosure have been made on these
> codecs. They can be found at=20
>=20
> http://datatracker.ietf.org/ipr/
>=20
>=20
> 2010-11-07   =20
> . ID # 1445
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> 2010-11-07   =20
> . ID # 1446
> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-00"
> 2010-11-12   =20
> . ID # 1447
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
> 2011-03-23   =20
> . ID # 1520
> "Qualcomm Incorporated's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-27   =20
> . ID # 1524
> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-29   =20
> . ID # 1526
> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-05"
> 2011-03-29   =20
> . ID # 1525
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-07-23   =20
> . ID # 1602
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
> 2012-01-25   =20
> . ID # 1670
> "Microsoft Corporation's Statement about IPR related to
> draft-ietf-codec-opus-10"
> 2012-03-12   =20
> . ID # 1712
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (1)"
> 2012-04-02   =20
> . ID # 1741
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (2)"
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Richard Shockey" <richard@shockey.us>
To: "'Monty Montgomery'" <xiphmont@gmail.com>, <rtcweb@ietf.org>
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Date: Thu, 16 Aug 2012 21:38:32 -0400
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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SOOO WRONG .. just do a simple search.  It just hasn't been fully litigated
...yet. 

-----Original Message-----
From: Monty Montgomery [mailto:xiphmont@gmail.com] 
Sent: Thursday, August 16, 2012 8:11 PM
To: Richard Shockey
Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

> It's not like VP8 doesn't have problems either.

Actually, no, it doesn't have problems.

Monty
Xiph.Org


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To: "'Bernard Aboba'" <bernard_aboba@hotmail.com>, <rtcweb@ietf.org>
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Cc: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Bernard Aboba
Sent: Thursday, August 16, 2012 8:42 PM
To: rtcweb@ietf.org
Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

I also support the selection of OPUS and G.711 as MTI. G.722 can be a
SHOULD; it is widely supported.

[RS> ] +1 


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From: Monty Montgomery <xiphmont@gmail.com>
To: Richard Shockey <richard@shockey.us>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On Thu, Aug 16, 2012 at 9:38 PM, Richard Shockey <richard@shockey.us> wrote:
> SOOO WRONG .. just do a simple search.  It just hasn't been fully litigated
> ...yet.

It hasn't been litigated at all. Or even credibly threatened beyond
the same substanceless mouthing off in to press we saw with Vorbis,
Theora, etc...

Monty
Xiph.Org

From lorenzo@meetecho.com  Fri Aug 17 00:52:54 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Bernard Aboba <bernard_aboba@hotmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Confirmation of consensus on audio codecs
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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I agree. OPUS because it will be the IETF standard for wideband audio on th=
e Internet, G.711 for maximum interoperability. G.722 is a reasonable thing=
 to put on the "SHOULD" list... there's some other good choices for the SHO=
ULD list, but they unfortunately come with additional IPR issues.

Matthew Kaufman

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: Thursday, August 16, 2012 5:42 PM
To: rtcweb@ietf.org
Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

I also support the selection of OPUS and G.711 as MTI. G.722 can be a SHOUL=
D; it is widely supported.




From jdrosen@jdrosen.net  Fri Aug 17 10:02:26 2012
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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I'll chime in here too - it'll surprise no one that I agree Opus
should be MTI. I agree on G.711 as well, as this has realistically
been the lingua franca for voip since the very beginning. Its easy to
put in, no IPR and will help with the many interop scenarios which the
market will eventually want. Even if intermediaries are needed on the
media path for other things - like ICE interop - at least we won't
need transcode.

-Jonathan R.

On Fri, Aug 17, 2012 at 12:30 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
> I agree. OPUS because it will be the IETF standard for wideband audio on =
the Internet, G.711 for maximum interoperability. G.722 is a reasonable thi=
ng to put on the "SHOULD" list... there's some other good choices for the S=
HOULD list, but they unfortunately come with additional IPR issues.
>
> Matthew Kaufman
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Bernard Aboba
> Sent: Thursday, August 16, 2012 5:42 PM
> To: rtcweb@ietf.org
> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
> I also support the selection of OPUS and G.711 as MTI. G.722 can be a SHO=
ULD; it is widely supported.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Jonathan Rosenberg, Ph.D.
GM Product Strategy and Research, Microsoft/Skype
jdrosen@{skype.net,jdrosen.net,microsoft.com}
http://www.jdrosen.net

From ietf@kenfischer.net  Fri Aug 17 10:47:00 2012
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 +1 on Matthew's recommendation.  I agree that .722 should be on the
SHOULD list, but honestly I think most people who implement beyond the
MUST will add what ever they like, and what ever they have access to, and
so the ultimate impact of the SHOULDs has always been questionable to me.

Ken

On 8/17/12 10:30 AM, "Matthew Kaufman" <matthew.kaufman@skype.net> wrote:

>I agree. OPUS because it will be the IETF standard for wideband audio on
>the Internet, G.711 for maximum interoperability. G.722 is a reasonable
>thing to put on the "SHOULD" list... there's some other good choices for
>the SHOULD list, but they unfortunately come with additional IPR issues.
>
>Matthew Kaufman
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Bernard Aboba
>Sent: Thursday, August 16, 2012 5:42 PM
>To: rtcweb@ietf.org
>Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
>Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
>I also support the selection of OPUS and G.711 as MTI. G.722 can be a
>SHOULD; it is widely supported.
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb



From tim@phonefromhere.com  Fri Aug 17 12:34:50 2012
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From: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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Harald, I see from reviewing the archives that my reply to this didn't =
go in for some reason,
so I just want to put on the record that you are (of course) right. =
There was no specific proposal to
action, so there was nothing for the chairs to ignore.

The point I failed to make politely was that many of us knew informally =
that Skype/MS were not=20
happy with the situation, so a document like this should not have been =
unexpected.

My apologies for my rudeness.

Tim.

On 8 Aug 2012, at 16:57, Harald Alvestrand wrote:

> On 08/07/2012 11:57 AM, Tim Panton wrote:
>> Ok, the timing is unfortunate, but can you honestly say that we =
didn't know that this was skype/microsofts opinion? We just chose to =
ignore it because it was inconvenient.
>>=20
>> Now that it is out there, are we seriously going to ignore a document =
with _those_ authors from a major browser maker and a team with =
extensive experience in the field jus because it is late!?!?
> Speaking with WG chair hat on:
>=20
> At the time of previous "what is the appropriate level" discussion, =
the WG chairs concluded that there was a rough WG consensus to stay with =
a higher-level API rather than trying to move the level of the API =
downwards towards a "low level API".
>=20
> Since that time and until this week, there has been no specific =
proposal to be evaluated, and the main proponents of such an  API have =
been silent. Normal behaviour is to assume that a previous consensus =
declaration stands until there is new technical evidence to be =
evaluated, and proceed with further work on that basis.
>=20
> So I can't agree that "because it was inconvenient" is an appropriate =
characterization.
>=20
>              Harald
>=20
>=20


From tim@phonefromhere.com  Fri Aug 17 12:34:56 2012
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From: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Microsoft tells W3C and IETF what we are doing no signs of offering real world interoperability
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Harald, I see from reviewing the archives that my reply to this didn't =
go in for some reason,
so I just want to put on the record that you are (of course) right. =
There was no specific proposal to
action, so there was nothing for the chairs to ignore.

The point I failed to make politely was that many of us knew informally =
that Skype/MS were not=20
happy with the situation, so a document like this should not have been =
unexpected.

My apologies for my rudeness.

Tim.

On 8 Aug 2012, at 16:57, Harald Alvestrand wrote:

> On 08/07/2012 11:57 AM, Tim Panton wrote:
>> Ok, the timing is unfortunate, but can you honestly say that we =
didn't know that this was skype/microsofts opinion? We just chose to =
ignore it because it was inconvenient.
>>=20
>> Now that it is out there, are we seriously going to ignore a document =
with _those_ authors from a major browser maker and a team with =
extensive experience in the field jus because it is late!?!?
> Speaking with WG chair hat on:
>=20
> At the time of previous "what is the appropriate level" discussion, =
the WG chairs concluded that there was a rough WG consensus to stay with =
a higher-level API rather than trying to move the level of the API =
downwards towards a "low level API".
>=20
> Since that time and until this week, there has been no specific =
proposal to be evaluated, and the main proponents of such an  API have =
been silent. Normal behaviour is to assume that a previous consensus =
declaration stands until there is new technical evidence to be =
evaluated, and proceed with further work on that basis.
>=20
> So I can't agree that "because it was inconvenient" is an appropriate =
characterization.
>=20
>             Harald
>=20
>=20


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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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+1 to Opus and G.711 being mandatory for audio.

From basilgohar@librevideo.org  Fri Aug 17 13:08:43 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On 08/16/2012 09:38 PM, Richard Shockey wrote:
> SOOO WRONG .. just do a simple search.  It just hasn't been fully litigated
> ...yet.
It's impossible for something to be fully litigated.  Even someone
licensing H.264 through MPEG-LA is liable to be sued, as Motorola has
taught us, and the promises from the cartel hold no guarantees for their
future.

The condition you are placing on VP8 is impossible to meet and falsely
assumed to apply to H.264.  What is real is that H.264 has a licensing
situation that excludes implementation in a free software context in
most places of the world that will use it, and VP8 does not.

Now, this is a discussion about the audio codecs, but FUD is FUD and
needs to be nipped in the bud.

From juberti@google.com  Fri Aug 17 16:23:18 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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I agree that Opus and G.711 should be the MTI audio codecs for WebRTC.


On Thu, Aug 16, 2012 at 5:41 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> I also support the selection of OPUS and G.711 as MTI. G.722 can be a
> SHOULD; it is widely supported.
>
>
>
> On Aug 16, 2012, at 4:24 PM, "Richard Shockey" <richard@shockey.us> wrote:
>
> > Reading this list is occasionally an act of torture banned by several
> > conventions ..but since you ask.
> >
> > I completely support the selection of Opus and G.711 as mandatory to
> > implement audio codec's ..however I'm very very open minded about
> supporting
> > G.722. It has it merits.  It should be totally obvious to most that if
> you
> > even think about interconnecting to public E.164 networks the default
> option
> > for VoLTE and Enterprise Voice networks is going to be G.722.
> >
> > If it is your goal to create globally useful stove pipes fine,  but
> > interconnection with existing carrier real time networks is IMHO a
> rational
> > goal.
> >
> > As for Video .. for goodness sakes just get over it people. H.264 is
> totally
> > implemented everywhere on the planet Earth. So what about the
> intellectual
> > property problems. It's not like VP8 doesn't have problems either.
> >
> > I'll save my comments about the SDP offer/answer issue for another day.
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of
> > Cullen Jennings (fluffy)
> > Sent: Thursday, August 16, 2012 1:16 PM
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Confirmation of consensus on audio codecs
> >
> >
> > At the last meeting we took a hum on selecting Opus and G.711 as the
> > mediatory to implement audio codecs. If there is any new opinions please
> > send them to the list by August 30th, after which the chairs will make a
> > determination of consensus.
> >
> > Thanks,
> > Cullen
> >
> > Please note that the following IPR disclosure have been made on these
> > codecs. They can be found at
> >
> > http://datatracker.ietf.org/ipr/
> >
> >
> > 2010-11-07
> > . ID # 1445
> > "Broadcom Corporation's Statement about IPR related to
> > draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> > 2010-11-07
> > . ID # 1446
> > "Xiph.Org Foundation's Statement about IPR related to
> > draft-ietf-codec-opus-00"
> > 2010-11-12
> > . ID # 1447
> > "Broadcom Corporation's Statement about IPR related to
> > draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
> > 2011-03-23
> > . ID # 1520
> > "Qualcomm Incorporated's Statement about IPR related to
> > draft-ietf-codec-opus-05"
> > 2011-03-27
> > . ID # 1524
> > "Xiph.Org Foundation's Statement about IPR related to
> > draft-ietf-codec-opus-05"
> > 2011-03-29
> > . ID # 1526
> > "Broadcom Corporation's Statement about IPR related to
> > draft-ietf-codec-opus-05"
> > 2011-03-29
> > . ID # 1525
> > "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
> > 2011-07-23
> > . ID # 1602
> > "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
> > 2012-01-25
> > . ID # 1670
> > "Microsoft Corporation's Statement about IPR related to
> > draft-ietf-codec-opus-10"
> > 2012-03-12
> > . ID # 1712
> > "Huawei Technologies Co.,Ltd's Statement about IPR related to
> > draft-ietf-codec-opus-11 (1)"
> > 2012-04-02
> > . ID # 1741
> > "Huawei Technologies Co.,Ltd's Statement about IPR related to
> > draft-ietf-codec-opus-11 (2)"
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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I agree that Opus and G.711 should be the MTI audio codecs for WebRTC.<div =
class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Aug 16, 20=
12 at 5:41 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernar=
d_aboba@hotmail.com" target=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</s=
pan> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I also support the selection of OPUS and G.7=
11 as MTI. G.722 can be a SHOULD; it is widely supported.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
On Aug 16, 2012, at 4:24 PM, &quot;Richard Shockey&quot; &lt;<a href=3D"mai=
lto:richard@shockey.us">richard@shockey.us</a>&gt; wrote:<br>
<br>
&gt; Reading this list is occasionally an act of torture banned by several<=
br>
&gt; conventions ..but since you ask.<br>
&gt;<br>
&gt; I completely support the selection of Opus and G.711 as mandatory to<b=
r>
&gt; implement audio codec&#39;s ..however I&#39;m very very open minded ab=
out supporting<br>
&gt; G.722. It has it merits. =C2=A0It should be totally obvious to most th=
at if you<br>
&gt; even think about interconnecting to public E.164 networks the default =
option<br>
&gt; for VoLTE and Enterprise Voice networks is going to be G.722.<br>
&gt;<br>
&gt; If it is your goal to create globally useful stove pipes fine, =C2=A0b=
ut<br>
&gt; interconnection with existing carrier real time networks is IMHO a rat=
ional<br>
&gt; goal.<br>
&gt;<br>
&gt; As for Video .. for goodness sakes just get over it people. H.264 is t=
otally<br>
&gt; implemented everywhere on the planet Earth. So what about the intellec=
tual<br>
&gt; property problems. It&#39;s not like VP8 doesn&#39;t have problems eit=
her.<br>
&gt;<br>
&gt; I&#39;ll save my comments about the SDP offer/answer issue for another=
 day.<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ie=
tf.org</a>] On Behalf Of<br>
&gt; Cullen Jennings (fluffy)<br>
&gt; Sent: Thursday, August 16, 2012 1:16 PM<br>
&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: [rtcweb] Confirmation of consensus on audio codecs<br>
&gt;<br>
&gt;<br>
&gt; At the last meeting we took a hum on selecting Opus and G.711 as the<b=
r>
&gt; mediatory to implement audio codecs. If there is any new opinions plea=
se<br>
&gt; send them to the list by August 30th, after which the chairs will make=
 a<br>
&gt; determination of consensus.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Cullen<br>
&gt;<br>
&gt; Please note that the following IPR disclosure have been made on these<=
br>
&gt; codecs. They can be found at<br>
&gt;<br>
&gt; <a href=3D"http://datatracker.ietf.org/ipr/" target=3D"_blank">http://=
datatracker.ietf.org/ipr/</a><br>
&gt;<br>
&gt;<br>
&gt; 2010-11-07<br>
&gt; . ID # 1445<br>
&gt; &quot;Broadcom Corporation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)&quot;=
<br>
&gt; 2010-11-07<br>
&gt; . ID # 1446<br>
&gt; &quot;Xiph.Org Foundation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-00&quot;<br>
&gt; 2010-11-12<br>
&gt; . ID # 1447<br>
&gt; &quot;Broadcom Corporation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)&quot;=
<br>
&gt; 2011-03-23<br>
&gt; . ID # 1520<br>
&gt; &quot;Qualcomm Incorporated&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-05&quot;<br>
&gt; 2011-03-27<br>
&gt; . ID # 1524<br>
&gt; &quot;Xiph.Org Foundation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-05&quot;<br>
&gt; 2011-03-29<br>
&gt; . ID # 1526<br>
&gt; &quot;Broadcom Corporation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-05&quot;<br>
&gt; 2011-03-29<br>
&gt; . ID # 1525<br>
&gt; &quot;Skype Limited&#39;s Statement about IPR related to draft-ietf-co=
dec-opus-05&quot;<br>
&gt; 2011-07-23<br>
&gt; . ID # 1602<br>
&gt; &quot;Skype Limited&#39;s Statement about IPR related to draft-ietf-co=
dec-opus-07&quot;<br>
&gt; 2012-01-25<br>
&gt; . ID # 1670<br>
&gt; &quot;Microsoft Corporation&#39;s Statement about IPR related to<br>
&gt; draft-ietf-codec-opus-10&quot;<br>
&gt; 2012-03-12<br>
&gt; . ID # 1712<br>
&gt; &quot;Huawei Technologies Co.,Ltd&#39;s Statement about IPR related to=
<br>
&gt; draft-ietf-codec-opus-11 (1)&quot;<br>
&gt; 2012-04-02<br>
&gt; . ID # 1741<br>
&gt; &quot;Huawei Technologies Co.,Ltd&#39;s Statement about IPR related to=
<br>
&gt; draft-ietf-codec-opus-11 (2)&quot;<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" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--0023544702ec9eafa304c77e7117--

From stpeter@stpeter.im  Fri Aug 17 16:44:04 2012
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Agreed (as long as by G.711 we mean both PCMA and PCMU).

On 8/17/12 5:22 PM, Justin Uberti wrote:
> I agree that Opus and G.711 should be the MTI audio codecs for
> WebRTC.
> 
> 
> On Thu, Aug 16, 2012 at 5:41 PM, Bernard Aboba 
> <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>>
> wrote:
> 
> I also support the selection of OPUS and G.711 as MTI. G.722 can
> be a SHOULD; it is widely supported.
> 
> 
> 
> On Aug 16, 2012, at 4:24 PM, "Richard Shockey" <richard@shockey.us 
> <mailto:richard@shockey.us>> wrote:
> 
>> Reading this list is occasionally an act of torture banned by
>> several conventions ..but since you ask.
>> 
>> I completely support the selection of Opus and G.711 as mandatory
>> to implement audio codec's ..however I'm very very open minded
>> about
> supporting
>> G.722. It has it merits.  It should be totally obvious to most
> that if you
>> even think about interconnecting to public E.164 networks the
> default option
>> for VoLTE and Enterprise Voice networks is going to be G.722.
>> 
>> If it is your goal to create globally useful stove pipes fine,
>> but interconnection with existing carrier real time networks is
>> IMHO a
> rational
>> goal.
>> 
>> As for Video .. for goodness sakes just get over it people.
>> H.264
> is totally
>> implemented everywhere on the planet Earth. So what about the
> intellectual
>> property problems. It's not like VP8 doesn't have problems
>> either.
>> 
>> I'll save my comments about the SDP offer/answer issue for
>> another
> day.
>> 
>> -----Original Message----- From: rtcweb-bounces@ietf.org
>> <mailto:rtcweb-bounces@ietf.org>
> [mailto:rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>]
> On Behalf Of
>> Cullen Jennings (fluffy) Sent: Thursday, August 16, 2012 1:16 PM 
>> To: rtcweb@ietf.org <mailto:rtcweb@ietf.org> Subject: [rtcweb]
>> Confirmation of consensus on audio codecs
>> 
>> 
>> At the last meeting we took a hum on selecting Opus and G.711 as
>> the mediatory to implement audio codecs. If there is any new
>> opinions
> please
>> send them to the list by August 30th, after which the chairs
>> will
> make a
>> determination of consensus.
>> 
>> Thanks, Cullen
>> 
>> Please note that the following IPR disclosure have been made on
>> these codecs. They can be found at
>> 
>> http://datatracker.ietf.org/ipr/
>> 
>> 
>> 2010-11-07 . ID # 1445 "Broadcom Corporation's Statement about
>> IPR related to draft-ietf-codec-opus-00 and
>> draft-ietf-codec-description-00 (1)" 2010-11-07 . ID # 1446 
>> "Xiph.Org Foundation's Statement about IPR related to 
>> draft-ietf-codec-opus-00" 2010-11-12 . ID # 1447 "Broadcom
>> Corporation's Statement about IPR related to 
>> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00
>> (2)" 2011-03-23 . ID # 1520 "Qualcomm Incorporated's Statement
>> about IPR related to draft-ietf-codec-opus-05" 2011-03-27 . ID #
>> 1524 "Xiph.Org Foundation's Statement about IPR related to 
>> draft-ietf-codec-opus-05" 2011-03-29 . ID # 1526 "Broadcom
>> Corporation's Statement about IPR related to 
>> draft-ietf-codec-opus-05" 2011-03-29 . ID # 1525 "Skype Limited's
>> Statement about IPR related to
> draft-ietf-codec-opus-05"
>> 2011-07-23 . ID # 1602 "Skype Limited's Statement about IPR
>> related to
> draft-ietf-codec-opus-07"
>> 2012-01-25 . ID # 1670 "Microsoft Corporation's Statement about
>> IPR related to draft-ietf-codec-opus-10" 2012-03-12 . ID # 1712 
>> "Huawei Technologies Co.,Ltd's Statement about IPR related to 
>> draft-ietf-codec-opus-11 (1)" 2012-04-02 . ID # 1741 "Huawei
>> Technologies Co.,Ltd's Statement about IPR related to 
>> draft-ietf-codec-opus-11 (2)"
>> 

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Cc: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Even though 711 quality actually sucks. I only mention 722 ..AMR WB btw
since it's obvious that if you may want to interoperate with the several
billion mobile endpoints that may be deploying VoLTE over the next few
years. It might prove useful, unless this entire exercise is about creating
N squared silos. 

You are still going to have to define a MUST support video codec eventually
so just get on with it ..saying VP8 only is just unrealistic. 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Friday, August 17, 2012 1:02 PM
To: Matthew Kaufman
Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

I'll chime in here too - it'll surprise no one that I agree Opus should be
MTI. I agree on G.711 as well, as this has realistically been the lingua
franca for voip since the very beginning. Its easy to put in, no IPR and
will help with the many interop scenarios which the market will eventually
want. Even if intermediaries are needed on the media path for other things -
like ICE interop - at least we won't need transcode.

-Jonathan R.

On Fri, Aug 17, 2012 at 12:30 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
> I agree. OPUS because it will be the IETF standard for wideband audio on
the Internet, G.711 for maximum interoperability. G.722 is a reasonable
thing to put on the "SHOULD" list... there's some other good choices for the
SHOULD list, but they unfortunately come with additional IPR issues.
>
> Matthew Kaufman
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
> Behalf Of Bernard Aboba
> Sent: Thursday, August 16, 2012 5:42 PM
> To: rtcweb@ietf.org
> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
> I also support the selection of OPUS and G.711 as MTI. G.722 can be a
SHOULD; it is widely supported.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--
Jonathan Rosenberg, Ph.D.
GM Product Strategy and Research, Microsoft/Skype
jdrosen@{skype.net,jdrosen.net,microsoft.com}
http://www.jdrosen.net
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Well nip on this for a while .. a simple search against "VP8 intellectual
property issues" brings up some interesting points.  It only takes one suit.


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Basil Mohamed Gohar
Sent: Friday, August 17, 2012 4:09 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

On 08/16/2012 09:38 PM, Richard Shockey wrote:
> SOOO WRONG .. just do a simple search.  It just hasn't been fully 
> litigated ...yet.
It's impossible for something to be fully litigated.  Even someone licensing
H.264 through MPEG-LA is liable to be sued, as Motorola has taught us, and
the promises from the cartel hold no guarantees for their future.

The condition you are placing on VP8 is impossible to meet and falsely
assumed to apply to H.264.  What is real is that H.264 has a licensing
situation that excludes implementation in a free software context in most
places of the world that will use it, and VP8 does not.

Now, this is a discussion about the audio codecs, but FUD is FUD and needs
to be nipped in the bud.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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To make my position clear. The WG should support both H 264 and VP8. VP8 for
obvious reasons. H 264 since it is nearly universally deployed. 

I would like to remind the WG of the importance of this decision in another
context. Our obligation to support persons of with hearing impairments.   

http://en.wikipedia.org/wiki/Video_relay_service

http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-11-184A1.pdf

Appendix B is instructive.  

There is a marvelous opportunity here to really do "the right thing" tm. 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Ted Hardie
Sent: Thursday, August 16, 2012 1:14 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Video codec proposals due October 15th, 2012

As agreed at the recent meeting, the chairs solicit internet-drafts naming
proposed mandatory-to-implement video codecs or codec sets by October 15th,
2012.  Please include any technical data you believe supports the choice you
are proposing.  A non-exhaustive list of data you may wish to include is
found in slides 3-6 of
http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-4.pptx .

Discussion based on these will commence immediately upon receipt, rather
than waiting until the October 15th deadline, so early submission is
encouraged.

regards,

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


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On Sat, Aug 18, 2012 at 2:36 PM, Richard Shockey <richard@shockey.us> wrote:
> To make my position clear. The WG should support both H 264 and VP8.

This misses the primary point of VP8: the fact that many stakeholders
here CANNOT implement h.264.  Every stakeholder can implement VP8.
Some refuse to do so as they have a sunk investment in h.264, but this
is not the same as 'cannot'.

> H 264 since it is nearly universally deployed.

You've lost me here.  I can't use it.  Oh sure, I have the code-- I'm
just not allowed to do anything with it, and MPEG-LA won't sell me a
license because I don't track downstream.

Monty
Xiph.Org

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On Sat, Aug 18, 2012 at 5:28 PM, Monty Montgomery <xiphmont@gmail.com> wrote:
> On Sat, Aug 18, 2012 at 2:36 PM, Richard Shockey <richard@shockey.us> wrote:
>> To make my position clear. The WG should support both H 264 and VP8.

Actually, I realize I may have misread what you wrote; you said
'support' which I took to mean 'mandate'.  I assumed that's what you
meant, but it is not what you wrote.

Monty
Xiph.Org

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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: [rtcweb] Video codec proposals due October 15th, 2012
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On 8.18.2012 14:28 , "Monty Montgomery" <xiphmont@gmail.com> wrote:

>On Sat, Aug 18, 2012 at 2:36 PM, Richard Shockey <richard@shockey.us>
>wrote:
>> To make my position clear. The WG should support both H 264 and VP8.
>
>This misses the primary point of VP8: the fact that many stakeholders
>here CANNOT implement h.264.  Every stakeholder can implement VP8.
>Some refuse to do so as they have a sunk investment in h.264, but this
>is not the same as 'cannot'.

Monty,

I refuse to accept this 'CANNOT" argument.  It has been made before, in
many variants.  It's bogus, or at least unproductive.

"CANNOT" necessarily is different from the viewpoint of various entities.
However, in our case, I believe it boils down to what the entity calls an
acceptable risk, and not so much into "investments sunk".  The definition
of an acceptable risk is up to the stakeholder that makes the
determination, not you or me.  If one stakeholder entity believes that
violating, however openly, an open source license of their choice means
they CANNOT implement a certain technology, so be it.  (We all know that
with respect to H.264 implementations, some folks beliefs are very
differently from what people argue in rtcweb; x264 is published under the
GPLv2, right?).  I mean, I can accept that the stakeholder has good and
valid reasons to make the argument.  If one entity believes that they
would need to pay MPEG-LA, and that paying $6.5M per year for getting a
license under most H.264 patents put them out of business, it's again a
useful argument, but not, in isolation, a decisive one.  Those entities
obviously have the right to make their point, as you do, and as the
Mozilla guys do.  (Not listing Google here; their motivations may well be
a bit less straightforward, but that's for them to comment on and not for
me.  Just a small stab: if "investments sunk" would really play a big
role, then let's look who has sunk one of the biggest investments into a
codec technology over the past few years :-)

However, if another stakeholder entity decides, for reasons of their own,
that an implementation of VP8 may expose them to risks to the point that
legal or business says cannot, then they CANNOT.

These two CANNOTs must have the same strength when determining rough
consensus.

We have heard both voices.  The two camps are approximately of the same
strength.  Even when looking at browser market share, neither camp is an
overwhelming "winner".  Which is why I suggested in Vancouver that it may
be time to follow the WhatWG and W3C's lead and leave the mandatory video
codec selection to the browser vendors, not to a standards community that,
as this discussion shows again and again, is ill-prepared to deal with
commercial realities.  Leaving it to the browser vendors knowing that, at
least initially, we may have interop problems.

I said in Vancouver that it is unacceptable to allow the commercial
considerations of an industry operating under one business model (for
example open source), to block standards development going in a certain
direction, if there are others operating under another business model
(close source and $$$ for licensing) and they want a different direction.
I will continue to say so.  And will object quite noisily if such an
argument would be decisive in the IETF's decision making process, one way
or another.


>
>> H 264 since it is nearly universally deployed.
>
>You've lost me here.  I can't use it.  Oh sure, I have the code-- I'm
>just not allowed to do anything with it, and MPEG-LA won't sell me a
>license because I don't track downstream.

So start tracking downstream.  Your business model does not allow for it?
Change it! =20

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



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> Your business model does not allow for it?
> Change it!

You're entitled to your model, but I'm not entitled to mine?  Well, of
course, neither of us is entitled to anything.

But to the point: No, I will not.

I will continue to push for and accept only RF-or-nothing, as will
many [most?] of the individuals actually contributing the work.  This
is non-negotiable with respect to many of our participation.  That's
not an ultimatum, it's a statement of fact (and part of Xiph's
corporate charter; we must pull out if we ourselves cannot deploy it.)

Others are welcome to bring the proceedings to a crashing halt if they
so desire, and set up a nice, tightly controlled little proprietary
sandbox where no one 'unapproved' is able to contribute, and we'll see
how far that goes.  Pity, there was already so much completed work.

The Web was invented many times.  No one cared until one came along
that was Free.  Lather, rinse, repeat.

Monty
Xiph.Org
"All progress relies on the unreasonable man."

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On that search: did you also notice that it got amazingly quiet since July 2011?
S.

On Sun, Aug 19, 2012 at 4:12 AM, Richard Shockey <richard@shockey.us> wrote:
> Well nip on this for a while .. a simple search against "VP8 intellectual
> property issues" brings up some interesting points.  It only takes one suit.
>
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Basil Mohamed Gohar
> Sent: Friday, August 17, 2012 4:09 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
> On 08/16/2012 09:38 PM, Richard Shockey wrote:
>> SOOO WRONG .. just do a simple search.  It just hasn't been fully
>> litigated ...yet.
> It's impossible for something to be fully litigated.  Even someone licensing
> H.264 through MPEG-LA is liable to be sued, as Motorola has taught us, and
> the promises from the cartel hold no guarantees for their future.
>
> The condition you are placing on VP8 is impossible to meet and falsely
> assumed to apply to H.264.  What is real is that H.264 has a licensing
> situation that excludes implementation in a free software context in most
> places of the world that will use it, and VP8 does not.
>
> Now, this is a discussion about the audio codecs, but FUD is FUD and needs
> to be nipped in the bud.
> _______________________________________________
> 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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I strongly support having 722 as mandatory - interop in the enterprise 
will greatly benefit.

Opus and g.711 are +1

Tom

On 8/18/2012 11:04 AM, Richard Shockey wrote:
> Even though 711 quality actually sucks. I only mention 722 ..AMR WB btw
> since it's obvious that if you may want to interoperate with the several
> billion mobile endpoints that may be deploying VoLTE over the next few
> years. It might prove useful, unless this entire exercise is about creating
> N squared silos.
>
> You are still going to have to define a MUST support video codec eventually
> so just get on with it ..saying VP8 only is just unrealistic.
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Friday, August 17, 2012 1:02 PM
> To: Matthew Kaufman
> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
> I'll chime in here too - it'll surprise no one that I agree Opus should be
> MTI. I agree on G.711 as well, as this has realistically been the lingua
> franca for voip since the very beginning. Its easy to put in, no IPR and
> will help with the many interop scenarios which the market will eventually
> want. Even if intermediaries are needed on the media path for other things -
> like ICE interop - at least we won't need transcode.
>
> -Jonathan R.
>
> On Fri, Aug 17, 2012 at 12:30 PM, Matthew Kaufman
> <matthew.kaufman@skype.net> wrote:
>> I agree. OPUS because it will be the IETF standard for wideband audio on
> the Internet, G.711 for maximum interoperability. G.722 is a reasonable
> thing to put on the "SHOULD" list... there's some other good choices for the
> SHOULD list, but they unfortunately come with additional IPR issues.
>> Matthew Kaufman
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Bernard Aboba
>> Sent: Thursday, August 16, 2012 5:42 PM
>> To: rtcweb@ietf.org
>> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
>> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>>
>> I also support the selection of OPUS and G.711 as MTI. G.722 can be a
> SHOULD; it is widely supported.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> --
> Jonathan Rosenberg, Ph.D.
> GM Product Strategy and Research, Microsoft/Skype
> jdrosen@{skype.net,jdrosen.net,microsoft.com}
> http://www.jdrosen.net
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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We have heard both voices.  The two camps are approximately of the same
strength.  Even when looking at browser market share, neither camp is an
overwhelming "winner".  Which is why I suggested in Vancouver that it may be
time to follow the WhatWG and W3C's lead and leave the mandatory video codec
selection to the browser vendors, not to a standards community that, as this
discussion shows again and again, is ill-prepared to deal with commercial
realities.  Leaving it to the browser vendors knowing that, at least
initially, we may have interop problems.

[RS> ] I suggest that this is not a can you can kick down the road
indefinitely. At some point we will have to make a decision since the
ultimate goal is a globally interoperable service.  


I said in Vancouver that it is unacceptable to allow the commercial
considerations of an industry operating under one business model (for
example open source), to block standards development going in a certain
direction, if there are others operating under another business model (close
source and $$$ for licensing) and they want a different direction.
I will continue to say so.  And will object quite noisily if such an
argument would be decisive in the IETF's decision making process, one way or
another.


>
>> H 264 since it is nearly universally deployed.
>
>You've lost me here.  I can't use it.  Oh sure, I have the code-- I'm 
>just not allowed to do anything with it, and MPEG-LA won't sell me a 
>license because I don't track downstream.

So start tracking downstream.  Your business model does not allow for it?
Change it!  

>
>Monty
>Xiph.Org
>_______________________________________________
>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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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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> not to a standards community that, as this
> discussion shows again and again, is ill-prepared to deal with commercial
> realities.

Commercial realities?  I thought we were the Lorax: we spoke for the
Net, not for any one group's commercial realities.  There's a world
outside of MPEG.

The net is a public resource, a public utility.  Everyone is allowed
to use the pipes.  The water in the pipes is not free, but it is
provided nearly at cost and ~ universally.  The public has a right to
the pipes and the water.  Companies do not own the pipes.  They do not
own the water.  But companies reap the benefits of clean, potable
water (and a sewer system, though it's not as much fun to talk about)
like everybody else.

When the water in the pipes must be licensed, tracked, and restricted,
by a single for-profit consortium with complete authority in the
matter, it is no longer a public resource.  It becomes a bad musical.

I will not be able to buy the water in these pipes.  Not because I
can't afford it; those who control it will not sell me a license
because I'm not willing to become an extension of their water
enforcement agency.  It is not about money, it is about control.

So I am actually slightly offended at the suggestion I do not
understand the commercial reality.  I am knowingly standing against
where it wants to take us.

If WebRTC becomes encumbered, it very simply becomes yet another
for-profit play, and that's... just not very interesting.  We already
have several of those in the same space.

Monty

From lorenzo@meetecho.com  Sun Aug 19 14:32:56 2012
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From: Lorenzo Miniero <lorenzo@meetecho.com>
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On Sun, 19 Aug 2012 17:21:54 -0400
Monty Montgomery <xiphmont@gmail.com> wrote:

> > not to a standards community that, as this
> > discussion shows again and again, is ill-prepared to deal with commercial
> > realities.
> 
> Commercial realities?  I thought we were the Lorax: we spoke for the
> Net, not for any one group's commercial realities.  There's a world
> outside of MPEG.
> 
> The net is a public resource, a public utility.  Everyone is allowed
> to use the pipes.  The water in the pipes is not free, but it is
> provided nearly at cost and ~ universally.  The public has a right to
> the pipes and the water.  Companies do not own the pipes.  They do not
> own the water.  But companies reap the benefits of clean, potable
> water (and a sewer system, though it's not as much fun to talk about)
> like everybody else.
> 
> When the water in the pipes must be licensed, tracked, and restricted,
> by a single for-profit consortium with complete authority in the
> matter, it is no longer a public resource.  It becomes a bad musical.
> 
> I will not be able to buy the water in these pipes.  Not because I
> can't afford it; those who control it will not sell me a license
> because I'm not willing to become an extension of their water
> enforcement agency.  It is not about money, it is about control.
> 
> So I am actually slightly offended at the suggestion I do not
> understand the commercial reality.  I am knowingly standing against
> where it wants to take us.
> 
> If WebRTC becomes encumbered, it very simply becomes yet another
> for-profit play, and that's... just not very interesting.  We already
> have several of those in the same space.
> 


I agree.

Besides, it's not like just the browsers need to agree on the codec. Endpoints may not be browsers, but something someone else builds to make new and exciting applications out of the bricks WebRTC provides us with. As I already said some time ago, an encumbered codec would cut out most of the people who could fuel WebRTC with new ideas, use cases and so on.

Go tell a student/geek that if he wants to implement a WebRTC endpoint that takes video from a peer, manipulates it somehow for fun and then gives it back or sends it to another peer that he needs a license for that. That of course also applies to small/tiny companies or startups: how many of them would embark the WebRTC road to try and make some business, knowing they should buy god-knows-how-many licences that could sink their business before it can even start? If they're even given the choice to buy some, of course, according to what Monty said.

Just my two ents,
Lorenzo


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


-- 
Lorenzo Miniero, COB

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

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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] Qualcomm's Position on Mandatory-to-Implement Audio Codecs for RTCWeb
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Hello Tim,
Thank you for your well-thought out reply.  Please note that I was using th=
e options presented in the straw poll conducted on mandatory-to-implement (=
MTI) audio codecs at the RTCWeb meeting in Vancouver IETF 84 as the basis f=
or this position statement.  Since I have not seen official RTCWeb minutes =
from that meeting yet, I will have to reproduce from memory the voting opti=
ons presented to the attendees:  (1) G.711 only for MTI, (2) G.711 and Opus=
 only for MTI, and (3) Opus only for MTI.  Based on Cullen's email from las=
t week, I assume that the call for consensus is still around these three op=
tions alone.

I made exactly the same argument about H261 and was told (correctly) that i=
t was unacceptable to mandate a codec that would give users a
poorer than necessary experience. I think what applies for video applies fo=
r audio.

Specifically g711 is a poor codec choice in a wide variety of network situa=
tions - e.g. 3g and edge of wifi. In my experience a lower bandwidth
codec which supports wideband, dynamically variable bitrates and error corr=
ection / packet loss concealment is required to give these users audio
that is acceptable to use in the  coffee-shop / airport scenario.

Can you clarify what you would suggest as an alternative?  If one is willin=
g to concede that Opus is not the best audio codec for many lower bandwidth=
 situations (which is depicted in http://www.opus-codec.org/comparison/), t=
hen "poorer than necessary" would seem to point to mandating the codec that=
 provides best-in-class performance at a given operating condition.  This w=
ould provide optimal user experience but would also lead to many codecs bei=
ng designated as MTI.

There are many codecs which that can be said for. We happily run speex on l=
ow end smartphones. - Heck we even use the java implementation in
applets and get acceptable performance. g722 has recently dropped out of pa=
tent, uses the same bandwidth as 711 and gives significantly better audio.
Any device capable of running SRTP is capable of running either 722 or spee=
x in my experience.

Sorry to request clarification again, but do you believe G.722 should be re=
-examined as an MTI option?  Based on my understanding, .722 is not conside=
red in the current call-for-consensus (although other participants have sug=
gested it as well).  If we are to open up such a debate, it would be helpfu=
l if you could provide quantitative data backing your position.

You are confusing mandatory-to-implement and mandatory-to-use. Just because=
 Opus must be available , you don't have to use it. The devices you
mention almost all have AMR in DSP - but unfortunately that is encumbered s=
o isn't ideal as a mandatory-to-implememnt codec but may well be
used when it is available.

That was not my intention - maybe my original wording was unclear.  It is a=
 burden on app developers to have to provide SW-based Opus libraries if Opu=
s is not available natively on the HW platforms they are targeting.  They w=
ill have to do this if Opus is MTI.  They will also have to optimize their =
SW-based implementation for each target HW platform, which also is costly.

Again - making Opus mandatory to implement does not preclude supporting or =
using any other codecs.

By the way point d) applies to any codec you choose to name - so you are pr=
obably arguing for no mandatory-to-implement codec - which would be
a mistake I think. It certainly applies to g711 - in spades!

And not having Opus as MTI does not preclude its use either.  If we assume =
the "poorer than necessary" criterion over a wide range of operating condit=
ions as a prerequisite for MTI, then neither Opus nor G.711 are adequate.  =
With G.711 as the sole MTI audio codec, implementations are free to negotia=
te the best possible user experience given the environment at the moment an=
d what each side supports.

From: Tim Panton [mailto:tim@phonefromhere.com]<mailto:[mailto:tim@phonefro=
mhere.com]>
Sent: Thursday, August 16, 2012 2:41 AM
To: Mandyam, Giridhar
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Qualcomm's Position on Mandatory-to-Implement Audio C=
odecs for RTCWeb


On 15 Aug 2012, at 14:48, Mandyam, Giridhar wrote:


Hello All,
Qualcomm is in favor of G.711 being the only mandatory-to-implement codec f=
or RTCWeb.  We do not support having both Opus and G.711 as mandatory-to-im=
plement codecs.  We state the following reasons:

a)      There is no technical need to have an audio codec besides G.711 be =
designated as mandatory to implement.  Because RTCWeb uses SDP to negotiate=
 the codecs, the only reason to have anything as mandatory is to make sure =
there is always at least one codec that both sides can use, a lowest common=
 denominator.  This allows the market to decide which codecs to support, ra=
ther than edict from a standards body.  Implementations can base their choi=
ce of which codec to support based on what they are trying to optimize.  Fo=
r example, better performance for the environment at the time of the sessio=
n, such as cell phone or desktop.

I made exactly the same argument about H261 and was told (correctly) that i=
t was unacceptable to mandate a codec that would give users a
poorer than necessary experience. I think what applies for video applies fo=
r audio.

Specifically g711 is a poor codec choice in a wide variety of network situa=
tions - e.g. 3g and edge of wifi. In my experience a lower bandwidth
codec which supports wideband, dynamically variable bitrates and error corr=
ection / packet loss concealment is required to give these users audio
that is acceptable to use in the  coffee-shop / airport scenario.


b)      G.711 is universal, unencumbered, and widely implemented.    In add=
ition, note that codec implementation and testing is quite costly for chip =
and device manufacturers, and that cost is ultimately reflected in the cost=
 of the end-user device.  The computational simplicity of G.711 and its lon=
g-time availability on a broad range of platforms means that the implementa=
tion is available on low/entry-tier devices and testing costs have already =
been amortized over many years, thereby enabling its use in low cost end-us=
er devices.  If we want to see widespread use of RTCWeb for voice, than we =
will reach a much broader population if the minimum requirements do not pro=
hibit low-cost devices.

There are many codecs which that can be said for. We happily run speex on l=
ow end smartphones. - Heck we even use the java implementation in
applets and get acceptable performance. g722 has recently dropped out of pa=
tent, uses the same bandwidth as 711 and gives significantly better audio.
Any device capable of running SRTP is capable of running either 722 or spee=
x in my experience.



c)       A mandate for Opus will limit initial RTCWeb clients to use softwa=
re-based  codecs (on general purpose processors) where Opus can be implemen=
ted and tested easily until it is available on a variety of DSPs.  Even the=
n, it will likely start on high-cost platforms.  This may in turn mean that=
 RTCWeb clients could consume significantly more battery power than DSP-bas=
ed codecs used in many traditional (circuit-switched) voice services, furth=
er inhibiting the end-user from choosing RTCWeb over those services.

You are confusing mandatory-to-implement and mandatory-to-use. Just because=
 Opus must be available , you don't have to use it. The devices you
mention almost all have AMR in DSP - but unfortunately that is encumbered s=
o isn't ideal as a mandatory-to-implememnt codec but may well be
used when it is available.


d)      We do not believe Opus is more versatile than other standardized co=
decs, because any measure of versatility should take into account quality a=
t a given bitrate.  If Opus is not able to deliver superior quality at all =
supported bitrates when compared to other codecs, then it cannot be deemed =
as being more versatile simply because it supports more bitrates when compa=
red to other standardized codecs.


Again - making Opus mandatory to implement does not preclude supporting or =
using any other codecs.

By the way point d) applies to any codec you choose to name - so you are pr=
obably arguing for no mandatory-to-implement codec - which would be
a mistake I think. It certainly applies to g711 - in spades!



In conclusion - the IETF has a strong tradition of starting work with the l=
east amount of complexity and specification possible, gaining operational e=
xperience, and then refining things with revisions or extensions.  So, the =
least amount of complexity would be G.711 as the sole audio codec.

RTCweb is never going to be a _least_possible_complexity_ project. The requ=
irements mean we need  SRTP (DTLS?), a video modern codec, Echo cancellatio=
n, AGC, etc. Adding a good audio codec will not significantly extend the co=
mplexity.




_______________________________________________
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">Hello Tim,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thank you for your well-t=
hought out reply.&nbsp; Please note that I was using the options presented =
in the straw poll conducted on mandatory-to-implement (MTI) audio
 codecs at the RTCWeb meeting in Vancouver IETF 84 as the basis for this po=
sition statement.&nbsp; Since I have not seen official RTCWeb minutes from =
that meeting yet, I will have to reproduce from memory the voting options p=
resented to the attendees:&nbsp; (1) G.711
 only for MTI, (2) G.711 and Opus only for MTI, and (3) Opus only for MTI.&=
nbsp; Based on Cullen&#8217;s email from last week, I assume that the call =
for consensus is still around these three options alone.<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">I made exactly the same argument about H261 and was =
told (correctly) that it was unacceptable to mandate a codec that would giv=
e users a<o:p></o:p></p>
<p class=3D"MsoNormal">poorer than necessary experience. I think what appli=
es for video applies for audio.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Specifically g711 is a poor codec choice in a wide v=
ariety of network situations - e.g. 3g and edge of wifi. In my experience a=
 lower bandwidth<o:p></o:p></p>
<p class=3D"MsoNormal">codec which supports&nbsp;wideband, dynamically vari=
able bitrates and error correction / packet loss concealment is required to=
 give these users audio<o:p></o:p></p>
<p class=3D"MsoNormal">that is acceptable to use in the &nbsp;coffee-shop /=
 airport scenario.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Can you clarify what you =
would suggest as an alternative?&nbsp; If one is willing to concede that Op=
us is not the best audio codec for many lower bandwidth situations
 (which is depicted in <a href=3D"http://www.opus-codec.org/comparison/">ht=
tp://www.opus-codec.org/comparison/</a>), then &#8220;poorer than necessary=
&#8221; would seem to point to mandating the codec that provides best-in-cl=
ass performance at a given operating condition.&nbsp;
 This would provide optimal user experience but would also lead to many cod=
ecs being designated as MTI.
<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">There are many codecs which that can be said for. We=
 happily run speex on low end smartphones. - Heck we even use the java impl=
ementation in<o:p></o:p></p>
<p class=3D"MsoNormal">applets and get acceptable performance. g722 has rec=
ently dropped out of patent, uses the same bandwidth as 711 and gives signi=
ficantly better audio.<o:p></o:p></p>
<p class=3D"MsoNormal">Any device capable of running SRTP is capable of run=
ning either 722 or speex in my experience.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><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">Sorry to request clarific=
ation again, but do you believe G.722 should be re-examined as an MTI optio=
n?&nbsp; Based on my understanding, .722 is not considered in
 the current call-for-consensus (although other participants have suggested=
 it as well).&nbsp; If we are to open up such a debate, it would be helpful=
 if you could provide quantitative data backing your position.<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">You are confusing mandatory-to-implement and mandato=
ry-to-use. Just because Opus must be available , you don't have to use it. =
The devices you&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">mention almost all have AMR in DSP - but unfortunate=
ly that is encumbered so isn't ideal as a mandatory-to-implememnt codec but=
 may well be&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">used when it is available.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><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">That was not my intention=
 &#8211; maybe my original wording was unclear.&nbsp; It is a burden on app=
 developers to have to provide SW-based Opus libraries if Opus is not
 available natively on the HW platforms they are targeting.&nbsp; They will=
 have to do this if Opus is MTI.&nbsp; They will also have to optimize thei=
r SW-based implementation for each target HW platform, which also is costly=
.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal">Again - making Opus mandatory to implement does not =
preclude supporting or using any other codecs.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">By the way point d) applies to any codec you choose =
to name - so you are probably arguing for no mandatory-to-implement codec -=
 which would be<o:p></o:p></p>
<p class=3D"MsoNormal">a mistake I think.&nbsp;It certainly applies to g711=
 - in spades!<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><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">And not having Opus as MT=
I does not preclude its use either.&nbsp; If we assume the &#8220;poorer th=
an necessary&#8221; criterion over a wide range of operating conditions as
 a prerequisite for MTI, then neither Opus nor G.711 are adequate.&nbsp; Wi=
th G.711 as the sole MTI audio codec, implementations are free to negotiate=
 the best possible user experience given the environment at the moment and =
what each side supports.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Tim Pant=
on
</span><a href=3D"mailto:[mailto:tim@phonefromhere.com]"><span style=3D"fon=
t-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">[mailt=
o:tim@phonefromhere.com]</span></a><span style=3D"font-size:10.0pt;font-fam=
ily:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<br>
<b>Sent:</b> Thursday, August 16, 2012 2:41 AM<br>
<b>To:</b> Mandyam, Giridhar<br>
<b>Cc:</b> </span><a href=3D"mailto:rtcweb@ietf.org"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">rtcweb@iet=
f.org</span></a><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&qu=
ot;,&quot;sans-serif&quot;"><br>
<b>Subject:</b> Re: [rtcweb] Qualcomm's Position on Mandatory-to-Implement =
Audio Codecs for RTCWeb<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On 15 Aug 2012, at 14:48, Mandyam, Giridhar wrote:<o=
:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hello All,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Qualcomm is in favor of G.711 being the=
 only mandatory-to-implement codec for RTCWeb.&nbsp; We do not support havi=
ng both Opus and G.711 as mandatory-to-implement codecs.&nbsp; We
 state the following reasons:<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">a)</span><=
span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D=
"apple-converted-space">&nbsp;</span></span><span style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">There
 is no technical need to have an audio codec besides G.711 be designated as=
 mandatory to implement.&nbsp; Because RTCWeb uses SDP to negotiate the cod=
ecs, the only reason to have anything as mandatory is to make sure there is=
 always at least one codec that both
 sides can use, a lowest common denominator.&nbsp; This allows the market t=
o decide which codecs to support, rather than edict from a standards body.&=
nbsp; Implementations can base their choice of which codec to support based=
 on what they are trying to optimize.&nbsp; For
 example, better performance for the environment at the time of the session=
, such as cell phone or desktop.<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I made exactly the same argument about H261 and was =
told (correctly) that it was unacceptable to mandate a codec that would giv=
e users a<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">poorer than necessary experience. I think what appli=
es for video applies for audio.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Specifically g711 is a poor codec choice in a wide v=
ariety of network situations - e.g. 3g and edge of wifi. In my experience a=
 lower bandwidth<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">codec which supports&nbsp;wideband, dynamically vari=
able bitrates and error correction / packet loss concealment is required to=
 give these users audio<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">that is acceptable to use in the &nbsp;coffee-shop /=
 airport scenario.&nbsp;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">b)</span><=
span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D=
"apple-converted-space">&nbsp;</span></span><span style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">G.711
 is universal, unencumbered, and widely implemented.&nbsp;&nbsp;&nbsp; In a=
ddition, note that codec implementation and testing is quite costly for chi=
p and device manufacturers, and that cost is ultimately reflected in the co=
st of the end-user device.&nbsp; The computational simplicity
 of G.711 and its long-time availability on a broad range of platforms mean=
s that the implementation is available on low/entry-tier devices and testin=
g costs have already been amortized over many years, thereby enabling its u=
se in low cost end-user devices.&nbsp;
 If we want to see widespread use of RTCWeb for voice, than we will reach a=
 much broader population if the minimum requirements do not prohibit low-co=
st devices.&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">There are many codecs which that can be said for. We=
 happily run speex on low end smartphones. - Heck we even use the java impl=
ementation in<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">applets and get acceptable performance. g722 has rec=
ently dropped out of patent, uses the same bandwidth as 711 and gives signi=
ficantly better audio.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Any device capable of running SRTP is capable of run=
ning either 722 or speex in my experience.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">c)</span><=
span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span cl=
ass=3D"apple-converted-space">&nbsp;</span></span><span style=3D"font-size:=
11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">A
 mandate for Opus will limit initial RTCWeb clients to use software-based&n=
bsp; codecs (on general purpose processors) where Opus can be implemented a=
nd tested easily until it is available on a variety of DSPs.&nbsp; Even the=
n, it will likely start on high-cost platforms.&nbsp;
 This may in turn mean that RTCWeb clients could consume significantly more=
 battery power than DSP-based codecs used in many traditional (circuit-swit=
ched) voice services, further inhibiting the end-user from choosing RTCWeb =
over those services.<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">You are confusing mandatory-to-implement and mandato=
ry-to-use. Just because Opus must be available , you don't have to use it. =
The devices you&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">mention almost all have AMR in DSP - but unfortunate=
ly that is encumbered so isn't ideal as a mandatory-to-implememnt codec but=
 may well be&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">used when it is available.&nbsp;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">d)</span><=
span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D=
"apple-converted-space">&nbsp;</span></span><span style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">We
 do not believe Opus is more versatile than other standardized codecs, beca=
use any measure of versatility should take into account quality at a given =
bitrate.&nbsp; If Opus is not able to deliver superior quality at all suppo=
rted bitrates when compared to other
 codecs, then it cannot be deemed as being more versatile simply because it=
 supports more bitrates when compared to other standardized codecs.<o:p></o=
:p></span></p>
</div>
</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">Again - making Opus mandatory to implement does not =
preclude supporting or using any other codecs.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">By the way point d) applies to any codec you choose =
to name - so you are probably arguing for no mandatory-to-implement codec -=
 which would be<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a mistake I think.&nbsp;It certainly applies to g711=
 - in spades!<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">In conclusion - the IETF has a strong t=
radition of starting work with the least amount of complexity and specifica=
tion possible, gaining operational experience, and then
 refining things with revisions or extensions.&nbsp; So, the least amount o=
f complexity would be G.711 as the sole audio codec.<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">RTCweb is never going to be a _least_possible_comple=
xity_ project. The requirements mean we need &nbsp;SRTP (DTLS?), a video mo=
dern codec, Echo cancellation, AGC, etc. Adding a good audio codec will not=
 significantly extend the complexity.<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>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
rtcweb mailing list<br>
</span><a href=3D"mailto:rtcweb@ietf.org"><span style=3D"font-size:13.5pt;f=
ont-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;">rtcweb@ietf.org</s=
pan></a><span style=3D"font-size:13.5pt;font-family:&quot;Helvetica&quot;,&=
quot;sans-serif&quot;"><br>
</span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span style=
=3D"font-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quo=
t;">https://www.ietf.org/mailman/listinfo/rtcweb</span></a><span style=3D"f=
ont-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;"><=
o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_CAC8DBE4E9704C41BCB290C2F3CC921A162BAAFEnasanexd01hnaqu_--

From john@jlc.net  Mon Aug 20 09:26:27 2012
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Date: Mon, 20 Aug 2012 12:26:26 -0400
From: John Leslie <john@jlc.net>
To: Tim Panton <tim@phonefromhere.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "Mandyam, Giridhar" <mandyam@quicinc.com>
Subject: [rtcweb] Mandatory-to-Implement Audio Codecs for RTCWeb
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Tim Panton <tim@phonefromhere.com> wrote:
> On 15 Aug 2012, at 14:48, Mandyam, Giridhar wrote:
> 
>> Qualcomm is in favor of G.711 being the only mandatory-to-implement
>> codec for RTCWeb.

   An otherwise valid argument is mortally wounded by this statement.
We participate as individuals in IETF Working Groups; and stating
something as a "position" tends to make others consign you to "the rough".

>> We do not support having both Opus and G.711 as mandatory-to-implement
>> codecs.  We state the following reasons:
>>  
>> There is no technical need to have an audio codec besides G.711 be
>> designated as mandatory to implement. Because RTCWeb uses SDP to
>> negotiate the codecs, the only reason to have anything as mandatory
>> is to make sure there is always at least one codec that both sides
>> can use, a lowest common denominator.

   Exactly true!

>> This allows the market to decide which codecs to support, rather than
>> edict from a standards body. Implementations can base their choice of
>> which codec to support based on what they are trying to optimize. For
>> example, better performance for the environment at the time of the
>> session, such as cell phone or desktop.

   The intended situation as RTCWEB matures is that the endpoints will
choose among a number of codecs which both support, taking into account
the characteristics of the link between them.

> I made exactly the same argument about H261 and was told (correctly)
> that it was unacceptable to mandate a codec that would give users a
> poorer than necessary experience.

   No, that is not "correctly"!

   It would be unacceptable to mandate the _use_ of a codec giving
sub-standard experience; but it is _exactly_ acceptable to mandate the
_implementation_ of a codec which gives "poorer than necessary
experience" in even a _majority_ of actual conditions.

   We certainly hope there will be better options available for
particular conditions; and we certainly hope that the conditions
available will improve.

> I think what applies for video applies for audio.

   Mostly true; but given a century of experience with audio-only
telephone use, there is perhaps wiggle-room for failure to agree on
a video codec. (What I mean here is that _if_ we can't find any video
codec that all RTCWEB stacks "can" implement, we could theoretically
fall back to a short list of SHOULD implement.)

> Specifically g711 is a poor codec choice in a wide variety of
> network situations - e.g. 3g and edge of wifi.

   True, but that's the wrong consideration. A mandatory-to-implement
codec isn't asked to give "good" results in every case.

> In my experience a lower bandwidth codec which supports wideband,
> dynamically variable bitrates and error correction / packet loss
> concealment is required to give these users audio that is acceptable
> to use in the coffee-shop / airport scenario. 

   That's a good argument; but it goes beyond the target we're aiming
for as "mandatory-to-implement".

>> G.711 is universal, unencumbered, and widely implemented. In addition,
>> note that codec implementation and testing is quite costly for chip
>> and device manufacturers, and that cost is ultimately reflected in
>> the cost of the end-user device. The computational simplicity of
>> G.711 and its long-time availability on a broad range of platforms
>> means that the implementation is available on low/entry-tier devices
>> and testing costs have already been amortized over many years,
>> thereby enabling its use in low cost end-user devices.

   This is also a good argument; and it is pretty much on-target.

>> If we want to see widespread use of RTCWeb for voice, than we will
>> reach a much broader population if the minimum requirements do not
>> prohibit low-cost devices. 

   This is a weaker argument, but still appropriate IMHO.

> There are many codecs which that can be said for. We happily run
> speex on low end smartphones. - Heck we even use the java
> implementation in applets and get acceptable performance. g722 has
> recently dropped out of patent, uses the same bandwidth as 711 and
> gives significantly better audio. Any device capable of running
> SRTP is capable of running either 722 or speex in my experience. 

   And this is a good counter-argument...

>> A mandate for Opus will limit initial RTCWeb clients to use
>> software-based codecs (on general purpose processors) where Opus
>> can be implemented and tested easily until it is available on a
>> variety of DSPs. Even then, it will likely start on high-cost
>> platforms.

   I find this argument weak. I would expect any RTCWEB device to
need to _start_ on this path.

>> This may in turn mean that RTCWeb clients could consume
>> significantly more battery power than DSP-based codecs used in
>> many traditional (circuit-switched) voice services, further
>> inhibiting the end-user from choosing RTCWeb over those services.

   That's a valid argument; but it's about a commercial tradeoff,
and I don't see a good reason why we need to optimize for a particular
commercial environment.

> You are confusing mandatory-to-implement and mandatory-to-use.

   A lot of us seem to be suffering from that confusion!

> Just because Opus must be available, you don't have to use it.

   You _might_ have to use it... But it's only required as a last
resort, and it's perfectly reasonable to warn the end-user that
battery life will suffer.

> The devices you mention almost all have AMR in DSP - but
> unfortunately that is encumbered so isn't ideal as a
> mandatory-to-implememnt codec but may well be used when it is
> available. 
> 
>> We do not believe Opus is more versatile than other standardized
>> codecs, because any measure of versatility should take into account
>> quality at a given bitrate. If Opus is not able to deliver superior
>> quality at all supported bitrates when compared to other codecs,
>> then it cannot be deemed as being more versatile simply because it
>> supports more bitrates when compared to other standardized codecs.

   This is a straw-man. :^(

> Again - making Opus mandatory to implement does not preclude
> supporting or using any other codecs. 
> 
>> In conclusion - the IETF has a strong tradition of starting work
>> with the least amount of complexity and specification possible,
>> gaining operational experience, and then refining things with
>> revisions or extensions.

   That's not as true as it used to be. :^(

>> So, the least amount of complexity would be G.711 as the sole
>> audio codec.
> 
> RTCweb is never going to be a _least_possible_complexity_ project.
> The requirements mean we need SRTP (DTLS?), a video modern codec,
> Echo cancellation, AGC, etc. Adding a good audio codec will not
> significantly extend the complexity.

   This thread is full of good arguments!

   I hope we will contiue it, but hold a little better to what we
mean by "mandatory-to-implement".

--
John Leslie <john@jlc.net>

From ted.ietf@gmail.com  Mon Aug 20 11:45:59 2012
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From: Ted Hardie <ted.ietf@gmail.com>
To: Monty Montgomery <xiphmont@gmail.com>, Stephan Wenger <stewe@stewe.org>,  Lorenzo Miniero <lorenzo@meetecho.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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Howdy,

I just want to remind folks contributing to this thread where we are
in the process:  getting documents which make *concrete, technical*
proposals.  Those need to be in by October 15th.  Please focus your
energy on that.

thanks,

Ted Hardie

From emcho@sip-communicator.org  Mon Aug 20 15:12:39 2012
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From: Emil Ivov <emcho@jitsi.org>
Date: Tue, 21 Aug 2012 00:12:16 +0200
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Subject: [rtcweb] H.264's freeness for Internet apps
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On February 2, 2010 MPEG-LA made an announcemnt that sounded like it
would not charge royalties for Internet video:

http://www.mpegla.com/main/Pages/Media.aspx

Does anyone have an idea as to how the exact licensing text would
apply to RTC[web] applications?

Emil

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On Mon, Aug 20, 2012 at 6:12 PM, Emil Ivov <emcho@jitsi.org> wrote:
> On February 2, 2010 MPEG-LA made an announcemnt that sounded like it
> would not charge royalties for Internet video:

There had been worry that the h.264 pool would add new royalties to
the content streams themselves in 2015; The announcement was that
there would be no new royalties in 2015, not that any existing
royalties were going away.

And yes, the exact wording confused many tech reporters who took it to
mean "H.264 is free now!"

Monty

From stephane.proust@orange.com  Tue Aug 21 00:57:55 2012
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From: <stephane.proust@orange.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Confirmation of consensus on audio codecs
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For voice and multimedia communication services as targeted by WebRTC, two =
codecs, G.711 and G.722, are fully fitting the conditions to get a mandator=
y to implement status: a fully ensured unencumbered status, a wide implemen=
tation in devices and legacy networks and a low complexity that ensures the=
y can be supported on any types of plate-forms and terminals.

So we support both G.711 and G.722 to be specified as "mandatory to impleme=
nt" codecs since G.722 will guarantee a much higher voice quality at same b=
it rate and almost no additional cost. At a minimum G.722 should be (strong=
ly) recommended.

With respect to OPUS, it has currently no footprint on the market, its impl=
ementation complexity and resulting costs are unknown, including with respe=
ct to IPRs issues, and some performance aspects are still to be more deeply=
 assessed, like for instance quality with packet losses and jitter which is=
 a key issue for usage over internet.=20

In addition, we believe that two additional requirements are necessary to m=
ake WebRTC a future proof technology and suitable for a wide range of appli=
cations and environments:
Firstly, the optional support of other widely deployed legacy codecs must b=
e allowed which includes at least the codecs implemented in millions of mob=
ile terminals: AMR and AMR-WB.
Secondly, the technology must be open enough to facilitate the usage of any=
 other codecs supported by the device but external to the browser.


-----Message d'origine-----
De=A0: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part =
de Cullen Jennings (fluffy)
Envoy=E9=A0: jeudi 16 ao=FBt 2012 19:16
=C0=A0: rtcweb@ietf.org
Objet=A0: [rtcweb] Confirmation of consensus on audio codecs


At the last meeting we took a hum on selecting Opus and G.711 as the mediat=
ory to implement audio codecs. If there is any new opinions please send the=
m to the list by August 30th, after which the chairs will make a determinat=
ion of consensus.

Thanks,
Cullen

Please note that the following IPR disclosure have been made on these codec=
s. They can be found at=20

http://datatracker.ietf.org/ipr/


2010-11-07=09
. ID # 1445
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (1)"
2010-11-07=09
. ID # 1446
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-00"
2010-11-12=09
. ID # 1447
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (2)"
2011-03-23=09
. ID # 1520
"Qualcomm Incorporated's Statement about IPR related to draft-ietf-codec-op=
us-05"
2011-03-27=09
. ID # 1524
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-05"
2011-03-29=09
. ID # 1526
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-05"
2011-03-29=09
. ID # 1525
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
2011-07-23=09
. ID # 1602
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
2012-01-25=09
. ID # 1670
"Microsoft Corporation's Statement about IPR related to draft-ietf-codec-op=
us-10"
2012-03-12=09
. ID # 1712
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (1)"
2012-04-02=09
. ID # 1741
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (2)"



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

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Thank you.


From randell-ietf@jesup.org  Tue Aug 21 08:02:43 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On 8/21/2012 3:57 AM, stephane.proust@orange.com wrote:
> For voice and multimedia communication services as targeted by WebRTC, two codecs, G.711 and G.722, are fully fitting the conditions to get a mandatory to implement status: a fully ensured unencumbered status, a wide implementation in devices and legacy networks and a low complexity that ensures they can be supported on any types of plate-forms and terminals.
>
> So we support both G.711 and G.722 to be specified as "mandatory to implement" codecs since G.722 will guarantee a much higher voice quality at same bit rate and almost no additional cost. At a minimum G.722 should be (strongly) recommended.

I agree G.722 should be a "SHOULD".

Note I said "I".  While in reality corporations and organizations that 
IETF contributors are associated with have opinions on IETF activities, 
officially IETF recognizes individuals as contributors.

> With respect to OPUS, it has currently no footprint on the market, its implementation complexity and resulting costs are unknown, including with respect to IPRs issues, and some performance aspects are still to be more deeply assessed, like for instance quality with packet losses and jitter which is a key issue for usage over internet.

Opus was designed within the IETF specifically to handle usage over the 
Internet better than existing codecs.

> In addition, we believe that two additional requirements are necessary to make WebRTC a future proof technology and suitable for a wide range of applications and environments:
> Firstly, the optional support of other widely deployed legacy codecs must be allowed which includes at least the codecs implemented in millions of mobile terminals: AMR and AMR-WB.

The spec does not restrict what codecs can be supported beyond the MTI 
codecs.  You can offer any codecs you support.

> Secondly, the technology must be open enough to facilitate the usage of any other codecs supported by the device but external to the browser.

This is an implementation detail for browsers and not part of the spec; 
a browser is free to offer any codecs supported by the platform if it 
wishes.

-- 
Randell Jesup
randell-ietf@jesup.org


From ted.ietf@gmail.com  Tue Aug 21 08:35:18 2012
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On Tue, Aug 21, 2012 at 12:57 AM,  <stephane.proust@orange.com> wrote:

> In addition, we believe that two additional requirements are necessary to make WebRTC a future proof technology and suitable for a wide range of applications and environments:
> Firstly, the optional support of other widely deployed legacy codecs must be allowed which includes at least the codecs implemented in millions of mobile terminals: AMR and AMR-WB.
> Secondly, the technology must be open enough to facilitate the usage of any other codecs supported by the device but external to the browser.
>
>
 Stephane,

On you first point, the fundamental design of RTCWEB allows for the
negotiation of any codec mutually supported.  The decision to chose a
Mandatory-to-implement was not made to eliminate other choices, but to
eliminate interoperability failures by ensuring that at least one
common codec is always available.

On your second point, the group presumes that codec support does not
come from the downloadable Javascript application, but from the
application environment into which it is downloaded (commonly a
browser or mobile environment using similar technology).  Promoting
support for those environments to have access to codecs supported by
the underlying hardware is the best way to further this goal, at least
in my personal opinion.

best regards,

Ted Hardie

From christer.holmberg@ericsson.com  Wed Aug 22 13:47:04 2012
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Subject: [rtcweb] Codec control and SDP offer/answer frequency
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Hi,

In Vancouver two proposals for codec were presented, one based on COP and o=
ne based on the SDP imageattr attribute. The result from the straw poll was=
 that people preferred to move forward with the SDP based solution.

When Harald presented his poker game example, he mentioned that there could=
 be changes with a few seconds interval.

For my clarification, does that mean that a UA would send SDP offer/answers=
 every few second???

If so, I don't think that would create good results.

First, it would cause heavy load on every entity that has to parse the SDP,=
 as they need to parse the whole message body, and then figure out what has=
 changed.

Second, due to race conditions rules, SIP transaction rules etc, and *other=
* usages of SDP offer/answer within the session, there is a high risk that =
SDP offers (not only those related to codec control) would be rejected. Off=
er/answer is a rather "heavy" mechanism, and I don't think it's intended to=
 be used in the suggested way. Normally you would send one (or a few) offer=
/answer(s) during call establishment, and between zero and a few during the=
 session.

Or, did I missunderstand something?

Regards,

Christer

From harald@alvestrand.no  Thu Aug 23 02:00:43 2012
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Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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On 08/22/2012 10:47 PM, Christer Holmberg wrote:
> Hi,
>
> In Vancouver two proposals for codec were presented, one based on COP and one based on the SDP imageattr attribute. The result from the straw poll was that people preferred to move forward with the SDP based solution.
>
> When Harald presented his poker game example, he mentioned that there could be changes with a few seconds interval.
>
> For my clarification, does that mean that a UA would send SDP offer/answers every few second???
>
> If so, I don't think that would create good results.
I don't quite buy the analysis - details below.

The example given (the "poker game") would almost certainly be a very 
specialized application, with no requirements for interworking (you 
can't dial into a game from a videophone), and would most likely be 
implemented via a central server (if nothing else, to take care that 
nobody's cheating).

For this type of application, where the end systems are already doing 
data exchange that is not representable in SDP (the game itself), I 
think a sensible architecture would be to pass the SDP offer/answer over 
the data channel. This means that traffic latency for passing SDP is on 
the same order as for media.
>
> First, it would cause heavy load on every entity that has to parse the SDP, as they need to parse the whole message body, and then figure out what has changed.
Given that in such a specialized application, the only entity that has 
to parse SDP is the central node - how much CPU time do you think SDP 
parsing will take? (number of CPU-milliseconds?)
>
> Second, due to race conditions rules, SIP transaction rules etc, and *other* usages of SDP offer/answer within the session, there is a high risk that SDP offers (not only those related to codec control) would be rejected.
Why, given that it's a point-to-point exchange? Admitted, glare will 
happen, and will have to be resolved in the usual ways (ROAP gave one 
mechanism that the game protocol can adopt).
> Offer/answer is a rather "heavy" mechanism, and I don't think it's intended to be used in the suggested way.
Well.. the Arpanet was originally intended for connecting teletypes to 
mainframes. It's proved to be useful in a few scenarios it was not 
originally created for.
> Normally you would send one (or a few) offer/answer(s) during call establishment, and between zero and a few during the session.
>
> Or, did I missunderstand something?
A specialized poker game is not intended to be interoperable with a 
videophone?
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Thu Aug 23 02:38:27 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 23 Aug 2012 11:38:23 +0200
Thread-Topic: [rtcweb] Codec control and SDP offer/answer frequency
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Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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Hi Harald,

>> In Vancouver two proposals for codec were presented, one based on COP an=
d one based on the SDP imageattr attribute. The result from the straw poll =
was that people preferred to move forward with the SDP based solution.
>>
>> When Harald presented his poker game example, he mentioned that there co=
uld be changes with a few seconds interval.
>>
>> For my clarification, does that mean that a UA would send SDP offer/answ=
ers every few second???
>>
>> If so, I don't think that would create good results.
>
> I don't quite buy the analysis - details below.
>
> The example given (the "poker game") would almost certainly be a very
> specialized application, with no requirements for interworking (you
> can't dial into a game from a videophone), and would most likely be
> implemented via a central server (if nothing else, to take care that
> nobody's cheating).
>
> For this type of application, where the end systems are already doing
> data exchange that is not representable in SDP (the game itself), I
> think a sensible architecture would be to pass the SDP offer/answer over
> the data channel. This means that traffic latency for passing SDP is on
> the same order as for media.

Offer/answer over the data channel???

Slide 3 of your presentation talks about the signalling channel.

Now, if you want to define a datachannel protocol that uses SDP syntax, fin=
e, but please don't call it offer/answer :)

Also, you would need to describe how the data channel protocol relates to n=
egotiating being done on the signalling plane.

>> First, it would cause heavy load on every entity that has to parse the S=
DP, as they need to parse the whole message body, and then figure out what =
has changed.
>
> Given that in such a specialized application, the only entity that has
> to parse SDP is the central node - how much CPU time do you think SDP
> parsing will take? (number of CPU-milliseconds?)

I don't have numbers, but I can tell you that it is a relatively heavy proc=
ess. And, it's not only about the parsing, but also about the processing of=
 the parsed result.

In addition, are you actually negotiating anything, or are you simply "indi=
cating"?


>> Second, due to race conditions rules, SIP transaction rules etc, and *ot=
her* usages of SDP offer/answer within the session, there is a high risk th=
at SDP offers (not only those related to codec control) would be rejected.
>
> Why, given that it's a point-to-point exchange? Admitted, glare will
> happen, and will have to be resolved in the usual ways (ROAP gave one
> mechanism that the game protocol can adopt).

Sure, there are procedures how to handle it. My point is that it may cause =
bad user experience - espeically if more critical offers are rejected (e.g.=
 offers where you are adding new streams, doing un-hold etc).

>> Offer/answer is a rather "heavy" mechanism, and I don't think it's inten=
ded to be used in the suggested way.
>Well.. the Arpanet was originally intended for connecting teletypes to
>mainframes. It's proved to be useful in a few scenarios it was not origina=
lly created for.
>> Normally you would send one (or a few) offer/answer(s) during call estab=
lishment, and between zero and a few during the session.
>>
>> Or, did I missunderstand something?
>A specialized poker game is not intended to be interoperable with a
>videophone?

Let's not get stucked on the poker game use-case then.

In general, if we think there will be use-cases which would trigger frequen=
t offer/answers, I think it's the wrong approach.

Regards,

Christer=

From harald@alvestrand.no  Thu Aug 23 03:40:41 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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On 08/23/2012 11:38 AM, Christer Holmberg wrote:
> Hi Harald,
>
>>> In Vancouver two proposals for codec were presented, one based on COP and one based on the SDP imageattr attribute. The result from the straw poll was that people preferred to move forward with the SDP based solution.
>>>
>>> When Harald presented his poker game example, he mentioned that there could be changes with a few seconds interval.
>>>
>>> For my clarification, does that mean that a UA would send SDP offer/answers every few second???
>>>
>>> If so, I don't think that would create good results.
>> I don't quite buy the analysis - details below.
>>
>> The example given (the "poker game") would almost certainly be a very
>> specialized application, with no requirements for interworking (you
>> can't dial into a game from a videophone), and would most likely be
>> implemented via a central server (if nothing else, to take care that
>> nobody's cheating).
>>
>> For this type of application, where the end systems are already doing
>> data exchange that is not representable in SDP (the game itself), I
>> think a sensible architecture would be to pass the SDP offer/answer over
>> the data channel. This means that traffic latency for passing SDP is on
>> the same order as for media.
> Offer/answer over the data channel???
>
> Slide 3 of your presentation talks about the signalling channel.
>
> Now, if you want to define a datachannel protocol that uses SDP syntax, fine, but please don't call it offer/answer :)
>
> Also, you would need to describe how the data channel protocol relates to negotiating being done on the signalling plane.
The format of the signalling channel is defined to be outside the scope 
of the specification, so it's perfectly legitimate to use the data 
channel for that purpose - application decision!
>
>>> First, it would cause heavy load on every entity that has to parse the SDP, as they need to parse the whole message body, and then figure out what has changed.
>> Given that in such a specialized application, the only entity that has
>> to parse SDP is the central node - how much CPU time do you think SDP
>> parsing will take? (number of CPU-milliseconds?)
> I don't have numbers, but I can tell you that it is a relatively heavy process. And, it's not only about the parsing, but also about the processing of the parsed result.
>
> In addition, are you actually negotiating anything, or are you simply "indicating"?
See draft-lennox - there may be negotiation going on, if we change the 
boundaries.
I see the normal use of this feature to be a change in q values only.
>
>
>>> Second, due to race conditions rules, SIP transaction rules etc, and *other* usages of SDP offer/answer within the session, there is a high risk that SDP offers (not only those related to codec control) would be rejected.
>> Why, given that it's a point-to-point exchange? Admitted, glare will
>> happen, and will have to be resolved in the usual ways (ROAP gave one
>> mechanism that the game protocol can adopt).
> Sure, there are procedures how to handle it. My point is that it may cause bad user experience - espeically if more critical offers are rejected (e.g. offers where you are adding new streams, doing un-hold etc).
All state changes have to be negotiated in a way that makes sure they 
eventually resolve, no matter what they are. This is not a problem 
specific to resolution negotiation.
>
>>> Offer/answer is a rather "heavy" mechanism, and I don't think it's intended to be used in the suggested way.
>> Well.. the Arpanet was originally intended for connecting teletypes to
>> mainframes. It's proved to be useful in a few scenarios it was not originally created for.
>>> Normally you would send one (or a few) offer/answer(s) during call establishment, and between zero and a few during the session.
>>>
>>> Or, did I missunderstand something?
>> A specialized poker game is not intended to be interoperable with a
>> videophone?
> Let's not get stucked on the poker game use-case then.
>
> In general, if we think there will be use-cases which would trigger frequent offer/answers, I think it's the wrong approach.
And I want to be a lot more specific before worrying.
In the interworking case, I see that there can be issues, because 
existing systems have existing procedures - so I accept the theory that 
every-few-seconds offer/answer is a problem there.

In the WebRTC application case, I want to see a specific problem before 
I start worrying.

>
> Regards,
>
> Christer


From christer.holmberg@ericsson.com  Thu Aug 23 04:05:24 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>
Date: Thu, 23 Aug 2012 13:05:20 +0200
Thread-Topic: [rtcweb] Codec control and SDP offer/answer frequency
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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Hi,=20

>>>> In Vancouver two proposals for codec were presented, one based on COP =
and one based on the SDP imageattr attribute. The result from the straw pol=
l was that people preferred to move forward with the SDP based solution.
>>>>
>>>> When Harald presented his poker game example, he mentioned that there =
could be changes with a few seconds interval.
>>>>
>>>> For my clarification, does that mean that a UA would send SDP offer/an=
swers every few second???
>>>>
>>>> If so, I don't think that would create good results.
>>>> I don't quite buy the analysis - details below.
>>>
>>> The example given (the "poker game") would almost certainly be a very=20
>>> specialized application, with no requirements for interworking (you=20
>>> can't dial into a game from a videophone), and would most likely be=20
>>> implemented via a central server (if nothing else, to take care that=20
>>> nobody's cheating).
>>>
>>> For this type of application, where the end systems are already doing=20
>>> data exchange that is not representable in SDP (the game itself), I=20
>>> think a sensible architecture would be to pass the SDP offer/answer=20
>>> over the data channel. This means that traffic latency for passing=20
>>> SDP is on the same order as for media.
>> Offer/answer over the data channel???
>>
>> Slide 3 of your presentation talks about the signalling channel.
>>
>> Now, if you want to define a datachannel protocol that uses SDP=20
>> syntax, fine, but please don't call it offer/answer :)
>>
>> Also, you would need to describe how the data channel protocol relates t=
o negotiating being done on the signalling plane.
> The format of the signalling channel is defined to be outside the scope o=
f the specification, so it's perfectly legitimate to use the data channel f=
or that purpose - application decision!

Sure, applications can do whatever they want.

But, I think it would be good to have a standardized mechanisms which works=
 for any application. AFAIK, that was the whole purpose of the discussion i=
n Vancouver.


>>>> First, it would cause heavy load on every entity that has to parse the=
 SDP, as they need to parse the whole message body, and then figure out wha=
t has changed.
>>> Given that in such a specialized application, the only entity that=20
>>> has to parse SDP is the central node - how much CPU time do you think=20
>>> SDP parsing will take? (number of CPU-milliseconds?)
>> I don't have numbers, but I can tell you that it is a relatively heavy p=
rocess. And, it's not only about the parsing, but also about the processing=
 of the parsed result.
>>
>> In addition, are you actually negotiating anything, or are you simply "i=
ndicating"?
>
> See draft-lennox - there may be negotiation going on, if we change the bo=
undaries.

Sure - for which offer/answer IS a good mechanism, as I assume the boundari=
es won't change that often.

> I see the normal use of this feature to be a change in q values only.
>>
>>
>>>> Second, due to race conditions rules, SIP transaction rules etc, and *=
other* usages of SDP offer/answer within the=20
>>>> session, there is a high risk that SDP offers (not only those related =
to codec control) would be rejected.
>>> Why, given that it's a point-to-point exchange? Admitted, glare will=20
>>> happen, and will have to be resolved in the usual ways (ROAP gave one=20
>>> mechanism that the game protocol can adopt).
>> Sure, there are procedures how to handle it. My point is that it may cau=
se bad user experience - espeically if=20
>> more critical offers are rejected (e.g. offers where you are adding new =
streams, doing un-hold etc).
> All state changes have to be negotiated in a way that makes sure they eve=
ntually resolve, no matter what=20
> they are. This is not a problem specific to resolution negotiation.

Correct. But, if you can indicate resolution changes without affecting othe=
r procedures, that makes the system more flexible.

If you need to re-NEGOTIATE something, then I agree you very likely will ne=
ed something like offer/answer.

And, if you are talking about a data channel protocol, my issues may not be=
 valid. So, maybe I'm the only one who missunderstood your message in Vanco=
uver, then :)


>>>> Offer/answer is a rather "heavy" mechanism, and I don't think it's int=
ended to be used in the suggested way.
>>> Well.. the Arpanet was originally intended for connecting teletypes=20
>>> to mainframes. It's proved to be useful in a few scenarios it was not o=
riginally created for.
>>>> Normally you would send one (or a few) offer/answer(s) during call est=
ablishment, and between zero and a few during the session.
>>>>
>>>> Or, did I missunderstand something?
>>> A specialized poker game is not intended to be interoperable with a=20
>>> videophone?
>> Let's not get stucked on the poker game use-case then.
>>
>> In general, if we think there will be use-cases which would trigger freq=
uent offer/answers, I think it's the wrong approach.
>> And I want to be a lot more specific before worrying.
> In the interworking case, I see that there can be issues, because existin=
g systems have existing procedures - so I accept the theory that every-few-=
seconds offer/answer is a problem there.
>
> In the WebRTC application case, I want to see a specific problem before I=
 start worrying.

I belive also CLUE may have usage for this mechanism, eventhough I don't th=
ink there is any such formal decission made.

Regards,

Christer



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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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Clipping to a single issue....

On 08/23/2012 01:05 PM, Christer Holmberg wrote:
> Also, you would need to describe how the data channel protocol relates to negotiating being done on the signalling plane.
>> The format of the signalling channel is defined to be outside the scope of the specification, so it's perfectly legitimate to use the data channel for that purpose - application decision!
> Sure, applications can do whatever they want.
>
> But, I think it would be good to have a standardized mechanisms which works for any application. AFAIK, that was the whole purpose of the discussion in Vancouver.
>
Since we've settled on using SDP at the UA/Application interface, 
representing the data in SDP is a valid way of describing that interface 
- thus indeed making it available for use by any application written to 
that interface.

The particular worry you raised was how fast applications can do 
offer/answers - which is dependent on how their signalling channel is 
implemented, and how many intermediate nodes have to process the data.

That's what I'm thinking will be different in the Pokerstars example 
than in a SIP-telephone-gateway example.


From christer.holmberg@ericsson.com  Thu Aug 23 08:11:25 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>
Date: Thu, 23 Aug 2012 17:11:21 +0200
Thread-Topic: [rtcweb] Codec control and SDP offer/answer frequency
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Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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Hi,=20

>>>> Also, you would need to describe how the data channel protocol relates=
 to negotiating being done on the signalling plane.
>>> The format of the signalling channel is defined to be outside the scope=
 of the specification, so it's perfectly legitimate to use the data channel=
 for that purpose - application decision!
>> Sure, applications can do whatever they want.
>>
>> But, I think it would be good to have a standardized mechanisms which wo=
rks for any application. AFAIK, that was the whole purpose of the discussio=
n in Vancouver.
>>
> Since we've settled on using SDP at the UA/Application interface,

Correct (eventhough someone in Seattle may have a different opinion ;)

> representing the data in SDP is a valid way of describing that interface
>- thus indeed making it available for use by any application written to th=
at interface.

Correct.

> The particular worry you raised was how fast applications can do offer/an=
swers - which is dependent on=20
> how their signalling channel is implemented, and how many intermediate no=
des have to process the data.

Correct.

> That's what I'm thinking will be different in the Pokerstars example than=
 in a SIP-telephone-gateway example.

I agree that the impacts are different - but I'd like to have a solution th=
at works in all cases.

So, if I understand you correctly: you suggest that the codec control infor=
mation is transported using SDP/a=3Dimageattr "internally" (between app and=
 browser), but then people can send it over the wire however they want?

Regards,

Christer


From harald@alvestrand.no  Thu Aug 23 13:36:41 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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On 08/23/2012 05:11 PM, Christer Holmberg wrote:
> Hi,
>
>>>>> Also, you would need to describe how the data channel protocol relates to negotiating being done on the signalling plane.
>>>> The format of the signalling channel is defined to be outside the scope of the specification, so it's perfectly legitimate to use the data channel for that purpose - application decision!
>>> Sure, applications can do whatever they want.
>>>
>>> But, I think it would be good to have a standardized mechanisms which works for any application. AFAIK, that was the whole purpose of the discussion in Vancouver.
>>>
>> Since we've settled on using SDP at the UA/Application interface,
> Correct (eventhough someone in Seattle may have a different opinion ;)
>
>> representing the data in SDP is a valid way of describing that interface
>> - thus indeed making it available for use by any application written to that interface.
> Correct.
>
>> The particular worry you raised was how fast applications can do offer/answers - which is dependent on
>> how their signalling channel is implemented, and how many intermediate nodes have to process the data.
> Correct.
>
>> That's what I'm thinking will be different in the Pokerstars example than in a SIP-telephone-gateway example.
> I agree that the impacts are different - but I'd like to have a solution that works in all cases.
>
> So, if I understand you correctly: you suggest that the codec control information is transported using SDP/a=imageattr "internally" (between app and browser), but then people can send it over the wire however they want?
Yes. I believe that's the model we have agreed on.


From Markus.Isomaki@nokia.com  Thu Aug 23 14:00:13 2012
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From: <Markus.Isomaki@nokia.com>
To: <harald@alvestrand.no>, <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Codec control and SDP offer/answer frequency
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Codec control and SDP offer/answer frequency
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Hi,

Harald Alvestrand wrote:
>
>On 08/23/2012 05:11 PM, Christer Holmberg wrote:
>>
>> So, if I understand you correctly: you suggest that the codec control
>information is transported using SDP/a=3Dimageattr "internally" (between a=
pp
>and browser), but then people can send it over the wire however they want?
>
>Yes. I believe that's the model we have agreed on.
>

I think that is fully in line with how SDP is treated in RTCWeb in general =
(at least so far), so there does not seem to be anything special about this=
 case.

Markus

From lishitao@huawei.com  Thu Aug 23 21:17:35 2012
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From: Lishitao <lishitao@huawei.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Confirmation of consensus on audio codecs
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Randell Jesup
> Sent: Tuesday, August 21, 2012 11:02 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>=20
> > With respect to OPUS, it has currently no footprint on the market, its
> implementation complexity and resulting costs are unknown, including with
> respect to IPRs issues, and some performance aspects are still to be more
> deeply assessed, like for instance quality with packet losses and jitter =
which is a
> key issue for usage over internet.
>=20
> Opus was designed within the IETF specifically to handle usage over the
> Internet better than existing codecs.
>

My understanding for choosing the MTI codec is to find the baseline so that=
 the negotiation can be successful,=20
We are not here to find the best one.

If we consider the situation that G.711 has been widely used on the market,=
 mandating G.711 only as the baseline(MTI) is more appropriate

shitao.

From christer.holmberg@ericsson.com  Thu Aug 23 23:22:39 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>, "harald@alvestrand.no" <harald@alvestrand.no>
Date: Fri, 24 Aug 2012 08:18:37 +0200
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Hi,

>>> So, if I understand you correctly: you suggest that the codec control
>>information is transported using SDP/a=3Dimageattr "internally" (between =
app
>>and browser), but then people can send it over the wire however they want=
?
>>
>>Yes. I believe that's the model we have agreed on.
>>
>
> I think that is fully in line with how SDP is treated in RTCWeb in genera=
l (at least so far), so there does not seem to be anything special about th=
is case.

Fair enough.

But, that means that IF we use SDP offer/answer on the wire, and that offer=
/answer is not appropriate for codec control, we need to define another mec=
hanism.

(By "we" I don't necessarily mean the rtcweb wg, but the community in gener=
al)

Regards,

Christer=

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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On Fri, Aug 24, 2012 at 5:10 AM, Lishitao <lishitao@huawei.com> wrote:
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Randell Jesup
>> Sent: Tuesday, August 21, 2012 11:02 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>>
>> > With respect to OPUS, it has currently no footprint on the market, its
>> implementation complexity and resulting costs are unknown, including with
>> respect to IPRs issues, and some performance aspects are still to be more
>> deeply assessed, like for instance quality with packet losses and jitter which is a
>> key issue for usage over internet.
>>
>> Opus was designed within the IETF specifically to handle usage over the
>> Internet better than existing codecs.
>>
>
> My understanding for choosing the MTI codec is to find the baseline so that the negotiation can be successful,
> We are not here to find the best one.
>
> If we consider the situation that G.711 has been widely used on the market, mandating G.711 only as the baseline(MTI) is more appropriate
>
> shitao.

This should not be a question purely of interop with legacy devices,
but primarily one about what is the minimum level of interop that is
acceptable for WebRTC to WebRTC negotiated calls.

If G711 is the only MTI codec then we can easily end up in a situation
where that is all we have available for cross browser WebRTC to WebRTC
calls, which would greatly undermine the effort to replace browser
plugins.

Neil

From ekr@rtfm.com  Fri Aug 24 08:15:22 2012
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Subject: [rtcweb] Filling in details on "trickle ICE"
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I've been working on doing Firefox's trickle ICE implementation and
have had to make a bunch of decisions that suggest to me that perhaps
we need an update to RFC 5245 to specify exactly how trickle ICE
behaves. Here are some of the things we need to resolve.


WHEN CHECKS START
RFC 5245 currently specifies that connectivity checks start upon
receiving the offer/answer (See Section 5):

   When an agent receives an initial offer, it will check if the offerer
   supports ICE, determine its own role, gather candidates, prioritize
   them, choose default candidates, encode and send an answer, and for
   full implementations, form the check lists and begin connectivity
   checks.

However, if you are trickling all your candidates, then this algorithm
is no longer appropriate, since the check list will be empty, so we
need a new algorithm here. There seem to be a number of natural algorithms
here, including:

(a) as soon as any check list is non-empty, start that media stream.
(b) as soon as the first media stream has a non-empty check list
start that media stream
(c) as soon as all check lists are non-empty start the first
media stream.

Note that S 5.7.4 specifies that you are supposed to start with the
first media stream, which is what makes (b) and (c) compelling.


WHEN ICE ENDS
7.1.3.3 describes the conditions required to update the check list
based on transaction completion:

   Regardless of whether the check was successful or failed, the
   completion of the transaction may require updating of check list and
   timer states.

   If all of the pairs in the check list are now either in the Failed or
   Succeeded state:

   o  If there is not a pair in the valid list for each component of the
      media stream, the state of the check list is set to Failed.


Consider the following case:

- Alice and Bob are on different addresses which both use RFC 1918
  space.
- Alice sends Bob the candidate 10.0.0.10 as her first candidate at
  This just happens to correspond to an existing host on Bob's network.
- Bob creates a check list consisting solely of 10.0.0.10 and starts
  checks.
- Because 10.0.0.10 is a real host on Bob's network, it generates an ICMP
  error and per S 7.1.3.1 Bob marks the transaction as Failed:

Unless I have missed something, Given the algorithm above, the check
list now consists solely of Failed candidates and the valid list is
now empty, so this causes the media stream to be Failed. If this is
the only media stream, according to 8.1.2. the entire ICE process is
now marked as Failed.

What's letting this happen is a race condition between candidates
arriving and checks failing. I suspect that there are others, so we
probably need some way to handle these cases.


WHEN NEW TRICKLE CANDIDATES CAN BE PROCESSED
A related question is when new trickle candidates can be processed.
Say that a new candidate is received when a stream has a nominated
candidate on the valid list? Do we add a new candidate to the
list? Note that S 8.1.2 would ordinarily have us stopping checks
on all other extant candidates pairs.


HOW DO OTHER MEDIA STREAMS GET UNFROZEN
Say I finish checks on one stream but another stream successfully.
Ordinarily, I would start to unfreeze other check lists, but what
if those are empty? How do they get unfrozen when candidates come
in?


INTERACTION WITH NON-TRICKLE ICE
I'm concerned about the potential interactions between trickle ICE
and non trickle ICE. Obviously, an extant ICE implementation can
only handle getting all the candidates at once, so in order to
have interop we need to hold all the candidates until we get the
"null" finished candidate indication. But this means that in the
case where the non-trickle version is the offerer, then checks
aren't performed simultaneously. Consider the following case:

- Alice is a non-trickle endpoint
- Bob is a trickle endpoint
- Alice and Bob are behind standard address-dependent filtering NATs
- Bob has two STUN servers but one is down.
- The JS holds off on sending Bob's offer to Alice until all the
  candidates are in. (Which, as I say above, is I think mandatory).

I believe we get the following behavior.


Alice          Alice NAT                Bob NAT           Bob
 Bob STUN 1           Bob STUN 2

OFFER ------------------------------------------------------>
						            BINDING REQUEST ->
 						            BINDING REQUEST -------------------------->
							    <- BINDING RESPONSE
					            onicecandidate(...)
                X <----------------------- CONNECTIVITY CHECK
 						            BINDING REQUEST (RT) --------------------->
                X <------------------ CONNECTIVITY CHECK (RT)
		  		                          ...
						     [Give up on STUN 2]
					            onicecandidate(null)
                                 /----------------------  ANSWER
                                /              [Give up on
CONNECTIVITY CHECK; ICE fails]
<------------------------------/				


The key thing here is that if the STUN timers for connectivity checks
and for initial binding requests are similar, then Bob's checks can
fail before Alice has time to open up a pinhole. This doesn't happen
in regular ICE because the checks are synched in each direction.


I suspect that this list is incomplete, but I think it's enough to
suggest that there may be some details in trickle ICE that need to
at least be documented and perhaps investigated/resolved.

To the chairs: can you provide some guidance as to how to proceed with
this discussion, what forum it should be discussed in, etc.?

Thanks,
-Ekr

From fluffy@iii.ca  Fri Aug 24 08:26:10 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE" - where to do this work
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>=20
> To the chairs: can you provide some guidance as to how to proceed with
> this discussion, what forum it should be discussed in, etc.?


With my chair hat on here=85. this work clearly needs to change some =
stuff in RFC 5245 (the ICE spec) which was done in mmusic WG. So I think =
the default location for updating it is probably the mmusic WG. I'll =
ping the AD's and chairs but I think unless we here something different, =
mmusic is the place for this.

Cullen <rtcweb co-chair>



From martin.thomson@gmail.com  Fri Aug 24 08:55:07 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 24 August 2012 08:14, Eric Rescorla <ekr@rtfm.com> wrote:
> I've been working on doing Firefox's trickle ICE implementation and
> have had to make a bunch of decisions that suggest to me that perhaps
> we need an update to RFC 5245 to specify exactly how trickle ICE
> behaves. Here are some of the things we need to resolve.

These are not necessarily interoperability issues.  You can solve all
of these problems without creating another RFC.

You make the decision to start and stop, succeed or fail, nominate or
leave.  That decision might be guided by an RFC, but you are
responsible for what your code does.

...unless you consider it impossible to take actions unless those
actions have the blessing of RFC5245.

--Martin

From ekr@rtfm.com  Fri Aug 24 09:00:16 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On Fri, Aug 24, 2012 at 8:55 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 24 August 2012 08:14, Eric Rescorla <ekr@rtfm.com> wrote:
>> I've been working on doing Firefox's trickle ICE implementation and
>> have had to make a bunch of decisions that suggest to me that perhaps
>> we need an update to RFC 5245 to specify exactly how trickle ICE
>> behaves. Here are some of the things we need to resolve.
>
> These are not necessarily interoperability issues.  You can solve all
> of these problems without creating another RFC.
>
> You make the decision to start and stop, succeed or fail, nominate or
> leave.  That decision might be guided by an RFC, but you are
> responsible for what your code does.

I suppose it depends on whether you think it's important for
independently developed Web applications and/or softphones
to be able to talk to each other.


> ...unless you consider it impossible to take actions unless those
> actions have the blessing of RFC5245.

Not at all. We could take actions blessed by other RFCs.

-Ekr

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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 24 August 2012 08:59, Eric Rescorla <ekr@rtfm.com> wrote:
> On Fri, Aug 24, 2012 at 8:55 AM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
> I suppose it depends on whether you think it's important for
> independently developed Web applications and/or softphones
> to be able to talk to each other.

Maybe I needed to be clearer:

>> These are not necessarily interoperability issues.  You can solve all
>> of these problems without creating another RFC.

I'd be interested in hearing the interoperability reason for, for
example, giving up on a particular component instead of waiting for
additional candidates.  What effect does that have on
interoperability.

It's fairly clear that if you are trickling candidates out, then an
strictly conforming implementation of RFC 5245 on the other end could
fail.  That's just a reason to not trickle at all.

--Martin

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On Fri, Aug 24, 2012 at 9:12 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 24 August 2012 08:59, Eric Rescorla <ekr@rtfm.com> wrote:
>> On Fri, Aug 24, 2012 at 8:55 AM, Martin Thomson
>> <martin.thomson@gmail.com> wrote:
>> I suppose it depends on whether you think it's important for
>> independently developed Web applications and/or softphones
>> to be able to talk to each other.
>
> Maybe I needed to be clearer:
>
>>> These are not necessarily interoperability issues.  You can solve all
>>> of these problems without creating another RFC.
>
> I'd be interested in hearing the interoperability reason for, for
> example, giving up on a particular component instead of waiting for
> additional candidates.  What effect does that have on
> interoperability.

I'm not following what you are saying here. Ignoring the question of
non-trickle (i.e., strict RFC 5245) endpoints, if you are a trickle endpoint
that gives up under conditions when RFC 5245 says you should give
up, you are likely to have problems. You don't think it's worth writing
that down somewhere more permanent/definitive than an email
to the mailing list?

-Ekr

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On 24 August 2012 09:42, Eric Rescorla <ekr@rtfm.com> wrote:
> I'm not following what you are saying here. Ignoring the question of
> non-trickle (i.e., strict RFC 5245) endpoints, if you are a trickle endpoint
> that gives up under conditions when RFC 5245 says you should give
> up, you are likely to have problems.

If you are a trickle endpoint, then you are not implementing RFC 5245.
 RFC 5245 might not explicitly exclude the possibility, but it is
certainly implicit, as the problems you cite in your email attest.

What I think we want to with trickle is to create a new method.  That
method is only like RFC 5245 to the extent that it shares some
properties: the good parts.

> You don't think it's worth writing
> that down somewhere more permanent/definitive than an email
> to the mailing list?

Only if it addresses a specific interoperability need does it need to
be in an RFC.

From fluffy@iii.ca  Fri Aug 24 11:16:43 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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I'm confused on what you are trying to say. Are you saying=20

1) we don't need trickle ICE

2) we can do trickily ICE but no standards need to be written for two =
different devices to do trickle ICE


On Aug 24, 2012, at 9:56 , Martin Thomson wrote:

> On 24 August 2012 09:42, Eric Rescorla <ekr@rtfm.com> wrote:
>> I'm not following what you are saying here. Ignoring the question of
>> non-trickle (i.e., strict RFC 5245) endpoints, if you are a trickle =
endpoint
>> that gives up under conditions when RFC 5245 says you should give
>> up, you are likely to have problems.
>=20
> If you are a trickle endpoint, then you are not implementing RFC 5245.
> RFC 5245 might not explicitly exclude the possibility, but it is
> certainly implicit, as the problems you cite in your email attest.
>=20
> What I think we want to with trickle is to create a new method.  That
> method is only like RFC 5245 to the extent that it shares some
> properties: the good parts.
>=20
>> You don't think it's worth writing
>> that down somewhere more permanent/definitive than an email
>> to the mailing list?
>=20
> Only if it addresses a specific interoperability need does it need to
> be in an RFC.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Fri Aug 24 14:22:43 2012
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From: Martin Thomson <martin.thomson@gmail.com>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 24 August 2012 11:16, Cullen Jennings <fluffy@iii.ca> wrote:
>
> I'm confused on what you are trying to say. Are you saying
>
> 1) we don't need trickle ICE
>
> 2) we can do trickily ICE but no standards need to be written for two different devices to do trickle ICE

It's just 2)

Trickle candidates is a *new thing*, a new problem that needs to be
solved.  It's fairly clear that trickling is an optimization that was
never considered in the design of ICE.

We can solve the problem by writing an RFC, but you can also write
fewer standards and enable the same outcome.

An RFC would specify how this differs from ICE and what new behaviours
are expected from endpoints.  That RFC would also describe how
implementations would discover if peers support the new behaviour so
that they don't do things that would surprise or break implementations
that did not.


The alternative is to recognize that ICE specifies a number of
features that are really, really important, both from an
interoperability and a security standpoint:
  The use of STUN Binding requests to gather server reflexive addresses.
  The use of TURN to allocate TURN relays.
  The use of STUN Binding requests to establish connectivity and consent.
  Constraints over how checks can be sent (which are actually not all
suitable in this context for other reasons).
  A way to resolve the problem of who performs candidate pair selection.

Then ICE has a bunch of really useful advice on what implementations
might do: gather candidates, prioritize them, check them, retry
checks, track those that succeed, etc...  None of that is really
necessary, just super handy.

The signalling stuff in ICE is not relevant to the rtcweb
architecture.  So we can ignore that part.  WebRTC decided to use it,
but that's their problem.

The most invasive thing that ICE does is describe a specific algorithm
in some detail.  Implicit in this algorithm is the fact that all
candidates have to be signaled together.  This is also stuff that we
don't *need* in rtcweb.

This might have been useful when you use RFCs to describe every facet
of your application, but it is in fact unnecessary for applications of
the sort that exist in rtcweb.  Many of these applications want things
like trickle candidates, which can give better session initialization
performance.  Many of these will have code on both endpoints.

Obviously, applications that want to interoperate with ICE
implementations will have to play nice with ICE.  But this is usually
known a priori.  Building signaling for this is presumptuous.

I understand how someone might conclude that you need to standardize
this behaviour.  It depends on what preconceptions you have.  If you
wanted to port trickle candidates to your SIP PBX, then I can see how
the standardization option might seem attractive.

The Microsoft API proposal makes all of this possible by only taking
the really important stuff from RFC 5245.

--Martin

From ted.ietf@gmail.com  Fri Aug 24 15:04:49 2012
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Subject: [rtcweb] Please review: draft minutes
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--bcaec54d4b0206dd5904c80a2a33
Content-Type: text/plain; charset=ISO-8859-1

Additions and corrections to these would be welcome; they will be
posted (as draft) from Wednesday of next week.
Thanks to our note takers:  Christer for Day 1 and Brian for Day 2.

Ted

--bcaec54d4b0206dd5904c80a2a33
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From Jim.Barnett@genesyslab.com  Fri Aug 24 15:32:24 2012
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Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Martin Thomson" <martin.thomson@gmail.com>, "Cullen Jennings" <fluffy@iii.ca>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Martin, =20
  Just to make sure that I understand your position, I take you to be
saying: =20
1) in the case where both endpoints have downloaded their apps from the
same server, they can do trickle ICE any way that they want.
2) in the case where an application is  talking to an unknown or legacy
peer, trickle  ICE is a bad idea because there is no standard way to do
it. (i.e. try it at your own risk)

Is this correct?

- Jim=20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Martin Thomson
Sent: Friday, August 24, 2012 5:23 PM
To: Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

On 24 August 2012 11:16, Cullen Jennings <fluffy@iii.ca> wrote:
>
> I'm confused on what you are trying to say. Are you saying
>
> 1) we don't need trickle ICE
>
> 2) we can do trickily ICE but no standards need to be written for two=20
> different devices to do trickle ICE

It's just 2)

Trickle candidates is a *new thing*, a new problem that needs to be
solved.  It's fairly clear that trickling is an optimization that was
never considered in the design of ICE.

We can solve the problem by writing an RFC, but you can also write fewer
standards and enable the same outcome.

An RFC would specify how this differs from ICE and what new behaviours
are expected from endpoints.  That RFC would also describe how
implementations would discover if peers support the new behaviour so
that they don't do things that would surprise or break implementations
that did not.


The alternative is to recognize that ICE specifies a number of features
that are really, really important, both from an interoperability and a
security standpoint:
  The use of STUN Binding requests to gather server reflexive addresses.
  The use of TURN to allocate TURN relays.
  The use of STUN Binding requests to establish connectivity and
consent.
  Constraints over how checks can be sent (which are actually not all
suitable in this context for other reasons).
  A way to resolve the problem of who performs candidate pair selection.

Then ICE has a bunch of really useful advice on what implementations
might do: gather candidates, prioritize them, check them, retry checks,
track those that succeed, etc...  None of that is really necessary, just
super handy.

The signalling stuff in ICE is not relevant to the rtcweb architecture.
So we can ignore that part.  WebRTC decided to use it, but that's their
problem.

The most invasive thing that ICE does is describe a specific algorithm
in some detail.  Implicit in this algorithm is the fact that all
candidates have to be signaled together.  This is also stuff that we
don't *need* in rtcweb.

This might have been useful when you use RFCs to describe every facet of
your application, but it is in fact unnecessary for applications of the
sort that exist in rtcweb.  Many of these applications want things like
trickle candidates, which can give better session initialization
performance.  Many of these will have code on both endpoints.

Obviously, applications that want to interoperate with ICE
implementations will have to play nice with ICE.  But this is usually
known a priori.  Building signaling for this is presumptuous.

I understand how someone might conclude that you need to standardize
this behaviour.  It depends on what preconceptions you have.  If you
wanted to port trickle candidates to your SIP PBX, then I can see how
the standardization option might seem attractive.

The Microsoft API proposal makes all of this possible by only taking the
really important stuff from RFC 5245.

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

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On 24 August 2012 15:33, Jim Barnett <Jim.Barnett@genesyslab.com> wrote:
> Martin,
>   Just to make sure that I understand your position, I take you to be
> saying:
> 1) in the case where both endpoints have downloaded their apps from the
> same server, they can do trickle ICE any way that they want.
> 2) in the case where an application is  talking to an unknown or legacy
> peer, trickle  ICE is a bad idea because there is no standard way to do
> it. (i.e. try it at your own risk)
>
> Is this correct?

A fair summation.  In 1) they can do whatever they want within the
necessary security constraints.

From matthew.kaufman@skype.net  Mon Aug 27 12:53:57 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Jim Barnett <Jim.Barnett@genesyslab.com>, Martin Thomson <martin.thomson@gmail.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
Thread-Index: AQHNggtYU4RJsiahn0ePX6kIvBvYA5dpHYyAgAABRICAAAOYgIAACHQAgAADxICAABZ7AIAAM/2AgAATuQCABIcVcA==
Date: Mon, 27 Aug 2012 19:53:50 +0000
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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABkgnnVBBAH=HCkn_cksBs_9A_hm=VfFwcTtvOM3C7XB2h2KTA@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca> <CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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If both ends want to do a full, standards compliant (which also implies *no=
t* trickle) ICE, then we can bake that into the browser following the exist=
ing RFC as specification.

If both ends want to do something that isn't that, then we either need to w=
rite down *exactly* how they do that "something else" (which would imply an=
 RFC or three for things like how trickle ICE works, how it is discovered, =
what SDP implications it has, etc.) *or* we need to provide knobs that allo=
w the developer, through Javascript, to ensure that both ends do the same (=
or compatible) "something elses".

Note that the only reason ICE-like STUN connectivity tests are a MUST is th=
at it is required for consent verification. There are any number of reasons=
 why an endpoint might wish to do something other than what a full standard=
s-compliant ICE implementation would require... this thread has been about =
the issues around trickle candidates, but there's also the case where you'r=
e on a webpage of mine and I know I'm going to send your call via a gateway=
 that has a public IP address. There is no reason to run any of what ICE re=
quires *except* the security-considerations-mandated consent verification, =
and only in the browser-to-gateway test direction.

Again, we could write another RFC covering that case... or we could just do=
 what our (Microsoft's) proposal suggests and provide the developer with th=
e controls necessary to implement *any* of these use cases, including the m=
ode that matches the current ICE RFC.

As a side effect, the developer then *also* has the flexibility to improve =
interoperation with things like pre-final ICE implementations, as long as t=
hey meet the requirements around STUN connectivity tests.

So to recap, if you want something fancy like ICE with trickle candidates y=
ou have two options:

X) Give the developer the flexibility to build variations upon ICE within t=
he security constraints, or
Y) Start writing Internet Drafts describing all the variations upon ICE you=
 might wish to use and then get every browser vendor to add them

Matthew Kaufman


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Jim Barnett
Sent: Friday, August 24, 2012 3:33 PM
To: Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

Martin,
  Just to make sure that I understand your position, I take you to be
saying: =20
1) in the case where both endpoints have downloaded their apps from the sam=
e server, they can do trickle ICE any way that they want.
2) in the case where an application is  talking to an unknown or legacy pee=
r, trickle  ICE is a bad idea because there is no standard way to do it. (i=
.e. try it at your own risk)

Is this correct?

- Jim=20



From Jim.Barnett@genesyslab.com  Mon Aug 27 13:14:26 2012
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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABkgnnVBBAH=HCkn_cksBs_9A_hm=VfFwcTtvOM3C7XB2h2KTA@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca><CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com> <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7B56@tk5ex14mbxc272.redmond.corp.microsoft.com>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Matthew Kaufman" <matthew.kaufman@skype.net>, "Martin Thomson" <martin.thomson@gmail.com>, "Cullen Jennings" <fluffy@iii.ca>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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One question I have is whether we consider the "Browser RTC Trapezoid"
to be in scope.  In this use case, the two UAs download their
applications from different web servers.  In such a case, I don't see
how we can enable trickle ICE without specifying _exactly_ how  it is
supposed to work (or, alternatively, specifying a protocol that the two
web servers will use to negotiate how to do it). =20

Handling the trapezoid is a _lot_ more work than the case where both UAs
download their applications from the same server (or from the case where
a single WebRTC UA is talking to a legacy device).  Have we made a
decision on whether it is in scope?  In any case, it would certainly
clarify the discussion for me if I knew whether we were considering this
use case or not.  A number of claims have been and are being made on the
list that strike me as obviously false if this use case is in scope -
and perfectly sensible if it's not. =20

- Jim
P.S.  My personal opinion is that it would make sense to defer the
trapezoid until a hypothetical version 2.  That way it would not inform
any immediate decisions about the APIs, but we would have to consider
what it would take to extend them to handle it. (I would think that
would involve mostly adding more detail, so forward compatibility might
not be hard to achieve.)

-----Original Message-----
From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]=20
Sent: Monday, August 27, 2012 3:54 PM
To: Jim Barnett; Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] Filling in details on "trickle ICE"

If both ends want to do a full, standards compliant (which also implies
*not* trickle) ICE, then we can bake that into the browser following the
existing RFC as specification.

If both ends want to do something that isn't that, then we either need
to write down *exactly* how they do that "something else" (which would
imply an RFC or three for things like how trickle ICE works, how it is
discovered, what SDP implications it has, etc.) *or* we need to provide
knobs that allow the developer, through Javascript, to ensure that both
ends do the same (or compatible) "something elses".

Note that the only reason ICE-like STUN connectivity tests are a MUST is
that it is required for consent verification. There are any number of
reasons why an endpoint might wish to do something other than what a
full standards-compliant ICE implementation would require... this thread
has been about the issues around trickle candidates, but there's also
the case where you're on a webpage of mine and I know I'm going to send
your call via a gateway that has a public IP address. There is no reason
to run any of what ICE requires *except* the
security-considerations-mandated consent verification, and only in the
browser-to-gateway test direction.

Again, we could write another RFC covering that case... or we could just
do what our (Microsoft's) proposal suggests and provide the developer
with the controls necessary to implement *any* of these use cases,
including the mode that matches the current ICE RFC.

As a side effect, the developer then *also* has the flexibility to
improve interoperation with things like pre-final ICE implementations,
as long as they meet the requirements around STUN connectivity tests.

So to recap, if you want something fancy like ICE with trickle
candidates you have two options:

X) Give the developer the flexibility to build variations upon ICE
within the security constraints, or
Y) Start writing Internet Drafts describing all the variations upon ICE
you might wish to use and then get every browser vendor to add them

Matthew Kaufman


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Jim Barnett
Sent: Friday, August 24, 2012 3:33 PM
To: Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

Martin,
  Just to make sure that I understand your position, I take you to be
saying: =20
1) in the case where both endpoints have downloaded their apps from the
same server, they can do trickle ICE any way that they want.
2) in the case where an application is  talking to an unknown or legacy
peer, trickle  ICE is a bad idea because there is no standard way to do
it. (i.e. try it at your own risk)

Is this correct?

- Jim=20



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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Jim Barnett <Jim.Barnett@genesyslab.com>, Martin Thomson <martin.thomson@gmail.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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The "two different web servers" "trapezoid" case would require a trunking p=
rotocol between these servers, such as SIP or Jingle, which can describe "t=
rickle ICE" such that the browsers could be instructed properly by their re=
spective web servers.

I believe that with Jingle, there is a written spec for that, and for SIP, =
there is not (at present).

Matthew Kaufman

-----Original Message-----
From: Jim Barnett [mailto:Jim.Barnett@genesyslab.com]=20
Sent: Monday, August 27, 2012 1:15 PM
To: Matthew Kaufman; Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] Filling in details on "trickle ICE"

One question I have is whether we consider the "Browser RTC Trapezoid"
to be in scope.  In this use case, the two UAs download their applications =
from different web servers.  In such a case, I don't see how we can enable =
trickle ICE without specifying _exactly_ how  it is supposed to work (or, a=
lternatively, specifying a protocol that the two web servers will use to ne=
gotiate how to do it). =20

Handling the trapezoid is a _lot_ more work than the case where both UAs do=
wnload their applications from the same server (or from the case where a si=
ngle WebRTC UA is talking to a legacy device).  Have we made a decision on =
whether it is in scope?  In any case, it would certainly clarify the discus=
sion for me if I knew whether we were considering this use case or not.  A =
number of claims have been and are being made on the list that strike me as=
 obviously false if this use case is in scope - and perfectly sensible if i=
t's not. =20

- Jim
P.S.  My personal opinion is that it would make sense to defer the trapezoi=
d until a hypothetical version 2.  That way it would not inform any immedia=
te decisions about the APIs, but we would have to consider what it would ta=
ke to extend them to handle it. (I would think that would involve mostly ad=
ding more detail, so forward compatibility might not be hard to achieve.)

-----Original Message-----
From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]
Sent: Monday, August 27, 2012 3:54 PM
To: Jim Barnett; Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] Filling in details on "trickle ICE"

If both ends want to do a full, standards compliant (which also implies
*not* trickle) ICE, then we can bake that into the browser following the ex=
isting RFC as specification.

If both ends want to do something that isn't that, then we either need to w=
rite down *exactly* how they do that "something else" (which would imply an=
 RFC or three for things like how trickle ICE works, how it is discovered, =
what SDP implications it has, etc.) *or* we need to provide knobs that allo=
w the developer, through Javascript, to ensure that both ends do the same (=
or compatible) "something elses".

Note that the only reason ICE-like STUN connectivity tests are a MUST is th=
at it is required for consent verification. There are any number of reasons=
 why an endpoint might wish to do something other than what a full standard=
s-compliant ICE implementation would require... this thread has been about =
the issues around trickle candidates, but there's also the case where you'r=
e on a webpage of mine and I know I'm going to send your call via a gateway=
 that has a public IP address. There is no reason to run any of what ICE re=
quires *except* the security-considerations-mandated consent verification, =
and only in the browser-to-gateway test direction.

Again, we could write another RFC covering that case... or we could just do=
 what our (Microsoft's) proposal suggests and provide the developer with th=
e controls necessary to implement *any* of these use cases, including the m=
ode that matches the current ICE RFC.

As a side effect, the developer then *also* has the flexibility to improve =
interoperation with things like pre-final ICE implementations, as long as t=
hey meet the requirements around STUN connectivity tests.

So to recap, if you want something fancy like ICE with trickle candidates y=
ou have two options:

X) Give the developer the flexibility to build variations upon ICE within t=
he security constraints, or
Y) Start writing Internet Drafts describing all the variations upon ICE you=
 might wish to use and then get every browser vendor to add them

Matthew Kaufman


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Jim Barnett
Sent: Friday, August 24, 2012 3:33 PM
To: Martin Thomson; Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

Martin,
  Just to make sure that I understand your position, I take you to be
saying: =20
1) in the case where both endpoints have downloaded their apps from the sam=
e server, they can do trickle ICE any way that they want.
2) in the case where an application is  talking to an unknown or legacy pee=
r, trickle  ICE is a bad idea because there is no standard way to do it. (i=
.e. try it at your own risk)

Is this correct?

- Jim=20





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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 8/27/12 2:42 PM, Matthew Kaufman wrote:
> The "two different web servers" "trapezoid" case would require a 
> trunking protocol between these servers, such as SIP or Jingle,
> which can describe "trickle ICE" such that the browsers could be
> instructed properly by their respective web servers.
> 
> I believe that with Jingle, there is a written spec for that, and
> for SIP, there is not (at present).

Correct: http://xmpp.org/extensions/xep-0176.html

/psa


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--_1caa062c-84c3-4923-9ffd-9d2939a669ad_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Right. =20

 XEP-0176 Sections 5.5-5.9 does describe how various aspects of "Trickle IC=
E" are handled in XMPP/Jingle=2C answering a number of the questions asked =
in Eric's original post (see http://www.ietf.org/mail-archive/web/rtcweb/cu=
rrent/msg05121.html).  These sections are consistent with RFC 5245 with res=
pect to STUN/TURN behavior=2C so that it isn't clear to me that a revision =
of RFC 5245 is needed.  Or am I missing something?

Peter St. Andre said:

>>  I believe that with Jingle=2C there is a written spec for that=2C and f=
or SIP=2C there is not (at present).
>=20
> Correct: http://xmpp.org/extensions/xep-0176.html
>=20
> /psa

Eric Rescorla said:

I've been working on doing Firefox's trickle ICE implementation and=0A=
have had to make a bunch of decisions that suggest to me that perhaps=0A=
we need an update to RFC 5245 to specify exactly how trickle ICE=0A=
behaves. Here are some of the things we need to resolve.....

 		 	   		  =

--_1caa062c-84c3-4923-9ffd-9d2939a669ad_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Right.&nbsp=3B <br><br>&nbsp=3BX=
EP-0176 Sections 5.5-5.9 does describe how various aspects of "Trickle ICE"=
 are handled in XMPP/Jingle=2C answering a number of the questions asked in=
 Eric's original post (see http://www.ietf.org/mail-archive/web/rtcweb/curr=
ent/msg05121.html).&nbsp=3B These sections are consistent with RFC 5245 wit=
h respect to STUN/TURN behavior=2C so that it isn't clear to me that a revi=
sion of RFC 5245 is needed.&nbsp=3B Or am I missing something?<br><br><div>=
Peter St. Andre said:<br><br>&gt=3B&gt=3B&nbsp=3B I believe that with Jingl=
e=2C there is a written spec for that=2C and for SIP=2C there is not (at pr=
esent).<br>&gt=3B <br>&gt=3B Correct: http://xmpp.org/extensions/xep-0176.h=
tml<br>&gt=3B <br>&gt=3B /psa<br><br>Eric Rescorla said:<br><br><pre>I've b=
een working on doing Firefox's trickle ICE implementation and=0A=
have had to make a bunch of decisions that suggest to me that perhaps=0A=
we need an update to RFC 5245 to specify exactly how trickle ICE=0A=
behaves. Here are some of the things we need to resolve.....</pre><br><br><=
/div> 		 	   		  </div></body>
</html>=

--_1caa062c-84c3-4923-9ffd-9d2939a669ad_--

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On 27.08.12, 23:51, Bernard Aboba wrote:
> Right. 
> 
>  XEP-0176 Sections 5.5-5.9 does describe how various aspects of "Trickle
> ICE" are handled in XMPP/Jingle, answering a number of the questions
> asked in Eric's original post (see
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg05121.html). 
> These sections are consistent with RFC 5245 with respect to STUN/TURN
> behavior, so that it isn't clear to me that a revision of RFC 5245 is
> needed.  Or am I missing something?

Well an ICE stack that implements trickle as per XEP-0176 would
currently not interoperate with a 5245 implementation or, at best, would
lead to unpredictable results.

The description in XEP-0176 is actually quite perfunctory and can't
really be considered a proper specification.

A document that describes a proper way of implementing this would hence
be quite helpful.

Emil

> 
> Peter St. Andre said:
> 
>>>  I believe that with Jingle, there is a written spec for that, and
> for SIP, there is not (at present).
>>
>> Correct: http://xmpp.org/extensions/xep-0176.html
>>
>> /psa
> 
> Eric Rescorla said:
> 
> I've been working on doing Firefox's trickle ICE implementation and
> have had to make a bunch of decisions that suggest to me that perhaps
> we need an update to RFC 5245 to specify exactly how trickle ICE
> behaves. Here are some of the things we need to resolve.....
> 
> 
> 
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 

-- 
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31

From bernard_aboba@hotmail.com  Mon Aug 27 15:29:12 2012
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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABkgnnVBBAH=HCkn_cksBs_9A_hm=VfFwcTtvOM3C7XB2h2KTA@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca><CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com>, <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7B56@tk5ex14mbxc272.redmond.corp.microsoft.com>, <E17CAD772E76C742B645BD4DC602CD81069D8500@NAHALD.us.int.genesyslab.com>, <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7C02@tk5ex14mbxc272.redmond.corp.microsoft.com>, <503BDC75.7050008@stpeter.im> <BLU002-W2286956624CC6600038246993A20@phx.gbl> <503BEEFD.40301@jitsi.org>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Emil Ivov said: 

Well an ICE stack that implements trickle as per XEP-0176 would currently
not interoperate with a 5245 implementation or, at best, would lead to
unpredictable results.

The description in XEP-0176 is actually quite perfunctory and can't really
be considered a proper specification.

A document that describes a proper way of implementing this would hence be
quite helpful.

[BA]  You are correct that XMPP/Jingle signaling will not be understood by a
SIP UA.  However,   I don't see anything inherent in the use of STUN/TURN 
within XEP-0176 that violates RFC 5245.  If the issue is lack of clarity in
XEP-0176, shouldn't that be brought up in the XSF, which owns the
specification?


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From: Francois Audet <francois.audet@skype.net>
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Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Wouldn't the same mechanism as XEP-0176 work with SIP too, by sending new o=
ffer/answers with new candidates?

In either cases, it is true that the "trapezoid" model makes this process m=
ore difficult than with the "triangle" model.

But you don't *have* to do trickle ICE don't you, in the trapezoid scenario=
 (especially when involving protocol mapping).

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: Monday, August 27, 2012 3:29 PM
To: 'Emil Ivov'
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

Emil Ivov said:=20

Well an ICE stack that implements trickle as per XEP-0176 would currently n=
ot interoperate with a 5245 implementation or, at best, would lead to unpre=
dictable results.

The description in XEP-0176 is actually quite perfunctory and can't really =
be considered a proper specification.

A document that describes a proper way of implementing this would hence be =
quite helpful.

[BA]  You are correct that XMPP/Jingle signaling will not be understood by =
a
SIP UA.  However,   I don't see anything inherent in the use of STUN/TURN=20
within XEP-0176 that violates RFC 5245.  If the issue is lack of clarity in=
 XEP-0176, shouldn't that be brought up in the XSF, which owns the specific=
ation?

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



From kaiduanx@gmail.com  Mon Aug 27 18:40:54 2012
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From: Kaiduan Xie <kaiduanx@gmail.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] Clarification on offer/answer in jsep-01
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Hi all,

I do not understand the statement below from 4.2. Session Descriptions
and State Machine

"As in [RFC3264], an offerer can send an offer, and update it as long
as it has not been answered."

However, per rfc3264 http://tools.ietf.org/html/rfc3264 section 4
Protocol Operation,

"At any time, either agent MAY generate a new offer that updates the
 session. However, it MUST NOT generate a new offer if it has
 received an offer which it has not yet answered or rejected.
 Furthermore, it MUST NOT generate a new offer if it has generated a
 prior offer for which it has not yet received an answer or a
 rejection."

Please look the last sentence. Can anyone explain why JSEP introduces
something different than RFC3264 please?

Best regards,

/Kaiduan

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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABkgnnVBBAH=HCkn_cksBs_9A_hm=VfFwcTtvOM3C7XB2h2KTA@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca><CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com>, , <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com>, , <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7B56@tk5ex14mbxc272.redmond.corp.microsoft.com>, , <E17CAD772E76C742B645BD4DC602CD81069D8500@NAHALD.us.int.genesyslab.com>, , <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7C02@tk5ex14mbxc272.redmond.corp.microsoft.com>, , <503BDC75.7050008@stpeter.im> <BLU002-W2286956624CC6600038246993A20@phx.gbl>, <503BEEFD.40301@jitsi.org> <BLU169-DS457B54FA311A048E5E68B093A20@phx.gbl>, <66F9C712A970F74A8376BBE53013193C0DD0C76C@tk5ex14mbxc272.redmond.corp.microsoft.com>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Francois said:=20

> Wouldn't the same mechanism as XEP-0176 work with SIP too=2C by sending n=
ew offer/answers with new candidates?
>=20
> In either cases=2C it is true that the "trapezoid" model makes this proce=
ss more difficult than with the "triangle" model.
>=20
> But you don't *have* to do trickle ICE don't you=2C in the trapezoid scen=
ario (especially when involving protocol mapping).

[BA]  Looking at XEP-0176 Section 5.8=2C it does seem possible to use new o=
ffer/answers with new candidates to similar effect.  Here is what Section 5=
.8 says:=20
=0A=

=0A=
Section 5.8 Negotiating a new candidate
=0A=

=0A=
Even after media has begun to flow=2C either party MAY continue to send =0A=
additional candidates to the other party (e.g.=2C because the user agent =
=0A=
has become aware of a new media proxy or network interface card). Such =0A=
candidates are shared by sending a transport-info message....  The =0A=
receiving party MUST acknowledge receipt of the candidate. The parties =0A=
would check the newly-offered candidate for =0A=
connectivity=2C as described previously. If the parties determine that =0A=
media can flow over the candidate=2C they MAY then use the new candidate =
=0A=
in subsequent communications. =20
=0A=

=0A=
[BA]  If ICE completed and media was already flowing=2C the controlling age=
nt can generate a RE-INVITE with a candidate from the in-use pair as the de=
fault as well as additional candidates.  There is no need to do an ICE rest=
art.  As described in RFC 5245 Section 8.1.2=2C if the additional candidate=
 ends up in a highest-priority nominated candidate-pair=2C then an updated =
offer is sent:=20


=0A=
      *  If an agent is controlling=2C it examines the highest-priority=0A=
         nominated candidate pair for each component of each media=0A=
         stream.  If any of those candidate pairs differ from the=0A=
         default candidate pairs in the most recent offer/answer=0A=
         exchange=2C the controlling agent MUST generate an updated offer=
=0A=
         as described in Section 9.  If the controlling agent is using=0A=
         an aggressive nomination algorithm=2C this may result in several=
=0A=
         updated offers as the pairs selected for media change.  An=0A=
         agent MAY delay sending the offer for a brief interval (one=0A=
         second is RECOMMENDED) in order to allow the selected pairs to=0A=
         stabilize.=0A=

Details of the updated offer are described in RFC 5245 Section 9.1.2.2:
=0A=
=0A=
   If an agent generates an updated offer including a media stream that=0A=
   was previously established=2C and for which ICE checks are in the=0A=
   Completed state=2C the agent follows the procedures defined here.=0A=
=0A=
   The default destination for media (i.e.=2C the values of the IP=0A=
   addresses and ports in the m and c lines used for that media stream)=0A=
   MUST be the local candidate from the highest-priority nominated pair=0A=
   in the valid list for each component.  This "fixes" the default=0A=
   destination for media to equal the destination ICE has selected for=0A=
   media.=0A=
=0A=
   The agent MUST include candidate attributes for candidates matching=0A=
   the default destination for each component of the media stream=2C and=0A=
   MUST NOT include any other candidates.=0A=
=0A=
   In addition=2C if the agent is controlling=2C it MUST include the=0A=
   a=3Dremote-candidates attribute for each media stream whose check list=
=0A=
   is in the Completed state.  The attribute contains the remote=0A=
   candidates from the highest-priority nominated pair in the valid list=0A=
   for each component of that media stream.  It is needed to avoid a=0A=
   race condition whereby the controlling agent chooses its pairs=2C but=0A=
   the updated offer beats the connectivity checks to the controlled=0A=
   agent=2C which doesn't even know these pairs are valid=2C let alone=0A=
   selected.  See Appendix B.6 for elaboration on this race condition.
 		 	   		  =

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<html>
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<body class=3D'hmmessage'><div dir=3D'ltr'>Francois said: <br><br><div>&gt=
=3B Wouldn't the same mechanism as XEP-0176 work with SIP too=2C by sending=
 new offer/answers with new candidates?<br>&gt=3B <br>&gt=3B In either case=
s=2C it is true that the "trapezoid" model makes this process more difficul=
t than with the "triangle" model.<br>&gt=3B <br>&gt=3B But you don't *have*=
 to do trickle ICE don't you=2C in the trapezoid scenario (especially when =
involving protocol mapping).<br><br>[BA]&nbsp=3B Looking at XEP-0176 Sectio=
n 5.8=2C it does seem possible to use new offer/answers with new candidates=
 to similar effect.&nbsp=3B Here is what Section 5.8 says: <br>=0A=
<br>=0A=
Section 5.8 Negotiating a new candidate<br>=0A=
<br>=0A=
Even after media has begun to flow=2C either party MAY continue to send =0A=
additional candidates to the other party (e.g.=2C because the user agent =
=0A=
has become aware of a new media proxy or network interface card). Such =0A=
candidates are shared by sending a transport-info message....&nbsp=3B The =
=0A=
receiving party MUST acknowledge receipt of the candidate. The parties =0A=
would check the newly-offered candidate for =0A=
connectivity=2C as described previously. If the parties determine that =0A=
media can flow over the candidate=2C they MAY then use the new candidate =
=0A=
in subsequent communications.&nbsp=3B <br>=0A=
<br>=0A=
[BA]&nbsp=3B If ICE completed and media was already flowing=2C the controll=
ing agent can generate a RE-INVITE with a candidate from the in-use pair as=
 the default as well as additional candidates.&nbsp=3B There is no need to =
do an ICE restart.&nbsp=3B As described in RFC 5245 Section 8.1.2=2C if the=
 additional candidate ends up in a highest-priority nominated candidate-pai=
r=2C then an updated offer is sent: <br><br><br>=0A=
<pre>      *  If an agent is controlling=2C it examines the highest-priorit=
y=0A=
         nominated candidate pair for each component of each media=0A=
         stream.  If any of those candidate pairs differ from the=0A=
         default candidate pairs in the most recent offer/answer=0A=
         exchange=2C the controlling agent MUST generate an updated offer=
=0A=
         as described in Section 9.  If the controlling agent is using=0A=
         an aggressive nomination algorithm=2C this may result in several=
=0A=
         updated offers as the pairs selected for media change.  An=0A=
         agent MAY delay sending the offer for a brief interval (one=0A=
         second is RECOMMENDED) in order to allow the selected pairs to=0A=
         stabilize.</pre>=0A=
<br>Details of the updated offer are described in RFC 5245 Section 9.1.2.2:=
<br><pre>=0A=
=0A=
   If an agent generates an updated offer including a media stream that=0A=
   was previously established=2C and for which ICE checks are in the=0A=
   Completed state=2C the agent follows the procedures defined here.=0A=
=0A=
   The default destination for media (i.e.=2C the values of the IP=0A=
   addresses and ports in the m and c lines used for that media stream)=0A=
   MUST be the local candidate from the highest-priority nominated pair=0A=
   in the valid list for each component.  This "fixes" the default=0A=
   destination for media to equal the destination ICE has selected for=0A=
   media.=0A=
=0A=
   The agent MUST include candidate attributes for candidates matching=0A=
   the default destination for each component of the media stream=2C and=0A=
   MUST NOT include any other candidates.=0A=
=0A=
   In addition=2C if the agent is controlling=2C it MUST include the=0A=
   a=3Dremote-candidates attribute for each media stream whose check list=
=0A=
   is in the Completed state.  The attribute contains the remote=0A=
   candidates from the highest-priority nominated pair in the valid list=0A=
   for each component of that media stream.  It is needed to avoid a=0A=
   race condition whereby the controlling agent chooses its pairs=2C but=0A=
   the updated offer beats the connectivity checks to the controlled=0A=
   agent=2C which doesn't even know these pairs are valid=2C let alone=0A=
   selected.  See Appendix B.6 for elaboration on this race condition.</pre=
><br></div> 		 	   		  </div></body>
</html>=

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From francois.audet@skype.net  Mon Aug 27 21:40:54 2012
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From: Francois Audet <francois.audet@skype.net>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Date: Tue, 28 Aug 2012 04:40:47 +0000
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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABkgnnVBBAH=HCkn_cksBs_9A_hm=VfFwcTtvOM3C7XB2h2KTA@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca><CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com>, , <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com>, , <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7B56@tk5ex14mbxc272.redmond.corp.microsoft.com>, , <E17CAD772E76C742B645BD4DC602CD81069D8500@NAHALD.us.int.genesyslab.com>, , <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7C02@tk5ex14mbxc272.redmond.corp.microsoft.com>, , <503BDC75.7050008@stpeter.im> <BLU002-W2286956624CC6600038246993A20@phx.gbl>, <503BEEFD.40301@jitsi.org> <BLU169-DS457B54FA311A048E5E68B093A20@phx.gbl>, <66F9C712A970F74A8376BBE53013193C0DD0C76C@tk5ex14mbxc272.redmond.corp.microsoft.com>, <BLU002-W161FE7AE76153E3E8A6A81293A10@phx.gbl>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Yes, this is exactly the section of 5245 I was referring to. Seems to be ef=
fectively trickling.

On Aug 27, 2012, at 20:12, "Bernard Aboba" <bernard_aboba@hotmail.com<mailt=
o:bernard_aboba@hotmail.com>> wrote:

Francois said:

> Wouldn't the same mechanism as XEP-0176 work with SIP too, by sending new=
 offer/answers with new candidates?
>
> In either cases, it is true that the "trapezoid" model makes this process=
 more difficult than with the "triangle" model.
>
> But you don't *have* to do trickle ICE don't you, in the trapezoid scenar=
io (especially when involving protocol mapping).

[BA]  Looking at XEP-0176 Section 5.8, it does seem possible to use new off=
er/answers with new candidates to similar effect.  Here is what Section 5.8=
 says:

Section 5.8 Negotiating a new candidate

Even after media has begun to flow, either party MAY continue to send addit=
ional candidates to the other party (e.g., because the user agent has becom=
e aware of a new media proxy or network interface card). Such candidates ar=
e shared by sending a transport-info message....  The receiving party MUST =
acknowledge receipt of the candidate. The parties would check the newly-off=
ered candidate for connectivity, as described previously. If the parties de=
termine that media can flow over the candidate, they MAY then use the new c=
andidate in subsequent communications.

[BA]  If ICE completed and media was already flowing, the controlling agent=
 can generate a RE-INVITE with a candidate from the in-use pair as the defa=
ult as well as additional candidates.  There is no need to do an ICE restar=
t.  As described in RFC 5245 Section 8.1.2, if the additional candidate end=
s up in a highest-priority nominated candidate-pair, then an updated offer =
is sent:



      *  If an agent is controlling, it examines the highest-priority
         nominated candidate pair for each component of each media
         stream.  If any of those candidate pairs differ from the
         default candidate pairs in the most recent offer/answer
         exchange, the controlling agent MUST generate an updated offer
         as described in Section 9.  If the controlling agent is using
         an aggressive nomination algorithm, this may result in several
         updated offers as the pairs selected for media change.  An
         agent MAY delay sending the offer for a brief interval (one
         second is RECOMMENDED) in order to allow the selected pairs to
         stabilize.

Details of the updated offer are described in RFC 5245 Section 9.1.2.2:


   If an agent generates an updated offer including a media stream that
   was previously established, and for which ICE checks are in the
   Completed state, the agent follows the procedures defined here.

   The default destination for media (i.e., the values of the IP
   addresses and ports in the m and c lines used for that media stream)
   MUST be the local candidate from the highest-priority nominated pair
   in the valid list for each component.  This "fixes" the default
   destination for media to equal the destination ICE has selected for
   media.

   The agent MUST include candidate attributes for candidates matching
   the default destination for each component of the media stream, and
   MUST NOT include any other candidates.

   In addition, if the agent is controlling, it MUST include the
   a=3Dremote-candidates attribute for each media stream whose check list
   is in the Completed state.  The attribute contains the remote
   candidates from the highest-priority nominated pair in the valid list
   for each component of that media stream.  It is needed to avoid a
   race condition whereby the controlling agent chooses its pairs, but
   the updated offer beats the connectivity checks to the controlled
   agent, which doesn't even know these pairs are valid, let alone
   selected.  See Appendix B.6 for elaboration on this race condition.


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<div>Yes, this is exactly the section of 5245 I was referring to. Seems to =
be effectively trickling.<br>
<br>
On Aug 27, 2012, at 20:12, &quot;Bernard Aboba&quot; &lt;<a href=3D"mailto:=
bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
<br>
</div>
<div></div>
<blockquote type=3D"cite">
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<div dir=3D"ltr">Francois said: <br>
<br>
<div>&gt; Wouldn't the same mechanism as XEP-0176 work with SIP too, by sen=
ding new offer/answers with new candidates?<br>
&gt; <br>
&gt; In either cases, it is true that the &quot;trapezoid&quot; model makes=
 this process more difficult than with the &quot;triangle&quot; model.<br>
&gt; <br>
&gt; But you don't *have* to do trickle ICE don't you, in the trapezoid sce=
nario (especially when involving protocol mapping).<br>
<br>
[BA]&nbsp; Looking at XEP-0176 Section 5.8, it does seem possible to use ne=
w offer/answers with new candidates to similar effect.&nbsp; Here is what S=
ection 5.8 says:
<br>
<br>
Section 5.8 Negotiating a new candidate<br>
<br>
Even after media has begun to flow, either party MAY continue to send addit=
ional candidates to the other party (e.g., because the user agent has becom=
e aware of a new media proxy or network interface card). Such candidates ar=
e shared by sending a transport-info
 message....&nbsp; The receiving party MUST acknowledge receipt of the cand=
idate. The parties would check the newly-offered candidate for connectivity=
, as described previously. If the parties determine that media can flow ove=
r the candidate, they MAY then use the
 new candidate in subsequent communications.&nbsp; <br>
<br>
[BA]&nbsp; If ICE completed and media was already flowing, the controlling =
agent can generate a RE-INVITE with a candidate from the in-use pair as the=
 default as well as additional candidates.&nbsp; There is no need to do an =
ICE restart.&nbsp; As described in RFC 5245 Section
 8.1.2, if the additional candidate ends up in a highest-priority nominated=
 candidate-pair, then an updated offer is sent:
<br>
<br>
<br>
<pre>      *  If an agent is controlling, it examines the highest-priority
         nominated candidate pair for each component of each media
         stream.  If any of those candidate pairs differ from the
         default candidate pairs in the most recent offer/answer
         exchange, the controlling agent MUST generate an updated offer
         as described in Section 9.  If the controlling agent is using
         an aggressive nomination algorithm, this may result in several
         updated offers as the pairs selected for media change.  An
         agent MAY delay sending the offer for a brief interval (one
         second is RECOMMENDED) in order to allow the selected pairs to
         stabilize.</pre>
<br>
Details of the updated offer are described in RFC 5245 Section 9.1.2.2:<br>
<pre>
   If an agent generates an updated offer including a media stream that
   was previously established, and for which ICE checks are in the
   Completed state, the agent follows the procedures defined here.

   The default destination for media (i.e., the values of the IP
   addresses and ports in the m and c lines used for that media stream)
   MUST be the local candidate from the highest-priority nominated pair
   in the valid list for each component.  This &quot;fixes&quot; the defaul=
t
   destination for media to equal the destination ICE has selected for
   media.

   The agent MUST include candidate attributes for candidates matching
   the default destination for each component of the media stream, and
   MUST NOT include any other candidates.

   In addition, if the agent is controlling, it MUST include the
   a=3Dremote-candidates attribute for each media stream whose check list
   is in the Completed state.  The attribute contains the remote
   candidates from the highest-priority nominated pair in the valid list
   for each component of that media stream.  It is needed to avoid a
   race condition whereby the controlling agent chooses its pairs, but
   the updated offer beats the connectivity checks to the controlled
   agent, which doesn't even know these pairs are valid, let alone
   selected.  See Appendix B.6 for elaboration on this race condition.</pre=
>
<br>
</div>
</div>
</div>
</blockquote>
</body>
</html>

--_000_DDFF5412966F47C8B7773C57E96AD689skypenet_--

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From: Francois Audet <francois.audet@skype.net>
To: Kaiduan Xie <kaiduanx@gmail.com>
Thread-Topic: [rtcweb] Clarification on offer/answer in jsep-01
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Indeed, this appear to contradict 3264, and would cause breakage.

On Aug 27, 2012, at 18:41, "Kaiduan Xie" <kaiduanx@gmail.com> wrote:

> Hi all,
>=20
> I do not understand the statement below from 4.2. Session Descriptions
> and State Machine
>=20
> "As in [RFC3264], an offerer can send an offer, and update it as long
> as it has not been answered."
>=20
> However, per rfc3264 http://tools.ietf.org/html/rfc3264 section 4
> Protocol Operation,
>=20
> "At any time, either agent MAY generate a new offer that updates the
> session. However, it MUST NOT generate a new offer if it has
> received an offer which it has not yet answered or rejected.
> Furthermore, it MUST NOT generate a new offer if it has generated a
> prior offer for which it has not yet received an answer or a
> rejection."
>=20
> Please look the last sentence. Can anyone explain why JSEP introduces
> something different than RFC3264 please?
>=20
> Best regards,
>=20
> /Kaiduan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 28.08.12, 00:29, Bernard Aboba wrote:
> Emil Ivov said: 
> 
>> Well an ICE stack that implements trickle as per XEP-0176 would currently
>> not interoperate with a 5245 implementation or, at best, would lead to
>> unpredictable results.
>> 
>> The description in XEP-0176 is actually quite perfunctory and can't really
>> be considered a proper specification.
>> 
>> A document that describes a proper way of implementing this would hence be
>> quite helpful.
> 
> [BA]  You are correct that XMPP/Jingle signaling will not be understood by a
> SIP UA. 

Right. And of course, I was not referring to signalling.

> However,   I don't see anything inherent in the use of STUN/TURN 
> within XEP-0176 that violates RFC 5245. 

It's more about ICE processing rather than STUN and TURN and the fact
that, if an ICE implementation is not prepared to handle trickling it
would not work well with one that does. (more below)

>From your other mail:

> Details of the updated offer are described in RFC 5245 Section 9.1.2.2:

I am not sure how 9.1.2.2 would apply here given that it's about sending
updated offers once ICE has completed. It even says that:

    The agent MUST include candidate attributes for candidates matching
    the default destination for each component of the media stream, and
    MUST NOT include any other candidates.

Did you instead mean to quote section 9.1.2.1 which contains the following:

   The agent
   MAY include additional candidates it did not offer previously, but
   which it has gathered since the last offer/answer exchange, including
   peer reflexive candidates.

9.1.2.1 does indeed refer to updating ICE for "Existing Media Streams
with ICE Running".

While I agree that this does look like trickling I am not sure it was
intended to be used in exactly the same way that we are discussing here
and the text is not enough to cover a proper trickling implementation.

The main issue is that there's currently nothing in 5245 that would
allow an agent to determine if more candidates are to be expected. If
trickling is in use there's no way for a controlled agent to know that
(or if) more candidates are to be expected. This creates a race
condition which may result in the controlled agent moving into the "ICE
Failed" state before it has received all candidates from the controlling
agent.

Such premature failures can happen very easily if the first offer only
contains addresses that can be determined as unusable without even
running any checks. This would be the case for an IPv4-only host
receiving an offer with IPv6 candidates only (and vice versa). One can
probably come up with a few more cases like this one.

> If the issue is lack of clarity in
> XEP-0176, shouldn't that be brought up in the XSF, which owns the
> specification?

Nope, XEP-0176 would be crystal clear if the IETF comes up with a
document that explains how ICE agents should behave in situations where
addresses continue to arrive after the original offer.

Emil

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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 28.08.12, 08:04, Emil Ivov wrote:
> The main issue is that there's currently nothing in 5245 that would
> allow an agent to determine if more candidates are to be expected. If
> trickling is in use there's no way for a controlled agent to know that
> (or if) more candidates are to be expected. This creates a race
> condition which may result in the controlled agent moving into the "ICE
> Failed" state before it has received all candidates from the controlling
> agent.

I just noticed that this is also what Eric's original mail was about, so
I suppose I just second his concern. We had the issue in ice4j and we
basically just had to implement and tune it in a way that would work
against other common ICE impls.


Text that specifies exactly how this is to be done would have been most
helpful.

Emil

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From srikar.rao@wipro.com  Tue Aug 28 04:42:54 2012
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From: <srikar.rao@wipro.com>
To: <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Clarification on offer/answer in jsep-01
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Date: Tue, 28 Aug 2012 11:37:25 +0000
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Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
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> "As in [RFC3264], an offerer can send an offer, and update it as long
> as it has not been answered."

That statement doesn't comply to [RFC3264] on the same level as PRANSWER do=
esn't. PRANSWER may not directly fit in the offer-answer pair model of [RFC=
3264] as indicated in <http://www.ietf.org/mail-archive/web/rtcweb/current/=
msg04675.html>. But it is considered just an API construct <http://www.ietf=
.org/mail-archive/web/rtcweb/current/msg04830.html> to feed-in evolving ans=
wers (probably from different sources) to local media elements. =20

Similarly, Offer-updated-before-Answer may be meant as evolving-offer(Can't=
 imagine a use-case except new ICE candidates).
The statement definitely needs clarification, specially because it starts w=
ith "As in [RFC3264]...".

In general, it looks like JSEP-01 picks up rules pertaining to construction=
 of Offers/Answers from [RFC3264] but is not exactly in-line with the O/A e=
xchange part of [RFC3264].

Regards,
Srikar.

------------------------------
--Original Message--
Message: 2
Date: Tue, 28 Aug 2012 04:48:07 +0000
From: Francois Audet <francois.audet@skype.net>
To: Kaiduan Xie <kaiduanx@gmail.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
Message-ID: <A9BAC738-4077-450F-ACC4-DD246292A2EF@microsoft.com>
Content-Type: text/plain; charset=3D"us-ascii"

Indeed, this appear to contradict 3264, and would cause breakage.

On Aug 27, 2012, at 18:41, "Kaiduan Xie" <kaiduanx@gmail.com> wrote:

> Hi all,
>
> I do not understand the statement below from 4.2. Session Descriptions
> and State Machine
>
> "As in [RFC3264], an offerer can send an offer, and update it as long
> as it has not been answered."
>
> However, per rfc3264 http://tools.ietf.org/html/rfc3264 section 4
> Protocol Operation,
>
> "At any time, either agent MAY generate a new offer that updates the
> session. However, it MUST NOT generate a new offer if it has
> received an offer which it has not yet answered or rejected.
> Furthermore, it MUST NOT generate a new offer if it has generated a
> prior offer for which it has not yet received an answer or a
> rejection."
>
> Please look the last sentence. Can anyone explain why JSEP introduces
> something different than RFC3264 please?
>
> Best regards,
>
> /Kaiduan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



------------------------------


From stefan.lk.hakansson@ericsson.com  Tue Aug 28 05:36:41 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
>
> At the last meeting we took a hum on selecting Opus and G.711 as the
> mediatory to implement audio codecs. If there is any new opinions
> please send them to the list by August 30th, after which the chairs
> will make a determination of consensus.

As much as I would like to say "I agree", I come to the conclusion it is 
not that simple.

I have two things that concern me:

* "Opus" not well enough defined
* Opus not mature and proven enough (this is the more important concern)

Lets talk about both of these.

"Opus" not well enough defined
==============================
Just saying "Opus" is a bit vague for me (who has not been in the codec
WG). Is a certain version meant? Which one? Or is it a moving target
(i.e. always implement the latest version)?

What is included (is jitter management, loss concealment part of Opus or 
something that is up to the implementation)?

And how is it judged what is a compliant implementation? Is there a ref
implementation that you test against? How do you test? Are there
test-vectors that must be conformed to? Or is "Opus" standardized as a
wire format (meaning that any implementation that can produce/decode
that wire format is compliant)? If the latter is true, implementers have 
more wiggle room to e.g. reduce complexity or design around IPRs
(submarine IPRs may surface). Are implementors required to support all
aspects of Opus, or are there sub-sets/profiles that would be sufficient 
(perhaps for certain classes of devices)? [1] suggests that not all 
ptime options need to be supported, but are there others things?

Many of the answers are probably available, but I think they should be
put in front of the WG before making this decision.

Opus not mature and proven enough
=================================
As far as I understand, Opus is a brand new standard. There are
implementations on-going, but yet no deployment, no experience from real 
world use (and definitely not from use in larger scale over a wide range 
of conditions, involving a broad range of end user devices). In
addition, I've been told that so far Opus has gone through less
characterization tests than codecs usually are put through.

I don't think it is fair to mandate the implementation of a codec (which 
is a large and complex piece) under these circumstances. Some real 
deployment (involving many device types) and experience (including
operation in different conditions) should be gained before doing so. It
could also be argued that there is a bigger risk from a licensing
perspective to mandate a codec that has not been in use.


Do I mean we should mandate just G.711?
=======================================
No, not necessarily. Looking ahead, I am convinced that new and improved 
codecs for audio and video will be introduced as they become available. 
There is a logic in adopting technology that reduces the transmitted 
number of bits and enhances the end user experience, and the WG is 
designing a solution that enables new codecs to be introduced and used 
(and I really hope they can be made use of even if the applications is 
untouched) IIUC. However, if G.711 would be the only MTI audio codec, 
services that expect audio BW that goes beyond narrowband could suffer 
since there is no guarantee that there is any other codec than G.711 
supported by both endpoints.

In addition G.711 is not that well suited for for challenging network 
conditions (the most important parts like packet loss concealment and 
jitter management depends on the implementation - or are we going to 
standardize those parts?; what I am after is that it is unable to lower 
the send rate as reaction to congestion - the only method available is 
to use longer frames to make the IP header overhead lower).

I see three possible ways forward:

a) Mandate G.711 only, and let the market decide if there will be a
universally available codec giving higher quality.

b) Postpone the decision for some time. Let Opus (and other new codecs
if such are introduced) get deployed, tested, perhaps improved. The
drawback with this approach is that the selection is delayed. If this
approach is chosen, I would also recommend that the WG spends some time
on selection criteria. In the presentation given in Vancouver ([2]) six
criteria were used (three of them considered more important). I think
they make sense, but can't remember we actually discussed them (or what
other criteria that may make sense).

c) Mandate some other codec (that is already deployed and in use).

Of these approaches, perhaps b) is the most reasonable one.

Stefan

[1] http://datatracker.ietf.org/doc/draft-cbran-rtcweb-codec/?include_text=1
[2] http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-6.pdf




>
> Thanks, Cullen
>
> Please note that the following IPR disclosure have been made on these
> codecs. They can be found at
>
> http://datatracker.ietf.org/ipr/
>
>
> 2010-11-07 • ID # 1445 "Broadcom Corporation's Statement about IPR
> related to draft-ietf-codec-opus-00 and
> draft-ietf-codec-description-00 (1)" 2010-11-07 • ID # 1446 "Xiph.Org
> Foundation's Statement about IPR related to
> draft-ietf-codec-opus-00" 2010-11-12 • ID # 1447 "Broadcom
> Corporation's Statement about IPR related to draft-ietf-codec-opus-00
> and draft-ietf-codec-description-00 (2)" 2011-03-23 • ID # 1520
> "Qualcomm Incorporated's Statement about IPR related to
> draft-ietf-codec-opus-05" 2011-03-27 • ID # 1524 "Xiph.Org
> Foundation's Statement about IPR related to
> draft-ietf-codec-opus-05" 2011-03-29 • ID # 1526 "Broadcom
> Corporation's Statement about IPR related to
> draft-ietf-codec-opus-05" 2011-03-29 • ID # 1525 "Skype Limited's
> Statement about IPR related to draft-ietf-codec-opus-05" 2011-07-23 •
> ID # 1602 "Skype Limited's Statement about IPR related to
> draft-ietf-codec-opus-07" 2012-01-25 • ID # 1670 "Microsoft
> Corporation's Statement about IPR related to
> draft-ietf-codec-opus-10" 2012-03-12 • ID # 1712 "Huawei Technologies
> Co.,Ltd's Statement about IPR related to draft-ietf-codec-opus-11
> (1)" 2012-04-02 • ID # 1741 "Huawei Technologies Co.,Ltd's Statement
> about IPR related to draft-ietf-codec-opus-11 (2)"
>
>
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>


From fluffy@iii.ca  Tue Aug 28 05:57:59 2012
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On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net> =
wrote:

>> "As in [RFC3264], an offerer can send an offer, and update it as long
>> as it has not been answered."

yah that won't work or all the reasons mentioned before and in my =
opinion does not represent the WG consensus. BTW - that was added in the =
-01 version of the draft and submitted without me ever seeing it - I'd =
be happy to fix it but I don't have the source for the draft. Chairs?




From fluffy@cisco.com  Tue Aug 28 06:05:19 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On Aug 27, 2012, at 4:29 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrot=
e:

> Emil Ivov said:=20
>=20
> Well an ICE stack that implements trickle as per XEP-0176 would currently
> not interoperate with a 5245 implementation or, at best, would lead to
> unpredictable results.
>=20
> The description in XEP-0176 is actually quite perfunctory and can't reall=
y
> be considered a proper specification.
>=20
> A document that describes a proper way of implementing this would hence b=
e
> quite helpful.
>=20
> [BA]  You are correct that XMPP/Jingle signaling will not be understood b=
y a
> SIP UA.  However,   I don't see anything inherent in the use of STUN/TURN=
=20
> within XEP-0176 that violates RFC 5245.  If the issue is lack of clarity =
in
> XEP-0176, shouldn't that be brought up in the XSF, which owns the
> specification?
>=20

XEP-0176 mostly tells you how to transport the trickle candidates. It does =
not answer the questions of ICE works with trickle candidates and there are=
 a bunch of them as raised in EKR's email and others. Note these are not ne=
w concerns - many of these were discussed with Rohan back when he was writi=
ng the Jingle stuff. We just never sorted them out. =

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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Cullen Jennings <fluffy@iii.ca>, Francois Audet <francois.audet@skype.net>
Thread-Topic: [rtcweb] Clarification on offer/answer in jsep-01
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Date: Tue, 28 Aug 2012 13:47:14 +0000
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Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
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Isn't this all moot now, given that the API discussion has moved to the W3C=
 WG?

Matthew Kaufman

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings
Sent: Tuesday, August 28, 2012 5:58 AM
To: Francois Audet
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01


On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net> wro=
te:

>> "As in [RFC3264], an offerer can send an offer, and update it as long=20
>> as it has not been answered."

yah that won't work or all the reasons mentioned before and in my opinion d=
oes not represent the WG consensus. BTW - that was added in the -01 version=
 of the draft and submitted without me ever seeing it - I'd be happy to fix=
 it but I don't have the source for the draft. Chairs?



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



From jmvalin@mozilla.com  Tue Aug 28 09:30:33 2012
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 08/28/2012 08:36 AM, Stefan Hakansson LK wrote:
> "Opus" not well enough defined ============================== Just
> saying "Opus" is a bit vague for me (who has not been in the codec 
> WG). Is a certain version meant? Which one? Or is it a moving
> target (i.e. always implement the latest version)?

The exact implementation (or version of that implementation) does not
matter as long as the implementation is compliant with (upcoming) RFC
6716.

> What is included (is jitter management, loss concealment part of
> Opus or something that is up to the implementation)?

The jitter buffer is not part of Opus, nor is it part of any other
codec I know of. As for loss concealment, the Opus reference
implementation proposes one, but implementers can choose what they
want because there is no interoperability issue with
changing/improving loss concealment.

> And how is it judged what is a compliant implementation? Is there a
> ref implementation that you test against? How do you test? Are
> there test-vectors that must be conformed to? Or is "Opus"
> standardized as a wire format (meaning that any implementation that
> can produce/decode that wire format is compliant)? If the latter is
> true, implementers have more wiggle room to e.g. reduce complexity
> or design around IPRs (submarine IPRs may surface).

The text in the upcoming RFC 6716 is very clear that only the decoder
is standardized. This is similar to how codecs like MP3 and AAC are
defined. We also provide a set of test vectors, along with a program
that evaluates the decoded output to determine whether the decoder
under test is compliant.

> Opus not mature and proven enough 
> ================================= As far as I understand, Opus is a
> brand new standard. There are implementations on-going, but yet no
> deployment, no experience from real world use (and definitely not
> from use in larger scale over a wide range of conditions, involving
> a broad range of end user devices).

So far, Opus has been deployed in Firefox, GStreamer, FFMpeg,
Foobar2000, and a few other applications. This is far more deployment
than any ITU-T/MPEG codec has had by the time of standardization. Not
only that, but some of the underlying technology has been in use for
years now. The original SILK codec (not compatible with Opus but
having a lot in common) has been in Skype for about 3 years, while
CELT (again, not compatible but sharing a lot of technology) has been
deployed in many applications in the past 3 years, including Mumble,
which has about half a million users worldwide.

> In addition, I've been told that so far Opus has gone through less 
> characterization tests than codecs usually are put through.

When it comes to testing, I recommend reading
http://tools.ietf.org/html/draft-valin-codec-results-00 which
summarizes the listening tests that were done on Opus. Also, I
recommend reading
https://www.ietf.org/proceedings/82/slides/codec-4.pdf
This describes a whole series of tests we did that other SDOs
generally don't do on their codecs.

> I don't think it is fair to mandate the implementation of a codec
> (which is a large and complex piece) under these circumstances.

Well, compared to the rest of webrtc, I find that Opus is so small,
simple and well-defined :-)

Cheers,

	Jean-Marc
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From coverdale@sympatico.ca  Tue Aug 28 11:47:12 2012
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From: Paul Coverdale <coverdale@sympatico.ca>
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>> In addition, I've been told that so far Opus has gone through less
>> characterization tests than codecs usually are put through.
>
>When it comes to testing, I recommend reading
>http://tools.ietf.org/html/draft-valin-codec-results-00 which summarizes
>the listening tests that were done on Opus. Also, I recommend reading
>https://www.ietf.org/proceedings/82/slides/codec-4.pdf
>This describes a whole series of tests we did that other SDOs generally
>don't do on their codecs.
>

I think that Stefan was referring to the lack of formal, controlled,
characterization tests carried out on Opus, which typically ARE carried out
on codecs in other SDOs before approval. 


...Paul


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From: <Markus.Isomaki@nokia.com>
To: <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Confirmation of consensus on audio codecs
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Hi Stefan,

I very much share your views and would like to support especially this opti=
on:

>b) Postpone the decision for some time. Let Opus (and other new codecs if
>such are introduced) get deployed, tested, perhaps improved. The drawback
>with this approach is that the selection is delayed. If this approach is c=
hosen, I
>would also recommend that the WG spends some time on selection criteria.
>In the presentation given in Vancouver ([2]) six criteria were used (three=
 of
>them considered more important). I think they make sense, but can't
>remember we actually discussed them (or what other criteria that may make
>sense).

It is always nice to get decisions done early on. However, I don't think th=
at leaving the mandatory-to-implement codecs (for either WB audio or video)=
 open at this point will yet seriously impact the deployment or interoperab=
ility of WebRTC. There are still many other things to tackle. We should rev=
isit the codec issue when all the other critical things are in a good shape=
.=20

I'll post a separate mail on my views shortly.

Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Stefan Hakansson LK
>Sent: 28 August, 2012 15:37
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
>On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
>>
>> At the last meeting we took a hum on selecting Opus and G.711 as the
>> mediatory to implement audio codecs. If there is any new opinions
>> please send them to the list by August 30th, after which the chairs
>> will make a determination of consensus.
>
>As much as I would like to say "I agree", I come to the conclusion it is n=
ot that
>simple.
>
>I have two things that concern me:
>
>* "Opus" not well enough defined
>* Opus not mature and proven enough (this is the more important concern)
>
>Lets talk about both of these.
>
>"Opus" not well enough defined
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D
>Just saying "Opus" is a bit vague for me (who has not been in the codec WG=
).
>Is a certain version meant? Which one? Or is it a moving target (i.e. alwa=
ys
>implement the latest version)?
>
>What is included (is jitter management, loss concealment part of Opus or
>something that is up to the implementation)?
>
>And how is it judged what is a compliant implementation? Is there a ref
>implementation that you test against? How do you test? Are there test-
>vectors that must be conformed to? Or is "Opus" standardized as a wire
>format (meaning that any implementation that can produce/decode that wire
>format is compliant)? If the latter is true, implementers have more wiggle
>room to e.g. reduce complexity or design around IPRs (submarine IPRs may
>surface). Are implementors required to support all aspects of Opus, or are
>there sub-sets/profiles that would be sufficient (perhaps for certain clas=
ses of
>devices)? [1] suggests that not all ptime options need to be supported, bu=
t
>are there others things?
>
>Many of the answers are probably available, but I think they should be put=
 in
>front of the WG before making this decision.
>
>Opus not mature and proven enough
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D
>As far as I understand, Opus is a brand new standard. There are
>implementations on-going, but yet no deployment, no experience from real
>world use (and definitely not from use in larger scale over a wide range o=
f
>conditions, involving a broad range of end user devices). In addition, I'v=
e been
>told that so far Opus has gone through less characterization tests than co=
decs
>usually are put through.
>
>I don't think it is fair to mandate the implementation of a codec (which i=
s a
>large and complex piece) under these circumstances. Some real deployment
>(involving many device types) and experience (including operation in diffe=
rent
>conditions) should be gained before doing so. It could also be argued that
>there is a bigger risk from a licensing perspective to mandate a codec tha=
t has
>not been in use.
>
>
>Do I mean we should mandate just G.711?
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>No, not necessarily. Looking ahead, I am convinced that new and improved
>codecs for audio and video will be introduced as they become available.
>There is a logic in adopting technology that reduces the transmitted numbe=
r
>of bits and enhances the end user experience, and the WG is designing a
>solution that enables new codecs to be introduced and used (and I really h=
ope
>they can be made use of even if the applications is
>untouched) IIUC. However, if G.711 would be the only MTI audio codec,
>services that expect audio BW that goes beyond narrowband could suffer
>since there is no guarantee that there is any other codec than G.711
>supported by both endpoints.
>
>In addition G.711 is not that well suited for for challenging network cond=
itions
>(the most important parts like packet loss concealment and jitter
>management depends on the implementation - or are we going to
>standardize those parts?; what I am after is that it is unable to lower th=
e send
>rate as reaction to congestion - the only method available is to use longe=
r
>frames to make the IP header overhead lower).
>
>I see three possible ways forward:
>
>a) Mandate G.711 only, and let the market decide if there will be a univer=
sally
>available codec giving higher quality.
>
>b) Postpone the decision for some time. Let Opus (and other new codecs if
>such are introduced) get deployed, tested, perhaps improved. The drawback
>with this approach is that the selection is delayed. If this approach is c=
hosen, I
>would also recommend that the WG spends some time on selection criteria.
>In the presentation given in Vancouver ([2]) six criteria were used (three=
 of
>them considered more important). I think they make sense, but can't
>remember we actually discussed them (or what other criteria that may make
>sense).
>
>c) Mandate some other codec (that is already deployed and in use).
>
>Of these approaches, perhaps b) is the most reasonable one.
>
>Stefan
>
>[1] http://datatracker.ietf.org/doc/draft-cbran-rtcweb-
>codec/?include_text=3D1
>[2] http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-6.pdf
>
>
>
>
>>
>> Thanks, Cullen
>>
>> Please note that the following IPR disclosure have been made on these
>> codecs. They can be found at
>>
>> http://datatracker.ietf.org/ipr/
>>
>>
>> 2010-11-07 * ID # 1445 "Broadcom Corporation's Statement about IPR
>> related to draft-ietf-codec-opus-00 and
>> draft-ietf-codec-description-00 (1)" 2010-11-07 * ID # 1446 "Xiph.Org
>> Foundation's Statement about IPR related to draft-ietf-codec-opus-00"
>> 2010-11-12 * ID # 1447 "Broadcom Corporation's Statement about IPR
>> related to draft-ietf-codec-opus-00 and
>> draft-ietf-codec-description-00 (2)" 2011-03-23 * ID # 1520 "Qualcomm
>> Incorporated's Statement about IPR related to
>> draft-ietf-codec-opus-05" 2011-03-27 * ID # 1524 "Xiph.Org
>> Foundation's Statement about IPR related to draft-ietf-codec-opus-05"
>> 2011-03-29 * ID # 1526 "Broadcom Corporation's Statement about IPR
>> related to draft-ietf-codec-opus-05" 2011-03-29 * ID # 1525 "Skype
>> Limited's Statement about IPR related to draft-ietf-codec-opus-05"
>> 2011-07-23 * ID # 1602 "Skype Limited's Statement about IPR related to
>> draft-ietf-codec-opus-07" 2012-01-25 * ID # 1670 "Microsoft
>> Corporation's Statement about IPR related to draft-ietf-codec-opus-10"
>> 2012-03-12 * ID # 1712 "Huawei Technologies Co.,Ltd's Statement about
>> IPR related to draft-ietf-codec-opus-11 (1)" 2012-04-02 * ID # 1741
>> "Huawei Technologies Co.,Ltd's Statement about IPR related to
>> draft-ietf-codec-opus-11 (2)"
>>
>>
>>
>> _______________________________________________ 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 Markus.Isomaki@nokia.com  Tue Aug 28 12:26:37 2012
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From: <Markus.Isomaki@nokia.com>
To: <fluffy@cisco.com>, <rtcweb@ietf.org>
Thread-Topic: Confirmation of consensus on audio codecs
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Hi,

I'm having here both my individual thoughs and I'm also conveying the views=
 of Nokia in more general.=20

(as individual)

First of all, my feeling is that we may be trying to make the codec decisio=
ns pre-maturely. Codecs are after all a negotiable and well understood comp=
onent in RTCWeb. There are more critical things to get right to ensure inte=
roperability and success of RTCWeb. By my judgement it will take at least o=
ne more year to deliver them in IETF or W3C. So I believe we still have tim=
e to make the final call about codecs even for the "1.0" version. With new =
codecs such as Opus one more year may bring more information about its qual=
ity, IPR status and so on.

Many people in the WG seem to consider unencumberment of codecs as essentia=
l. Opus does have several non-RF declarations on it, and there are many pla=
yers who have not taken part in the IETF Codec WG and thus have no disclosu=
re obligation on their potential IPR. So if the WG wants to mandate a wideb=
and codec at this point, it seems to me that G.722 is the logical choice. (=
Opus is more efficient and if it indeed will enjoy RF status, I'm sure it w=
ill become the "de facto" RTCWeb codec in many environments.)

So, in summary, I think the WG should either not fix the mandatory codecs y=
et, or if that for some reason needs to be done right now, the MTI audio co=
decs should be G.711 and G.722. =20

(speaking for Nokia)

Nokia thinks that there are complexity issues with the Opus codec. There ar=
e also possible IPR risks associated with the "royalty-free" nature of Opus=
. We do not recommend Opus to be taken as a mandatory codec for RTCWeb at t=
his point. From high quality low bit rate and mobile applications point of =
view the 3GPP AMR-WB codec (also known as ITU-T G.722.2) is the most prefer=
able for us. For interoperability with implementations restricted to unencu=
mbered codecs, we prefer G.711 and G.722.=20

Regards,
	Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Cullen Jennings (fluffy)
>Sent: 16 August, 2012 20:16
>To: rtcweb@ietf.org
>Subject: [rtcweb] Confirmation of consensus on audio codecs
>
>
>At the last meeting we took a hum on selecting Opus and G.711 as the
>mediatory to implement audio codecs. If there is any new opinions please
>send them to the list by August 30th, after which the chairs will make a
>determination of consensus.
>
>Thanks,
>Cullen
>
>Please note that the following IPR disclosure have been made on these
>codecs. They can be found at
>
>http://datatracker.ietf.org/ipr/
>
>
>2010-11-07
>* ID # 1445
>"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-
>opus-00 and draft-ietf-codec-description-00 (1)"
>2010-11-07
>* ID # 1446
>"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opu=
s-
>00"
>2010-11-12
>* ID # 1447
>"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-
>opus-00 and draft-ietf-codec-description-00 (2)"
>2011-03-23
>* ID # 1520
>"Qualcomm Incorporated's Statement about IPR related to draft-ietf-codec-
>opus-05"
>2011-03-27
>* ID # 1524
>"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opu=
s-
>05"
>2011-03-29
>* ID # 1526
>"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-
>opus-05"
>2011-03-29
>* ID # 1525
>"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
>2011-07-23
>* ID # 1602
>"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
>2012-01-25
>* ID # 1670
>"Microsoft Corporation's Statement about IPR related to draft-ietf-codec-
>opus-10"
>2012-03-12
>* ID # 1712
>"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-
>codec-opus-11 (1)"
>2012-04-02
>* ID # 1741
>"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-
>codec-opus-11 (2)"
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Tue Aug 28 14:41:50 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On 08/28/2012 09:26 PM, Markus.Isomaki@nokia.com wrote:

> So, in summary, I think the WG should either not fix the mandatory
> codecs yet, or if that for some reason needs to be done right now,
> the MTI audio codecs should be G.711 and G.722.
I agree, there seem to be no reason why this must be decided right now, 
but if it must be G.711 and .722 are logical choices.

From lishitao@huawei.com  Tue Aug 28 18:26:14 2012
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From: Lishitao <lishitao@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Confirmation of consensus on audio codecs
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Hi all

Both as individual and HUAWEI's point of view,

We support this option, G.711 and G.722 (or G.711 only) are the best choice=
s to be MTI at this moment.

shitao


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Stefan Hakansson LK
> Sent: Wednesday, August 29, 2012 5:42 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>=20
> On 08/28/2012 09:26 PM, Markus.Isomaki@nokia.com wrote:
>=20
> > So, in summary, I think the WG should either not fix the mandatory
> > codecs yet, or if that for some reason needs to be done right now,
> > the MTI audio codecs should be G.711 and G.722.
> I agree, there seem to be no reason why this must be decided right now,
> but if it must be G.711 and .722 are logical choices.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From lishitao@huawei.com  Tue Aug 28 19:51:22 2012
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From: Lishitao <lishitao@huawei.com>
To: Cullen Jennings <fluffy@iii.ca>, Francois Audet <francois.audet@skype.net>
Thread-Topic: [rtcweb] Clarification on offer/answer in jsep-01
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Cullen Jennings
> Sent: Tuesday, August 28, 2012 8:58 PM
> To: Francois Audet
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
>=20
>=20
> On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net>
> wrote:
>=20
> >> "As in [RFC3264], an offerer can send an offer, and update it as long
> >> as it has not been answered."
>=20
> yah that won't work or all the reasons mentioned before and in my opinion=
 does
> not represent the WG consensus. BTW - that was added in the -01 version o=
f
> the draft and submitted without me ever seeing it - I'd be happy to fix i=
t but I
> don't have the source for the draft. Chairs?
>=20

The next sentence just after this one=20
" The answerer can send back zero or more
   provisional answers, and finally end the offer-answer exchange by
   sending a final answer." =20
does not describe quite well either, right?

AFAIK , the answerer can send back zero or more provisional answers, but th=
ey are not all SDP answers,=20

and I don't find any place in RFC 3264 has the definition about "final answ=
er".

shitao

From alan.b.johnston@gmail.com  Tue Aug 28 20:41:50 2012
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Subject: [rtcweb] RTCWEB needs an Internet Codec
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The RTCWEB effort needs an Internet codec.  This is why Opus is the
right choice.  RTCWEB also needs one codec for backwards compatibility
with the VoIP world.  This is why G.711 is also the right choice.

Any G.mumble codec is NOT an Internet codec and will not have the same
performance on the Internet as one that was designed for the Internet!
 If anyone doesn't understand what that means, go back and examine the
CODEC Working Group archives to get educated.

If someone wants another codec instead of Opus, then they need to
propose another Internet codec.  Otherwise, we are not serving the
Internet.

- Alan -

From roman@telurix.com  Tue Aug 28 21:13:58 2012
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--14dae9cfc88a85f07604c85fc9f8
Content-Type: text/plain; charset=ISO-8859-1

Not that I have anything against Opus, but what exactly makes Opus an
internet codec? What is internet codec anyway? Makes this all kind of a
meaningless argument.

I would argue that for my broadband internet connection G.722 is a perfect
internet codec. I do not care about bandwidth savings of Opus, and quality
wise, for the voice conversation, I cannot hear any difference.

I would argue G.711 should be the MTI codec. The rest can be left up to
browser implementers. We can argue all we want, but the best royalty free
low bitrate codec available will be the one everybody supports. We can
force it to be Opus, but even if we don't, it will still be Opus on its
merit alone. G.722 will probably end up being supported as well, since it
is free, pretty good quality, and easy to implement.

P.S. Not that I am arguing for it, I am surprised no one made a case for
iSAC, since it is also royalty free, low bit rate, and very high quality.
It is event named "internet Speech Audio Codec".
_____________
Roman Shpount


On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston
<alan.b.johnston@gmail.com>wrote:

> The RTCWEB effort needs an Internet codec.  This is why Opus is the
> right choice.  RTCWEB also needs one codec for backwards compatibility
> with the VoIP world.  This is why G.711 is also the right choice.
>
> Any G.mumble codec is NOT an Internet codec and will not have the same
> performance on the Internet as one that was designed for the Internet!
>  If anyone doesn't understand what that means, go back and examine the
> CODEC Working Group archives to get educated.
>
> If someone wants another codec instead of Opus, then they need to
> propose another Internet codec.  Otherwise, we are not serving the
> Internet.
>
> - Alan -
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--14dae9cfc88a85f07604c85fc9f8
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Not that I have anything against Opus, but what exactly makes Opus an inter=
net codec? What is internet codec anyway? Makes this all kind of a meaningl=
ess argument.<br><br>I would argue that for my broadband internet connectio=
n G.722 is a perfect internet codec. I do not care about bandwidth savings =
of Opus, and quality wise, for the voice conversation, I cannot hear any di=
fference.<br>
<br>I would argue G.711 should be the MTI codec. The rest can be left up to=
 browser implementers. We can argue all we want, but the best royalty free =
low bitrate codec available will be the one everybody supports. We can forc=
e it to be Opus, but even if we don&#39;t, it will still be Opus on its mer=
it alone. G.722 will probably end up being supported as well, since it is f=
ree, pretty good quality, and easy to implement.<br>
<br>P.S. Not that I am arguing for it, I am surprised no one made a case fo=
r iSAC, since it is also royalty free, low bit rate, and very high quality.=
 It is event named &quot;internet Speech Audio Codec&quot;.<br clear=3D"all=
">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Tue, Aug 28, 2012 at 11:41 PM, Alan J=
ohnston <span dir=3D"ltr">&lt;<a href=3D"mailto:alan.b.johnston@gmail.com" =
target=3D"_blank">alan.b.johnston@gmail.com</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">
The RTCWEB effort needs an Internet codec. =A0This is why Opus is the<br>
right choice. =A0RTCWEB also needs one codec for backwards compatibility<br=
>
with the VoIP world. =A0This is why G.711 is also the right choice.<br>
<br>
Any G.mumble codec is NOT an Internet codec and will not have the same<br>
performance on the Internet as one that was designed for the Internet!<br>
=A0If anyone doesn&#39;t understand what that means, go back and examine th=
e<br>
CODEC Working Group archives to get educated.<br>
<br>
If someone wants another codec instead of Opus, then they need to<br>
propose another Internet codec. =A0Otherwise, we are not serving the<br>
Internet.<br>
<br>
- Alan -<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>

--14dae9cfc88a85f07604c85fc9f8--

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29 aug 2012 kl. 03:24 skrev Lishitao <lishitao@huawei.com>:

> Hi all
>=20
> Both as individual and HUAWEI's point of view,
>=20
PLEASE. In the IETF we're all individuals. A company's point of view =
does not matter here.=20
I don't see the point of any more companies adding their "point of =
view".

Thanks.

/O

And that's Edvina's point of view too ;-)


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From: Lishitao <lishitao@huawei.com>
To: Emil Ivov <emcho@jitsi.org>, Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Date: Wed, 29 Aug 2012 06:34:48 +0000
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References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com><CABcZeBMFUFjU=FQo5LeJrcMfajeae0j+PWw5U2g5dUQNcJLWaA@mail.gmail.com><CABkgnnXiL3_U+Hci9ooDqBCsoV3KF8pwgcf9zbuN6EKZkK+aiQ@mail.gmail.com><CABcZeBNkkH93ybuMWoFg-ddKWnRgdn2Vgyb50W21A2GoMWxw6Q@mail.gmail.com><CABkgnnXQ25ZYNqeO+=FsYDR3aNvFS2zvrKWGs5o=h8m+Eq=Y+Q@mail.gmail.com><3B8DB12B-ABB3-4AC2-A0A0-93DC62C619D3@iii.ca><CABkgnnU3ecmhUwCYHmppwLJz-nbSA6=VRF7nF7wcpb+5QAWmdQ@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD81069D82BF@NAHALD.us.int.genesyslab.com>, <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7B56@tk5ex14mbxc272.redmond.corp.microsoft.com>, <E17CAD772E76C742B645BD4DC602CD81069D8500@NAHALD.us.int.genesyslab.com>, <AE1A6B5FD507DC4FB3C5166F3A05A4840E4E7C02@tk5ex14mbxc272.redmond.corp.microsoft.com>, <503BDC75.7050008@stpeter.im>	<BLU002-W2286956624CC6600038246993A20@phx.gbl> <503BEEFD.40301@jitsi.org>	<BLU169-DS457B54FA311A048E5E68B093A20@phx.gbl> <503C5F54.5000309@jitsi.org>
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> From your other mail:
>=20
> > Details of the updated offer are described in RFC 5245 Section 9.1.2.2:
>=20
> I am not sure how 9.1.2.2 would apply here given that it's about sending
> updated offers once ICE has completed. It even says that:
>=20
>     The agent MUST include candidate attributes for candidates matching
>     the default destination for each component of the media stream, and
>     MUST NOT include any other candidates.
>=20
> Did you instead mean to quote section 9.1.2.1 which contains the followin=
g:
>=20
>    The agent
>    MAY include additional candidates it did not offer previously, but
>    which it has gathered since the last offer/answer exchange, including
>    peer reflexive candidates.
>=20
> 9.1.2.1 does indeed refer to updating ICE for "Existing Media Streams
> with ICE Running".
>=20
> While I agree that this does look like trickling I am not sure it was
> intended to be used in exactly the same way that we are discussing here
> and the text is not enough to cover a proper trickling implementation.
>=20
> The main issue is that there's currently nothing in 5245 that would
> allow an agent to determine if more candidates are to be expected. If
> trickling is in use there's no way for a controlled agent to know that
> (or if) more candidates are to be expected. This creates a race
> condition which may result in the controlled agent moving into the "ICE
> Failed" state before it has received all candidates from the controlling
> agent.

Agree, and AFAIK, what described in RFC 5245 is not the exact mechanism as =
ICE trickling.=20

Moreover, using SDP for ICE trickling is not a good idea, because, the reas=
on by using ICE trickling is to shorten=20
the negotiation time, but the offerer can not send another update SDP offer=
 which including the new candidates=20
to the answerer unless the offerer has received the first SDP answer, this =
cannot increase the negotiation procedure.


> Such premature failures can happen very easily if the first offer only
> contains addresses that can be determined as unusable without even
> running any checks. This would be the case for an IPv4-only host
> receiving an offer with IPv6 candidates only (and vice versa). One can
> probably come up with a few more cases like this one.
>=20
> > If the issue is lack of clarity in
> > XEP-0176, shouldn't that be brought up in the XSF, which owns the
> > specification?
>=20
> Nope, XEP-0176 would be crystal clear if the IETF comes up with a
> document that explains how ICE agents should behave in situations where
> addresses continue to arrive after the original offer.


> Emil
>=20
> --
> https://jitsi.org
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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The text here is incorrect, this was a mistake on my part. While this usage
is technically permissible if you can deal with the complications it
introduces, it is not compliant with RFC 3264, as has been pointed out.

What I meant to say here was that the local description can be updated even
after an initial local description has been specified. There are real-world
use cases for this, such as setting an initial local description to tell
the ICE Agent the number of m= lines to gather candidates for, followed by
a later update when the getUserMedia MediaStream has been obtained. The
offer would not actually be sent until after the updated local description
has been appliced, so this would not violate 3264.


On Tue, Aug 28, 2012 at 5:57 AM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net>
> wrote:
>
> >> "As in [RFC3264], an offerer can send an offer, and update it as long
> >> as it has not been answered."
>
> yah that won't work or all the reasons mentioned before and in my opinion
> does not represent the WG consensus. BTW - that was added in the -01
> version of the draft and submitted without me ever seeing it - I'd be happy
> to fix it but I don't have the source for the draft. Chairs?
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--00235452fe2414d7de04c8628daa
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The text here is incorrect, this was a mistake on my part. While this usage=
 is technically permissible if you can deal with the complications it intro=
duces, it is not compliant with RFC 3264, as has been pointed out.<div><br>

</div><div>What I meant to say here was that the local description can be u=
pdated even after an initial local description has been specified. There ar=
e real-world use cases for this, such as setting an initial local descripti=
on to tell the ICE Agent the number of m=3D lines to gather candidates for,=
 followed by a later update when the getUserMedia MediaStream has been obta=
ined. The offer would not actually be sent until after the updated local de=
scription has been appliced, so this would not violate 3264.</div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Aug 2=
8, 2012 at 5:57 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto=
:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">

<div class=3D"im"><br>
On Aug 27, 2012, at 10:48 PM, Francois Audet &lt;<a href=3D"mailto:francois=
.audet@skype.net">francois.audet@skype.net</a>&gt; wrote:<br>
<br>
&gt;&gt; &quot;As in [RFC3264], an offerer can send an offer, and update it=
 as long<br>
&gt;&gt; as it has not been answered.&quot;<br>
<br>
</div>yah that won&#39;t work or all the reasons mentioned before and in my=
 opinion does not represent the WG consensus. BTW - that was added in the -=
01 version of the draft and submitted without me ever seeing it - I&#39;d b=
e happy to fix it but I don&#39;t have the source for the draft. Chairs?<br=
>


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

--00235452fe2414d7de04c8628daa--

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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On 08/28/2012 06:30 PM, Jean-Marc Valin wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 08/28/2012 08:36 AM, Stefan Hakansson LK wrote:
>> "Opus" not well enough defined ============================== Just
>> saying "Opus" is a bit vague for me (who has not been in the codec
>> WG). Is a certain version meant? Which one? Or is it a moving
>> target (i.e. always implement the latest version)?
>
> The exact implementation (or version of that implementation) does not
> matter as long as the implementation is compliant with (upcoming) RFC
> 6716.
>
>> What is included (is jitter management, loss concealment part of
>> Opus or something that is up to the implementation)?
>
> The jitter buffer is not part of Opus, nor is it part of any other
> codec I know of. As for loss concealment, the Opus reference
> implementation proposes one, but implementers can choose what they
> want because there is no interoperability issue with
> changing/improving loss concealment.
>
>> And how is it judged what is a compliant implementation? Is there a
>> ref implementation that you test against? How do you test? Are
>> there test-vectors that must be conformed to? Or is "Opus"
>> standardized as a wire format (meaning that any implementation that
>> can produce/decode that wire format is compliant)? If the latter is
>> true, implementers have more wiggle room to e.g. reduce complexity
>> or design around IPRs (submarine IPRs may surface).
>
> The text in the upcoming RFC 6716 is very clear that only the decoder
> is standardized. This is similar to how codecs like MP3 and AAC are
> defined. We also provide a set of test vectors, along with a program
> that evaluates the decoded output to determine whether the decoder
> under test is compliant.
Jean-Marc,

the info above goes a long way in answering my questions, thanks. I 
don't know the exact rules for IPR statements so I would like to double 
check: the fact that the RFC only defines the decoder, does that mean 
that there can be IPRs known by people in the codec WG that are not 
declared since they would apply to the encoder only (then there is the 
risk of IPRs held by those who have not participated in codec WG)?

Another question I had was if Opus is atomic, or if you can implement 
different profiles, and if so what parts should be MTI for rtcweb (and 
should the profile that is MTI be the same regardless of the device type 
- there has been issues raised regarding complexity?).

>
>> Opus not mature and proven enough
>> ================================= As far as I understand, Opus is a
>> brand new standard. There are implementations on-going, but yet no
>> deployment, no experience from real world use (and definitely not
>> from use in larger scale over a wide range of conditions, involving
>> a broad range of end user devices).
>
> So far, Opus has been deployed in Firefox, GStreamer, FFMpeg,
> Foobar2000, and a few other applications. This is far more deployment
> than any ITU-T/MPEG codec has had by the time of standardization.

That is great, but sort of underlines that there would be no harm in 
delaying the decision until there are experiences made from real world 
use - 'cause it would not be that long till that experience has been 
made (Markus also brought up the IPR status as a reason for waiting - I 
have no idea how long it would take to know more about that).

> Not
> only that, but some of the underlying technology has been in use for
> years now. The original SILK codec (not compatible with Opus but
> having a lot in common) has been in Skype for about 3 years, while
> CELT (again, not compatible but sharing a lot of technology) has been
> deployed in many applications in the past 3 years, including Mumble,
> which has about half a million users worldwide.
>
>> In addition, I've been told that so far Opus has gone through less
>> characterization tests than codecs usually are put through.
>
> When it comes to testing, I recommend reading
> http://tools.ietf.org/html/draft-valin-codec-results-00 which
> summarizes the listening tests that were done on Opus. Also, I
> recommend reading
> https://www.ietf.org/proceedings/82/slides/codec-4.pdf
> This describes a whole series of tests we did that other SDOs
> generally don't do on their codecs.

As Paul suggested, I was referring to the lack of formal, controlled,
characterization tests. That is how other SDOs do it. I don't think that 
is the only way to do it, but I think we should at least have either 
such tests conducted or experience from deployment and use (in a wide 
range of conditions and device types) before making it MTI.

>
>> I don't think it is fair to mandate the implementation of a codec
>> (which is a large and complex piece) under these circumstances.
>
> Well, compared to the rest of webrtc, I find that Opus is so small,
> simple and well-defined :-)

Especially if you draw a diagram with all the signaling paths, all 
components involved, and then make a small box for "codec" (or Opus) :-)

>
> Cheers,
>
> 	Jean-Marc
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>


From harald@alvestrand.no  Wed Aug 29 03:22:33 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Since the number of people stating their opinion has been large, I'll 
just reiterate the opinion I had (and hummed for) in Vancouver:

Opus and G.711 should be mandatory to implement for RTCWEB.

Since the question of whether there's any value to making the decision 
now has been raised:

The first interoperable products implementing RTCWEB are shipping within 
a very short timeframe. Those first implementations will shape the 
market for what's actually used in practice.
In order to allow applications requiring high quality sound to be among 
the first ones developed, those first products need to include a common 
choice of a high quality codec.

Having the RTCWEB WG state as a decision that this codec should be Opus 
helps in making sure these products ship with Opus.

The time to decide is now.

                   Harald

[Note - the same logic applies to video codecs, but I've accepted that 
it's impossible to make a consensus decision at this time on that issue. 
We'll just live with the consequences of that.]

On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
> At the last meeting we took a hum on selecting Opus and G.711 as the mediatory to implement audio codecs. If there is any new opinions please send them to the list by August 30th, after which the chairs will make a determination of consensus.
>
> Thanks,
> Cullen
>
> Please note that the following IPR disclosure have been made on these codecs. They can be found at
>
> http://datatracker.ietf.org/ipr/
>
>
> 2010-11-07	
> • ID # 1445
> "Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> 2010-11-07	
> • ID # 1446
> "Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus-00"
> 2010-11-12	
> • ID # 1447
> "Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
> 2011-03-23	
> • ID # 1520
> "Qualcomm Incorporated's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-03-27	
> • ID # 1524
> "Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-03-29	
> • ID # 1526
> "Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-03-29	
> • ID # 1525
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-07-23	
> • ID # 1602
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
> 2012-01-25	
> • ID # 1670
> "Microsoft Corporation's Statement about IPR related to draft-ietf-codec-opus-10"
> 2012-03-12	
> • ID # 1712
> "Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-codec-opus-11 (1)"
> 2012-04-02	
> • ID # 1741
> "Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-codec-opus-11 (2)"
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Markus.Isomaki@nokia.com  Wed Aug 29 03:23:20 2012
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From: <Markus.Isomaki@nokia.com>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Hi,

Olle E. Johansson wrote:
>
>29 aug 2012 kl. 03:24 skrev Lishitao <lishitao@huawei.com>:
>
>> Hi all
>>
>> Both as individual and HUAWEI's point of view,
>>
>PLEASE. In the IETF we're all individuals. A company's point of view does =
not
>matter here.
>I don't see the point of any more companies adding their "point of view".
>

I think it's useful to know if companies do have opinions in certain topics=
. Many browsers, codecs and other RTCWeb related implementations are going =
to happen or not happen based on company decisions. IMHO, more information =
is better than less. The main point continues to be that all opinions shoul=
d be backed by solid technical arguments before they become relevant.

Markus=20

From Martin.Taylor@metaswitch.com  Wed Aug 29 04:26:41 2012
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An Internet codec is one which was designed specifically to handle a signif=
icant level of packet loss.  The G.xxx series codecs were designed for circ=
uit-switched environments where packet loss is a non-issue.

If you are not experiencing voice quality issues with G.722, then you are l=
ucky enough to have a very low loss Internet connection.  We see frequent i=
ssues with G.722 when used for VoIP, especially over WiFi and upstream over=
 ADSL.  Data traffic that competes with voice for bandwidth frequently caus=
es choppy voice on these types of connection.  We have chosen to use SILK t=
o deal with this issue, and this has resulted in major improvements in subj=
ective voice quality.  SILK is one of the "ingredients" of Opus.

I believe that WebRTC applications which use Opus will deliver a far better=
 user experience in general than those that use G.711 or G.722.  Whether th=
at means Opus should be mandatory to implement is another matter.  In my vi=
ew, it is only necessary to specify one MTI codec to ensure that there is a=
 baseline for interop.  The market can decide what codecs it wishes to use =
to improve on this baseline interop.

Martin

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: 29 August 2012 05:14
To: Alan Johnston
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

Not that I have anything against Opus, but what exactly makes Opus an inter=
net codec? What is internet codec anyway? Makes this all kind of a meaningl=
ess argument.

I would argue that for my broadband internet connection G.722 is a perfect =
internet codec. I do not care about bandwidth savings of Opus, and quality =
wise, for the voice conversation, I cannot hear any difference.

I would argue G.711 should be the MTI codec. The rest can be left up to bro=
wser implementers. We can argue all we want, but the best royalty free low =
bitrate codec available will be the one everybody supports. We can force it=
 to be Opus, but even if we don't, it will still be Opus on its merit alone=
. G.722 will probably end up being supported as well, since it is free, pre=
tty good quality, and easy to implement.

P.S. Not that I am arguing for it, I am surprised no one made a case for iS=
AC, since it is also royalty free, low bit rate, and very high quality. It =
is event named "internet Speech Audio Codec".
_____________
Roman Shpount

On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston <alan.b.johnston@gmail.com<=
mailto:alan.b.johnston@gmail.com>> wrote:
The RTCWEB effort needs an Internet codec.  This is why Opus is the
right choice.  RTCWEB also needs one codec for backwards compatibility
with the VoIP world.  This is why G.711 is also the right choice.

Any G.mumble codec is NOT an Internet codec and will not have the same
performance on the Internet as one that was designed for the Internet!
 If anyone doesn't understand what that means, go back and examine the
CODEC Working Group archives to get educated.

If someone wants another codec instead of Opus, then they need to
propose another Internet codec.  Otherwise, we are not serving the
Internet.

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


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<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D">An Internet codec is one wh=
ich was designed specifically to handle a significant level of packet loss.=
&nbsp; The G.xxx series codecs were designed for circuit-switched
 environments where packet loss is a non-issue.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D">If you are not experiencing=
 voice quality issues with G.722, then you are lucky enough to have a very =
low loss Internet connection.&nbsp; We see frequent issues with
 G.722 when used for VoIP, especially over WiFi and upstream over ADSL.&nbs=
p; Data traffic that competes with voice for bandwidth frequently causes ch=
oppy voice on these types of connection.&nbsp; We have chosen to use SILK t=
o deal with this issue, and this has resulted
 in major improvements in subjective voice quality.&nbsp; SILK is one of th=
e &#8220;ingredients&#8221; of Opus.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D">I believe that WebRTC appli=
cations which use Opus will deliver a far better user experience in general=
 than those that use G.711 or G.722.&nbsp; Whether that means
 Opus should be mandatory to implement is another matter.&nbsp; In my view,=
 it is only necessary to specify one MTI codec to ensure that there is a ba=
seline for interop.&nbsp; The market can decide what codecs it wishes to us=
e to improve on this baseline interop.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D">Martin<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p=
>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.o=
rg]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> 29 August 2012 05:14<br>
<b>To:</b> Alan Johnston<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] RTCWEB needs an Internet Codec<o:p></o:p></spa=
n></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Not that I have anyth=
ing against Opus, but what exactly makes Opus an internet codec? What is in=
ternet codec anyway? Makes this all kind of a meaningless argument.<br>
<br>
I would argue that for my broadband internet connection G.722 is a perfect =
internet codec. I do not care about bandwidth savings of Opus, and quality =
wise, for the voice conversation, I cannot hear any difference.<br>
<br>
I would argue G.711 should be the MTI codec. The rest can be left up to bro=
wser implementers. We can argue all we want, but the best royalty free low =
bitrate codec available will be the one everybody supports. We can force it=
 to be Opus, but even if we don't,
 it will still be Opus on its merit alone. G.722 will probably end up being=
 supported as well, since it is free, pretty good quality, and easy to impl=
ement.<br>
<br>
P.S. Not that I am arguing for it, I am surprised no one made a case for iS=
AC, since it is also royalty free, low bit rate, and very high quality. It =
is event named &quot;internet Speech Audio Codec&quot;.<br clear=3D"all">
_____________<br>
Roman Shpount<br>
<br>
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston &lt;=
<a href=3D"mailto:alan.b.johnston@gmail.com" target=3D"_blank">alan.b.johns=
ton@gmail.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">The RTCWEB effort needs an Internet codec. &nbsp;Thi=
s is why Opus is the<br>
right choice. &nbsp;RTCWEB also needs one codec for backwards compatibility=
<br>
with the VoIP world. &nbsp;This is why G.711 is also the right choice.<br>
<br>
Any G.mumble codec is NOT an Internet codec and will not have the same<br>
performance on the Internet as one that was designed for the Internet!<br>
&nbsp;If anyone doesn't understand what that means, go back and examine the=
<br>
CODEC Working Group archives to get educated.<br>
<br>
If someone wants another codec instead of Opus, then they need to<br>
propose another Internet codec. &nbsp;Otherwise, we are not serving the<br>
Internet.<br>
<br>
- Alan -<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>
</body>
</html>

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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/29/2012 06:13 AM, Roman Shpount wrote:
> Not that I have anything against Opus, but what exactly makes Opus an 
> internet codec? What is internet codec anyway? Makes this all kind of 
> a meaningless argument.
>
> I would argue that for my broadband internet connection G.722 is a 
> perfect internet codec. I do not care about bandwidth savings of Opus, 
> and quality wise, for the voice conversation, I cannot hear any 
> difference.
>
> I would argue G.711 should be the MTI codec. The rest can be left up 
> to browser implementers. We can argue all we want, but the best 
> royalty free low bitrate codec available will be the one everybody 
> supports. We can force it to be Opus, but even if we don't, it will 
> still be Opus on its merit alone. G.722 will probably end up being 
> supported as well, since it is free, pretty good quality, and easy to 
> implement.
>
> P.S. Not that I am arguing for it, I am surprised no one made a case 
> for iSAC, since it is also royalty free, low bit rate, and very high 
> quality. It is event named "internet Speech Audio Codec".

Google, after considering both, would prefer that Opus be the standard.

At high bandwidth / bitrate, Opus is operating in CELT mode, which 
outperforms iSAC.


From jmvalin@mozilla.com  Wed Aug 29 05:15:41 2012
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> (speaking for Nokia)
> 
> Nokia thinks that there are complexity issues with the Opus codec.
> There are also possible IPR risks associated with the "royalty-free"
> nature of Opus. We do not recommend Opus to be taken as a mandatory
> codec for RTCWeb at this point. From high quality low bit rate and
> mobile applications point of view the 3GPP AMR-WB codec (also known
> as ITU-T G.722.2) is the most preferable for us. For interoperability
> with implementations restricted to unencumbered codecs, we prefer
> G.711 and G.722.

Sorry, but I think G.722 (or AMR-WB) is just not enough. Speech-only, AM
radio quality is not what we need here. Skype has been shipping a
super-wideband codec for about 3 years now, so going back to wideband as
the highest guaranteed interoperability would make webrtc irrelevant.
Let's look at the options here. On one side we have Opus, that can
easily handle all possible rtcweb applications because it does
narrowband, wideband, super-wideband and fullband, can handle both
speech and music, and scales to high-quality stereo (for music
applications). On the other side, if you want the same functionality,
you would need at least the following codecs: AMR-NB, AMR-WB, G.722.1C,
AAC-LD, HE-AAC, and AAC-LC. And on top of the insane IPR cost of
shipping these 6 codecs, you end up with an inflexible solution (hard to
switch between 6 codecs in real-time based on the network bandwidth),
with slightly lower quality, and that in practice nobody would fully
implement.

	Jean-Marc

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On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
> the info above goes a long way in answering my questions, thanks.
> I don't know the exact rules for IPR statements so I would like to
> double check: the fact that the RFC only defines the decoder, does
> that mean that there can be IPRs known by people in the codec WG
> that are not declared since they would apply to the encoder only
> (then there is the risk of IPRs held by those who have not
> participated in codec WG)?

The RFC includes and describes both the encoder and the decoder. What
I was saying is that only the decoder was described normatively, while
the encoder was allowed to change. So the IPR rules apply to both the
encoder and the decoder.

> Another question I had was if Opus is atomic, or if you can
> implement different profiles, and if so what parts should be MTI
> for rtcweb (and should the profile that is MTI be the same
> regardless of the device type - there has been issues raised
> regarding complexity?).

Opus does not have profiles. That's what ensures that any Opus
bitstream can be decoded by any compliant decoder. And that can happen
*before* the decoder has even seen the SDP.

> That is great, but sort of underlines that there would be no harm
> in delaying the decision until there are experiences made from real
> world use - 'cause it would not be that long till that experience
> has been made (Markus also brought up the IPR status as a reason
> for waiting - I have no idea how long it would take to know more
> about that).

As Harald is pointing out, rtcweb implementations are going to ship
pretty soon. I don't see a reason to postpone decisions that can be
made now. Are you expecting *another* single, standardized,
royalty-free codec that operates over vast ranges of bitrates and
operating conditions, from narrowband speech to stereo music, all with
low delay, coming out in the next year? If not, what are you really
waiting for?

> As Paul suggested, I was referring to the lack of formal,
> controlled, characterization tests. That is how other SDOs do it. I
> don't think that is the only way to do it, but I think we should at
> least have either such tests conducted or experience from
> deployment and use (in a wide range of conditions and device types)
> before making it MTI.

Opus has had "ITU-style" testing on English (
http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin (
http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you
don't trust Google on the tests I linked to, it's also been tested by
Nokia:
http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf

Now, unlike other SDOs, the testing did not stop there. ITU-T codecs
generally end up being testing with something in the order of tens of
minutes worth of audio. In *addition* to that kind of testing, Opus
also had automated testing with hundreds of years worth of audio. If
anything, I think other SDOs should learn something here.

	Jean-Marc

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Date: Wed, 29 Aug 2012 09:42:01 -0400
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From: Kaiduan Xie <kaiduanx@gmail.com>
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Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
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"The offer would not actually be sent until after the updated local
description has been applied, so this would not violate 3264."

It makes sense now, please add this sentence to the next draft.

/Kaiduan

On Wed, Aug 29, 2012 at 3:31 AM, Justin Uberti <juberti@google.com> wrote:
> The text here is incorrect, this was a mistake on my part. While this usage
> is technically permissible if you can deal with the complications it
> introduces, it is not compliant with RFC 3264, as has been pointed out.
>
> What I meant to say here was that the local description can be updated even
> after an initial local description has been specified. There are real-world
> use cases for this, such as setting an initial local description to tell the
> ICE Agent the number of m= lines to gather candidates for, followed by a
> later update when the getUserMedia MediaStream has been obtained. The offer
> would not actually be sent until after the updated local description has
> been appliced, so this would not violate 3264.
>
>
> On Tue, Aug 28, 2012 at 5:57 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>>
>>
>> On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net>
>> wrote:
>>
>> >> "As in [RFC3264], an offerer can send an offer, and update it as long
>> >> as it has not been answered."
>>
>> yah that won't work or all the reasons mentioned before and in my opinion
>> does not represent the WG consensus. BTW - that was added in the -01 version
>> of the draft and submitted without me ever seeing it - I'd be happy to fix
>> it but I don't have the source for the draft. Chairs?
>>
>>
>>
>> _______________________________________________
>> 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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Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
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> The next sentence just after this one
> " The answerer can send back zero or more
>    provisional answers, and finally end the offer-answer exchange by
>    sending a final answer."
> does not describe quite well either, right?
>
> AFAIK , the answerer can send back zero or more provisional answers, but they are not all SDP answers,

If we look the Section 12.1.1 of RFC5245 (ICE),

  "If an offer is received in an INVITE request, the answerer SHOULD
   begin to gather its candidates on receipt of the offer and then
   generate an answer in a provisional response once it has completed
   that process.  ICE requires that a provisional response with an SDP
   be transmitted reliably.  This can be done through the existing
   Provisional Response Acknowledgment (PRACK) mechanism [RFC3262] or
   through an optimization that is specific to ICE ..."

Here provisional response carriers answer, but the offer/answer
exchange is completed. Offerer can send PRACK with SDP to updates the
offer, and the answerer replies answer in 200 of PRACK, but this is a
new round of offer/answer exchange.

Also from the RFC 6337 (Session Initiation Protocol (SIP) Usage of the
Offer/Answer Model
), we can't find the case where the answerer sends multiple different
answers for one offer.

So it seems that we also need clarification on the following,

"The answerer can send back zero or more provisional answers, and
finally end the offer-answer exchange by
  sending a final answer."

/Kaiduan

On Tue, Aug 28, 2012 at 10:50 PM, Lishitao <lishitao@huawei.com> wrote:
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Cullen Jennings
>> Sent: Tuesday, August 28, 2012 8:58 PM
>> To: Francois Audet
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Clarification on offer/answer in jsep-01
>>
>>
>> On Aug 27, 2012, at 10:48 PM, Francois Audet <francois.audet@skype.net>
>> wrote:
>>
>> >> "As in [RFC3264], an offerer can send an offer, and update it as long
>> >> as it has not been answered."
>>
>> yah that won't work or all the reasons mentioned before and in my opinion does
>> not represent the WG consensus. BTW - that was added in the -01 version of
>> the draft and submitted without me ever seeing it - I'd be happy to fix it but I
>> don't have the source for the draft. Chairs?
>>
>
> The next sentence just after this one
> " The answerer can send back zero or more
>    provisional answers, and finally end the offer-answer exchange by
>    sending a final answer."
> does not describe quite well either, right?
>
> AFAIK , the answerer can send back zero or more provisional answers, but they are not all SDP answers,
>
> and I don't find any place in RFC 3264 has the definition about "final answer".
>
> shitao
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From Bernhard.Feiten@telekom.de  Wed Aug 29 07:01:18 2012
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The mandatory-to implement (MTI) codec for rtcweb should ensure interoperab=
ility and good quality.=20
It should be possible to implement the codec without hurdles.

The G.722 provides wideband quality at 64 kbit/s. It is easy to implement a=
nd causes no additional costs.
Hence G.722 should be a MTI codec for rtcweb.=20
It provides also interoperability to other voice communication services.

For interoperability reasons to other voice communication services G.711 sh=
ould be a MTI codec for rtcweb.=20
It provides only narrow-band quality. But it is easy to implement and cause=
s no additional costs.
=20
For interoperability reasons to other voice communication services  AMR and=
 AMR-WB should be supported.
As these codecs are not that easy to implement and may cause additional cos=
ts, they should not be MTI codecs for rtcweb.
If rtcweb is used for calls to mobile phones, it is likely that these codec=
s will be supported as this guarantees the best quality.=20
 =20
Other  low bitrate codec may be supported , but should not be MTI codec for=
 rtcweb. The codecs cause additional hurdles. Interoperability is already g=
uaranteed by the MTI codecs. The implementer can decide on its own to suppo=
rt the codec. =20

Best regards,
Bernhard


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings (fluffy)
Sent: Donnerstag, 16. August 2012 19:16
To: rtcweb@ietf.org
Subject: [rtcweb] Confirmation of consensus on audio codecs


At the last meeting we took a hum on selecting Opus and G.711 as the mediat=
ory to implement audio codecs. If there is any new opinions please send the=
m to the list by August 30th, after which the chairs will make a determinat=
ion of consensus.

Thanks,
Cullen

Please note that the following IPR disclosure have been made on these codec=
s. They can be found at=20

http://datatracker.ietf.org/ipr/


2010-11-07=09
* ID # 1445
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (1)"
2010-11-07=09
* ID # 1446
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-00"
2010-11-12=09
* ID # 1447
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-00 and draft-ietf-codec-description-00 (2)"
2011-03-23=09
* ID # 1520
"Qualcomm Incorporated's Statement about IPR related to draft-ietf-codec-op=
us-05"
2011-03-27=09
* ID # 1524
"Xiph.Org Foundation's Statement about IPR related to draft-ietf-codec-opus=
-05"
2011-03-29=09
* ID # 1526
"Broadcom Corporation's Statement about IPR related to draft-ietf-codec-opu=
s-05"
2011-03-29=09
* ID # 1525
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
2011-07-23=09
* ID # 1602
"Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
2012-01-25=09
* ID # 1670
"Microsoft Corporation's Statement about IPR related to draft-ietf-codec-op=
us-10"
2012-03-12=09
* ID # 1712
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (1)"
2012-04-02=09
* ID # 1741
"Huawei Technologies Co.,Ltd's Statement about IPR related to draft-ietf-co=
dec-opus-11 (2)"



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

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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Confirmation of consensus on audio codecs
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I support a) below.  This will allow us to gain operational experience with=
 various codecs, and an update to the RTCWeb specifications can later manda=
te additional codecs as needed.

-Giri Mandyam

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Stefan Hakansson LK
Sent: Tuesday, August 28, 2012 5:37 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
>
> At the last meeting we took a hum on selecting Opus and G.711 as the=20
> mediatory to implement audio codecs. If there is any new opinions=20
> please send them to the list by August 30th, after which the chairs=20
> will make a determination of consensus.

As much as I would like to say "I agree", I come to the conclusion it is no=
t that simple.

I have two things that concern me:

* "Opus" not well enough defined
* Opus not mature and proven enough (this is the more important concern)

Lets talk about both of these.

"Opus" not well enough defined
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D
Just saying "Opus" is a bit vague for me (who has not been in the codec WG)=
. Is a certain version meant? Which one? Or is it a moving target (i.e. alw=
ays implement the latest version)?

What is included (is jitter management, loss concealment part of Opus or so=
mething that is up to the implementation)?

And how is it judged what is a compliant implementation? Is there a ref imp=
lementation that you test against? How do you test? Are there test-vectors =
that must be conformed to? Or is "Opus" standardized as a wire format (mean=
ing that any implementation that can produce/decode that wire format is com=
pliant)? If the latter is true, implementers have more wiggle room to e.g. =
reduce complexity or design around IPRs (submarine IPRs may surface). Are i=
mplementors required to support all aspects of Opus, or are there sub-sets/=
profiles that would be sufficient (perhaps for certain classes of devices)?=
 [1] suggests that not all ptime options need to be supported, but are ther=
e others things?

Many of the answers are probably available, but I think they should be put =
in front of the WG before making this decision.

Opus not mature and proven enough
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
As far as I understand, Opus is a brand new standard. There are implementat=
ions on-going, but yet no deployment, no experience from real world use (an=
d definitely not from use in larger scale over a wide range of conditions, =
involving a broad range of end user devices). In addition, I've been told t=
hat so far Opus has gone through less characterization tests than codecs us=
ually are put through.

I don't think it is fair to mandate the implementation of a codec (which is=
 a large and complex piece) under these circumstances. Some real deployment=
 (involving many device types) and experience (including operation in diffe=
rent conditions) should be gained before doing so. It could also be argued =
that there is a bigger risk from a licensing perspective to mandate a codec=
 that has not been in use.


Do I mean we should mandate just G.711?
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
No, not necessarily. Looking ahead, I am convinced that new and improved co=
decs for audio and video will be introduced as they become available.=20
There is a logic in adopting technology that reduces the transmitted number=
 of bits and enhances the end user experience, and the WG is designing a so=
lution that enables new codecs to be introduced and used (and I really hope=
 they can be made use of even if the applications is
untouched) IIUC. However, if G.711 would be the only MTI audio codec, servi=
ces that expect audio BW that goes beyond narrowband could suffer since the=
re is no guarantee that there is any other codec than G.711 supported by bo=
th endpoints.

In addition G.711 is not that well suited for for challenging network condi=
tions (the most important parts like packet loss concealment and jitter man=
agement depends on the implementation - or are we going to standardize thos=
e parts?; what I am after is that it is unable to lower the send rate as re=
action to congestion - the only method available is to use longer frames to=
 make the IP header overhead lower).

I see three possible ways forward:

a) Mandate G.711 only, and let the market decide if there will be a univers=
ally available codec giving higher quality.

b) Postpone the decision for some time. Let Opus (and other new codecs if s=
uch are introduced) get deployed, tested, perhaps improved. The drawback wi=
th this approach is that the selection is delayed. If this approach is chos=
en, I would also recommend that the WG spends some time on selection criter=
ia. In the presentation given in Vancouver ([2]) six criteria were used (th=
ree of them considered more important). I think they make sense, but can't =
remember we actually discussed them (or what other criteria that may make s=
ense).

c) Mandate some other codec (that is already deployed and in use).

Of these approaches, perhaps b) is the most reasonable one.

Stefan

[1] http://datatracker.ietf.org/doc/draft-cbran-rtcweb-codec/?include_text=
=3D1
[2] http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-6.pdf




>
> Thanks, Cullen
>
> Please note that the following IPR disclosure have been made on these=20
> codecs. They can be found at
>
> http://datatracker.ietf.org/ipr/
>
>
> 2010-11-07 * ID # 1445 "Broadcom Corporation's Statement about IPR=20
> related to draft-ietf-codec-opus-00 and
> draft-ietf-codec-description-00 (1)" 2010-11-07 * ID # 1446 "Xiph.Org=20
> Foundation's Statement about IPR related to draft-ietf-codec-opus-00"=20
> 2010-11-12 * ID # 1447 "Broadcom Corporation's Statement about IPR=20
> related to draft-ietf-codec-opus-00 and=20
> draft-ietf-codec-description-00 (2)" 2011-03-23 * ID # 1520 "Qualcomm=20
> Incorporated's Statement about IPR related to=20
> draft-ietf-codec-opus-05" 2011-03-27 * ID # 1524 "Xiph.Org=20
> Foundation's Statement about IPR related to draft-ietf-codec-opus-05"=20
> 2011-03-29 * ID # 1526 "Broadcom Corporation's Statement about IPR=20
> related to draft-ietf-codec-opus-05" 2011-03-29 * ID # 1525 "Skype=20
> Limited's Statement about IPR related to draft-ietf-codec-opus-05"=20
> 2011-07-23 * ID # 1602 "Skype Limited's Statement about IPR related to=20
> draft-ietf-codec-opus-07" 2012-01-25 * ID # 1670 "Microsoft=20
> Corporation's Statement about IPR related to draft-ietf-codec-opus-10"=20
> 2012-03-12 * ID # 1712 "Huawei Technologies Co.,Ltd's Statement about=20
> IPR related to draft-ietf-codec-opus-11 (1)" 2012-04-02 * ID # 1741=20
> "Huawei Technologies Co.,Ltd's Statement about IPR related to=20
> draft-ietf-codec-opus-11 (2)"
>
>
>
> _______________________________________________ rtcweb mailing list=20
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>

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

From petithug@acm.org  Wed Aug 29 07:48:44 2012
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

I support the selection of Opus and G.711 as mandatory.

On 08/16/2012 10:15 AM, Cullen Jennings (fluffy) wrote:
> 
> At the last meeting we took a hum on selecting Opus and G.711 as the 
> mediatory to implement audio codecs. If there is any new opinions please
> send them to the list by August 30th, after which the chairs will make a 
> determination of consensus.
> 
> Thanks, Cullen
> 
> Please note that the following IPR disclosure have been made on these
> codecs. They can be found at
> 
> http://datatracker.ietf.org/ipr/
> 
> 
> 2010-11-07 â€¢ ID # 1445 "Broadcom Corporation's Statement about IPR related
> to draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> 2010-11-07 â€¢ ID # 1446 "Xiph.Org Foundation's Statement about IPR related
> to draft-ietf-codec-opus-00" 2010-11-12 â€¢ ID # 1447 "Broadcom
> Corporation's Statement about IPR related to draft-ietf-codec-opus-00 and 
> draft-ietf-codec-description-00 (2)" 2011-03-23 â€¢ ID # 1520 "Qualcomm 
> Incorporated's Statement about IPR related to draft-ietf-codec-opus-05" 
> 2011-03-27 â€¢ ID # 1524 "Xiph.Org Foundation's Statement about IPR related
> to draft-ietf-codec-opus-05" 2011-03-29 â€¢ ID # 1526 "Broadcom
> Corporation's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-03-29 â€¢ ID # 1525 "Skype Limited's Statement about IPR related to 
> draft-ietf-codec-opus-05" 2011-07-23 â€¢ ID # 1602 "Skype Limited's
> Statement about IPR related to draft-ietf-codec-opus-07" 2012-01-25 â€¢ ID #
> 1670 "Microsoft Corporation's Statement about IPR related to 
> draft-ietf-codec-opus-10" 2012-03-12 â€¢ ID # 1712 "Huawei Technologies 
> Co.,Ltd's Statement about IPR related to draft-ietf-codec-opus-11 (1)" 
> 2012-04-02 â€¢ ID # 1741 "Huawei Technologies Co.,Ltd's Statement about IPR 
> related to draft-ietf-codec-opus-11 (2)"
> 

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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From randell-ietf@jesup.org  Wed Aug 29 08:24:53 2012
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I support the selection of Opus and G.711 and MTI codecs (as I did in 
the room, even though I officially proposed Opus only as an option to 
vote on).

I suggest anyone considering suggesting G.711 to examine the slides from 
Tim's presentation, and if possible watch that section of the video or 
listen to the audio (which I assume are online).  It wasn't an accident 
that the vote in-room (and there were a LOT of people) was basically 
"Opus only: ~5 votes; G.711 only: 0 or 1 votes; rest of the room: Opus 
and G.711"

-- 
Randell Jesup
randell-ietf@jesup.org


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On 8/29/12 9:24 AM, Randell Jesup wrote:
> I support the selection of Opus and G.711 and MTI codecs (as I did
> in the room, even though I officially proposed Opus only as an
> option to vote on).

Yes, that seems like the most reasonable approach. I plan to push for
an update to XEP-0266 ("Codecs for Jingle Audio") to align XMPP/Jingle
here (we were waiting for draft-ietf-codec-opus to be approved by the
IESG).

Peter

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


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<!doctype html public "-//W3C//DTD W3 HTML//EN">
<html><head><style type="text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style><title>Re: [rtcweb] Confirmation of consensus on audio
codecs</title></head><body>
<div>At 7:26 PM +0000 8/28/12, &lt;Markus.Isomaki@nokia.com&gt;
wrote:</div>
<div><br></div>
<blockquote type="cite" cite>We do not recommend Opus to be taken as a
mandatory codec for RTCWeb at this point. From high quality low bit
rate and mobile applications point of view the 3GPP AMR-WB codec (also
known as ITU-T G.722.2) is the most preferable for us. For
interoperability with implementations restricted to unencumbered
codecs, we prefer G.711 and G.722.</blockquote>
<div><br></div>
<div>Mandating only G.711 provides a floor to prevent total failure to
negotiate codecs, while allowing implementations to support whichever
codecs make sense in their environments.</div>
<div><br></div>
<div><br></div>
<div>At 8:39 PM -0700 8/21/12, Ted Hardie &lt;ted.ietf at gmail.com&gt;
wrote:</div>
<div><br></div>
<blockquote type="cite" cite>the fundamental design of RTCWEB allows
for the</blockquote>
<blockquote type="cite" cite>negotiation of any codec mutually
supported.&nbsp; The decision to chose a<br>
Mandatory-to-implement was not made to eliminate other choices, but
to<br>
eliminate interoperability failures by ensuring that at least
one</blockquote>
<blockquote type="cite" cite>common codec is always
available.</blockquote>
<div><br></div>
<div>I think this is a key point.&nbsp; Codecs should be mandated only
to prevent negotiation failure, not to guarantee ideal performance,
especially in light of Ted's next statement:</div>
<div><br>
At 8:39 PM -0700 8/21/12, Ted Hardie &lt;ted.ietf at gmail.com&gt;
wrote:</div>
<div><br></div>
<blockquote type="cite" cite>the group presumes that codec support
does not</blockquote>
<blockquote type="cite" cite>come from the downloadable Javascript
application, but from the</blockquote>
<blockquote type="cite" cite>application environment into which it is
downloaded (commonly a<br>
browser or mobile environment using similar technology).&nbsp;
Promoting<br>
support for those environments to have access to codecs supported
by<br>
the underlying hardware is the best way to further this goal, at
least<br>
in my personal opinion.</blockquote>
<div><br></div>
<div>This makes a lot of sense. Especially on mobile devices, natively
supported codecs are likely to have optimized performance within the
operating environment (e.g., mobile device and cellular
channel).</div>
<div><br></div>
<div><tt>--</tt></div>
<div><font color="#000000">Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------</font></div>
<div><font color="#000000">#Random Tag</font></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div><font color="#000000">Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------<br>
When you're young, you look at television and think, There's a<br>
conspiracy. The networks have conspired to dumb us down. But when<br>
you get a little older, you realize that's not true. The networks<br>
are in business to give people exactly what they want. That's a<br>
far more depressing thought. Conspiracy is optimistic! You can<br>
shoot the bastards! We can have a revolution! But the networks<br>
are really in business to give people what they want. It's the<br>
truth. &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;--Steve Jobs, Wired Magazine, February 1996<br>
</font></div></body>
</html>

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At 10:41 PM -0500 8/28/12, Alan Johnston wrote:

>  The RTCWEB effort needs an Internet codec.  This is why Opus is the
>  right choice.  RTCWEB also needs one codec for backwards compatibility
>  with the VoIP world.  This is why G.711 is also the right choice.
>
>  Any G.mumble codec is NOT an Internet codec and will not have the same
>  performance on the Internet as one that was designed for the Internet!
>   If anyone doesn't understand what that means, go back and examine the
>  CODEC Working Group archives to get educated.
>
>  If someone wants another codec instead of Opus, then they need to
>  propose another Internet codec.  Otherwise, we are not serving the
>  Internet.

We only need one mandatory audio codec.  Everything else can be left 
up to implementations and experience.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
The two most common elements in the universe are hydrogen
and stupidity.                           --Harlan Ellison

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At 12:13 AM -0400 8/29/12, Roman Shpount wrote:

>  I would argue G.711 should be the MTI codec. The rest can be left 
> up to browser implementers.

I agree.

>  We can argue all we want, but the best royalty free low bitrate 
> codec available will be the one everybody supports.

Very good point, and why we should mandate only one audio codec.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
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We cause accidents.                              --Nathaniel Borenstein

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On 08/29/2012 11:24 AM, Randell Jesup wrote:
> I support the selection of Opus and G.711 and MTI codecs (as I did in
> the room, even though I officially proposed Opus only as an option to
> vote on).
>
> I suggest anyone considering suggesting G.711 to examine the slides
> from Tim's presentation, and if possible watch that section of the
> video or listen to the audio (which I assume are online).  It wasn't
> an accident that the vote in-room (and there were a LOT of people) was
> basically "Opus only: ~5 votes; G.711 only: 0 or 1 votes; rest of the
> room: Opus and G.711"
>
In light of the fact that the proposal is actually rtc*web*, I'm
surprised that this discussion has gone on for so long.  It's a standard
for the web first, and specifically for realtime communication on the
web.  Interop is important, but it shouldn't be a factor that has such a
strong bearing on it that it blocks adoption of a superior standard for
one that is less so.

I appreciate the sentiment of Opus only as MTI, and the only reason I
would give any bearing to G.711 is for including a bear-minimum degree
of interop with already existing standards that are *not* web-specific.

Again, I reiterate, I fully support Opus as MTI for openness,
performance, efficiency, and latency reasons and G.711 as MTI for legacy
interop reasons.

And I also plainly state that amorphous IPR issues with Opus should be
made concrete before they be taken seriously, since the same claims have
held-up the wider adoption of other free codecs in a similar vein for
years (e.g., Vorbis and Theora, to cite examples) and *nothing*
materialized for them despite very juicy targets implementing them.  "It
hasn't been enough time" should not hold up the standardization
process.  There will never be enough time for the conditions that have
been brought-up again-and-again.

If I can go so far as to say that even the Web itself has faced IPR
issues, what with the alleged patents on hyperlinking, methods of
clicking on a button, and so on.  If we waited for the patent landscape
on such issues to clear before settling on a standard, there'd *be* no
web so to speak.  Opus has IPR in it already that has been licensed in
an RF manner.  This alone is a strong protection, and it's been made
widely aware to all interested parties, so factors such as estoppal and
what not are in effect that will mitigate any claims at this time.  But,
and I reiterate, to say it needs to be *clear* of IPR concerns is an
impossble condition.

I know these are not issues that Randell brought up, but I think this
discussion is no longer productive, and the consensus is clear, both to
people that attended in Vancouver and those outside.  The issues that
have been raised are simply not serious and are merely stalling tactics
at this point holding-up the rtcweb process.  Need I be more explicit
than that, considering a former player has already proposed a competing
standard to rtcweb about the real factors in play here?

From giles@thaumas.net  Wed Aug 29 10:20:53 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 29/08/12 09:29 AM, Randall Gellens wrote:

> We only need one mandatory audio codec.  Everything else can be left up
> to implementations and experience.

I am in favour of both G.711 and Opus as MTI codecs. We need Opus to 
support all the use cases this working group chose to address and to 
offer a technically compelling standard.

In addition, I want G.711 for testing reasons. As an uncompressed format 
it simplifies debugging interoperability issues and bootstrap 
implementations, because you can just look at the data. This is 
especially so for developers not familiar with the details of particular 
codecs. Having *two* manditory to implement codecs ensures that codec 
selection is properly tested in even minimal implementations. I think 
that's a good thing.

  -r

From christer.holmberg@ericsson.com  Wed Aug 29 11:54:04 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "public-webrtc@w3.org" <public-webrtc@w3.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 29 Aug 2012 20:50:05 +0200
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Subject: [rtcweb] H.264 SVC and BUNDLE
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Hi,

At the telco yesterday one of the Microsoft guys claimed that H.264 SVC doe=
s not work with BUNDLE, but he didn't have the details.

Could someone please explain the reason why he/she doesn't think H.264 SVC =
work with BUNDLE?

(Note, that if using the same port for the different codec layers is a prob=
lem, then it's not a BUNDLE problem - it's a generic multiplexing problem.)

Thanks!

Regards,

Christer=

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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Fine works for me but please 722 WB for SHOULD. Of course none of this will
be of any significance if you can't agree on a reasonable offer/answer model
that interoperates with public SIP networks ( previously called the PSTN ) 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Harald Alvestrand
Sent: Wednesday, August 29, 2012 6:22 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Confirmation of consensus on audio codecs

Since the number of people stating their opinion has been large, I'll just
reiterate the opinion I had (and hummed for) in Vancouver:

Opus and G.711 should be mandatory to implement for RTCWEB.

Since the question of whether there's any value to making the decision now
has been raised:

The first interoperable products implementing RTCWEB are shipping within a
very short timeframe. Those first implementations will shape the market for
what's actually used in practice.
In order to allow applications requiring high quality sound to be among the
first ones developed, those first products need to include a common choice
of a high quality codec.

Having the RTCWEB WG state as a decision that this codec should be Opus
helps in making sure these products ship with Opus.

The time to decide is now.

                   Harald

[Note - the same logic applies to video codecs, but I've accepted that it's
impossible to make a consensus decision at this time on that issue. 
We'll just live with the consequences of that.]

On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
> At the last meeting we took a hum on selecting Opus and G.711 as the
mediatory to implement audio codecs. If there is any new opinions please
send them to the list by August 30th, after which the chairs will make a
determination of consensus.
>
> Thanks,
> Cullen
>
> Please note that the following IPR disclosure have been made on these 
> codecs. They can be found at
>
> http://datatracker.ietf.org/ipr/
>
>
> 2010-11-07	
> . ID # 1445
> "Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
> 2010-11-07	
> . ID # 1446
> "Xiph.Org Foundation's Statement about IPR related to
draft-ietf-codec-opus-00"
> 2010-11-12	
> . ID # 1447
> "Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
> 2011-03-23	
> . ID # 1520
> "Qualcomm Incorporated's Statement about IPR related to
draft-ietf-codec-opus-05"
> 2011-03-27	
> . ID # 1524
> "Xiph.Org Foundation's Statement about IPR related to
draft-ietf-codec-opus-05"
> 2011-03-29	
> . ID # 1526
> "Broadcom Corporation's Statement about IPR related to
draft-ietf-codec-opus-05"
> 2011-03-29	
> . ID # 1525
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
> 2011-07-23	
> . ID # 1602
> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
> 2012-01-25	
> . ID # 1670
> "Microsoft Corporation's Statement about IPR related to
draft-ietf-codec-opus-10"
> 2012-03-12	
> . ID # 1712
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
draft-ietf-codec-opus-11 (1)"
> 2012-04-02	
> . ID # 1741
> "Huawei Technologies Co.,Ltd's Statement about IPR related to
draft-ietf-codec-opus-11 (2)"
>
>
>
> _______________________________________________
> 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: "Dan Wing" <dwing@cisco.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, <public-webrtc@w3.org>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Wednesday, August 29, 2012 11:50 AM
> To: public-webrtc@w3.org; rtcweb@ietf.org
> Subject: [rtcweb] H.264 SVC and BUNDLE
> 
> Hi,
> 
> At the telco yesterday one of the Microsoft guys claimed that H.264 SVC
> does not work with BUNDLE, but he didn't have the details.
> 
> Could someone please explain the reason why he/she doesn't think H.264
> SVC work with BUNDLE?
> 
> (Note, that if using the same port for the different codec layers is a
> problem, then it's not a BUNDLE problem - it's a generic multiplexing
> problem.)

The general multiplexing of layers onto the same 5-tuple is a problem, 
*if* we want the layers to have different DSCP values.  This is because
we cannot successfully and reliably set different DSCP values for packets 
belonging to a 5-tuple over the Internet -- too many things rewrite 
the DSCP field (to a different value) or simply clear the DSCP field.  
This makes it impossible to restore the per-packet DSCP values later 
(e.g., using RSVP, or using any other better/stronger/faster signaling 
method).  If, however, each 5-tuple has its own DSCP value then that
DSCP value could be restored.

-d

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


From randy@qualcomm.com  Wed Aug 29 17:44:53 2012
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To: Basil Mohamed Gohar <basilgohar@librevideo.org>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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At 12:40 PM -0400 8/29/12, Basil Mohamed Gohar wrote:

>  I know these are not issues that Randell brought up, but I think this
>  discussion is no longer productive, and the consensus is clear, both to
>  people that attended in Vancouver and those outside.  The issues that
>  have been raised are simply not serious and are merely stalling tactics
>  at this point holding-up the rtcweb process.  Need I be more explicit
>  than that, considering a former player has already proposed a competing
>  standard to rtcweb about the real factors in play here?

I find the assertion that those who disagree with you are merely 
attempting to delay things to be false, unfair, and insulting.  There 
have been several viewpoints expressed, all backed by solid technical 
grounds.  Some of these views support Opus plus G.711, some support 
G.711 only, some support G.711 plus G.722, some support no MTI audio 
codec at this time.  None of these would prevent rtcweb from 
advancing.  Any of these views would allow implementations to support 
Opus as well as G.722, AMR and AMR-WB.  The nice thing about using 
SDP to negotiate a codec is that it allows any codec to be used.  The 
fact that a codec is not MTI doesn't mean it won't or shouldn't be 
supported (far from it).

It's a normal part of the IETF process that different views are 
expressed, each with technical reasoning.  Different participants are 
free to assign different weights to each reason, and this is also 
expected.  What's not a normal part of the process is to accuse other 
participants of bad intent.  That poisons the dialog and creates 
mistrust, especially of you.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
A long-forgotten loved one will appear soon.  Buy the negatives
at any price.

From harald@alvestrand.no  Wed Aug 29 22:49:38 2012
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On 08/30/2012 12:16 AM, Richard Shockey wrote:
> Fine works for me but please 722 WB for SHOULD. Of course none of this will
> be of any significance if you can't agree on a reasonable offer/answer model
> that interoperates with public SIP networks ( previously called the PSTN )
We've already got people writing Javascript that interworks with 
Asterisk, so I don't think this is going to be a problem (unless someone 
decides to start over again).

>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Harald Alvestrand
> Sent: Wednesday, August 29, 2012 6:22 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
> Since the number of people stating their opinion has been large, I'll just
> reiterate the opinion I had (and hummed for) in Vancouver:
>
> Opus and G.711 should be mandatory to implement for RTCWEB.
>
> Since the question of whether there's any value to making the decision now
> has been raised:
>
> The first interoperable products implementing RTCWEB are shipping within a
> very short timeframe. Those first implementations will shape the market for
> what's actually used in practice.
> In order to allow applications requiring high quality sound to be among the
> first ones developed, those first products need to include a common choice
> of a high quality codec.
>
> Having the RTCWEB WG state as a decision that this codec should be Opus
> helps in making sure these products ship with Opus.
>
> The time to decide is now.
>
>                     Harald
>
> [Note - the same logic applies to video codecs, but I've accepted that it's
> impossible to make a consensus decision at this time on that issue.
> We'll just live with the consequences of that.]
>
> On 08/16/2012 07:15 PM, Cullen Jennings (fluffy) wrote:
>> At the last meeting we took a hum on selecting Opus and G.711 as the
> mediatory to implement audio codecs. If there is any new opinions please
> send them to the list by August 30th, after which the chairs will make a
> determination of consensus.
>> Thanks,
>> Cullen
>>
>> Please note that the following IPR disclosure have been made on these
>> codecs. They can be found at
>>
>> http://datatracker.ietf.org/ipr/
>>
>>
>> 2010-11-07	
>> . ID # 1445
>> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (1)"
>> 2010-11-07	
>> . ID # 1446
>> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-00"
>> 2010-11-12	
>> . ID # 1447
>> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-00 and draft-ietf-codec-description-00 (2)"
>> 2011-03-23	
>> . ID # 1520
>> "Qualcomm Incorporated's Statement about IPR related to
> draft-ietf-codec-opus-05"
>> 2011-03-27	
>> . ID # 1524
>> "Xiph.Org Foundation's Statement about IPR related to
> draft-ietf-codec-opus-05"
>> 2011-03-29	
>> . ID # 1526
>> "Broadcom Corporation's Statement about IPR related to
> draft-ietf-codec-opus-05"
>> 2011-03-29	
>> . ID # 1525
>> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-05"
>> 2011-07-23	
>> . ID # 1602
>> "Skype Limited's Statement about IPR related to draft-ietf-codec-opus-07"
>> 2012-01-25	
>> . ID # 1670
>> "Microsoft Corporation's Statement about IPR related to
> draft-ietf-codec-opus-10"
>> 2012-03-12	
>> . ID # 1712
>> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (1)"
>> 2012-04-02	
>> . ID # 1741
>> "Huawei Technologies Co.,Ltd's Statement about IPR related to
> draft-ietf-codec-opus-11 (2)"
>>
>>
>> _______________________________________________
>> 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  Thu Aug 30 00:51:48 2012
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To: Dan Wing <dwing@cisco.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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Hi Dan,

Thanks for the information!

The issue you raise is of course not related to BUNDLE, but to multiplexing=
 in general.

Regards,

Christer
________________________________________
From: Dan Wing [dwing@cisco.com]
Sent: Thursday, August 30, 2012 1:32 AM
To: Christer Holmberg; public-webrtc@w3.org; rtcweb@ietf.org
Subject: RE: [rtcweb] H.264 SVC and BUNDLE

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Wednesday, August 29, 2012 11:50 AM
> To: public-webrtc@w3.org; rtcweb@ietf.org
> Subject: [rtcweb] H.264 SVC and BUNDLE
>
> Hi,
>
> At the telco yesterday one of the Microsoft guys claimed that H.264 SVC
> does not work with BUNDLE, but he didn't have the details.
>
> Could someone please explain the reason why he/she doesn't think H.264
> SVC work with BUNDLE?
>
> (Note, that if using the same port for the different codec layers is a
> problem, then it's not a BUNDLE problem - it's a generic multiplexing
> problem.)

The general multiplexing of layers onto the same 5-tuple is a problem,
*if* we want the layers to have different DSCP values.  This is because
we cannot successfully and reliably set different DSCP values for packets
belonging to a 5-tuple over the Internet -- too many things rewrite
the DSCP field (to a different value) or simply clear the DSCP field.
This makes it impossible to restore the per-packet DSCP values later
(e.g., using RSVP, or using any other better/stronger/faster signaling
method).  If, however, each 5-tuple has its own DSCP value then that
DSCP value could be restored.

-d

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


From ron.even.tlv@gmail.com  Thu Aug 30 01:56:32 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Dan Wing'" <dwing@cisco.com>, <public-webrtc@w3.org>, <rtcweb@ietf.org>
References: <503E0B63.3020708@ericsson.com>, <E17CAD772E76C742B645BD4DC602CD8106A9810E@NAHALD.us.int.genesyslab.com>	<7F2072F1E0DE894DA4B517B93C6A05853409FF2F4C@ESESSCMS0356.eemea.ericsson.se>, <035101cd8636$3b654160$b22fc420$@com> <7F2072F1E0DE894DA4B517B93C6A05853409FF2F5C@ESESSCMS0356.eemea.ericsson.se>
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Date: Thu, 30 Aug 2012 11:55:02 +0200
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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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Christer,
This is about the multiple session support in MST, look at the example in
section 7.3.3 of RFC6190, using the DDP grouping attribute from RFC5583.
I assume that you can solve it using RFC5576 ssrc-group but you will need to
define the DDP to be used also for ssrc-group

Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Christer Holmberg
Sent: 30 August, 2012 9:51 AM
To: Dan Wing; public-webrtc@w3.org; rtcweb@ietf.org
Subject: Re: [rtcweb] H.264 SVC and BUNDLE


Hi Dan,

Thanks for the information!

The issue you raise is of course not related to BUNDLE, but to multiplexing
in general.

Regards,

Christer
________________________________________
From: Dan Wing [dwing@cisco.com]
Sent: Thursday, August 30, 2012 1:32 AM
To: Christer Holmberg; public-webrtc@w3.org; rtcweb@ietf.org
Subject: RE: [rtcweb] H.264 SVC and BUNDLE

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
> Behalf Of Christer Holmberg
> Sent: Wednesday, August 29, 2012 11:50 AM
> To: public-webrtc@w3.org; rtcweb@ietf.org
> Subject: [rtcweb] H.264 SVC and BUNDLE
>
> Hi,
>
> At the telco yesterday one of the Microsoft guys claimed that H.264 
> SVC does not work with BUNDLE, but he didn't have the details.
>
> Could someone please explain the reason why he/she doesn't think H.264 
> SVC work with BUNDLE?
>
> (Note, that if using the same port for the different codec layers is a 
> problem, then it's not a BUNDLE problem - it's a generic multiplexing
> problem.)

The general multiplexing of layers onto the same 5-tuple is a problem,
*if* we want the layers to have different DSCP values.  This is because we
cannot successfully and reliably set different DSCP values for packets
belonging to a 5-tuple over the Internet -- too many things rewrite the DSCP
field (to a different value) or simply clear the DSCP field.
This makes it impossible to restore the per-packet DSCP values later (e.g.,
using RSVP, or using any other better/stronger/faster signaling method).
If, however, each 5-tuple has its own DSCP value then that DSCP value could
be restored.

-d

> Thanks!
>
> 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


From harald@alvestrand.no  Thu Aug 30 03:56:10 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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The counterpoint is that having G.711 as the only MTI means that you get 
interoperability failure for any application in which G.711 sound 
quality is not acceptable.

This includes all applications involving music.

On 08/29/2012 06:30 PM, Randall Gellens wrote:
> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>
>>  I would argue G.711 should be the MTI codec. The rest can be left up 
>> to browser implementers.
>
> I agree.
>
>>  We can argue all we want, but the best royalty free low bitrate 
>> codec available will be the one everybody supports.
>
> Very good point, and why we should mandate only one audio codec.
>


From stefan.lk.hakansson@ericsson.com  Thu Aug 30 04:58:18 2012
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On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
...
>
>> That is great, but sort of underlines that there would be no harm
>> in delaying the decision until there are experiences made from real
>> world use - 'cause it would not be that long till that experience
>> has been made (Markus also brought up the IPR status as a reason
>> for waiting - I have no idea how long it would take to know more
>> about that).
>
> As Harald is pointing out, rtcweb implementations are going to ship
> pretty soon.

If that is the case (and I think and hope so), why would we need to make 
it MTI before seeing it in action?

[...]

> Are you expecting *another* single, standardized,
> royalty-free codec that operates over vast ranges of bitrates and
> operating conditions, from narrowband speech to stereo music, all with
> low delay, coming out in the next year? If not, what are you really
> waiting for?

No, I'm not expecting that. I would just prefer us to see that Opus does 
indeed deliver as promised before making it MTI. If it does, fine. If 
not, we'd have to discuss what to do then.

To me it is like if you're going to place a bet, for an upcoming big 
race, on a horse that has never been in an actual race, but shows great 
promise. If you know that the horse is going to participate in a few 
practice races soon, would you not prefer to wait and see how it fares 
in those before placing the bet (given that the odds would not change)?

Anyway, that's my view. Let's see what the chairs say.
>
>> As Paul suggested, I was referring to the lack of formal,
>> controlled, characterization tests. That is how other SDOs do it. I
>> don't think that is the only way to do it, but I think we should at
>> least have either such tests conducted or experience from
>> deployment and use (in a wide range of conditions and device types)
>> before making it MTI.
>
> Opus has had "ITU-style" testing on English (
> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin (
> http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you
> don't trust Google on the tests I linked to, it's also been tested by
> Nokia:
> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf
Come on, those tests are very limited compared to a formal 
characterization test. (Example: 
www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx). There is 
very little info on the material used, the environment, processing and 
scripts and so on. And there seems to be no tests whatsoever (at least 
not involving humans) with actual channels introducing jitter and losses.

Note: I am in no way proposing that a formal characterization test is 
needed, or even the right thing to do. It is a costly and time consuming 
process, and alternative approaches could prove to be more efficient. 
What I am saying is that I think we should not mandate a codec that has 
neither gone through that kind of formal characterization testing nor 
has any field experience from actual use on a reasonable scale (covering 
different conditions and device types). It just seems wrong, especially 
given that we will soon have field experience.

>
> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs
> generally end up being testing with something in the order of tens of
> minutes worth of audio. In *addition* to that kind of testing, Opus
> also had automated testing with hundreds of years worth of audio. If
> anything, I think other SDOs should learn something here.

This may very well be true. I guess this comes down to how much you 
trust that the quality assessment models (e.g. PEAQ, POLQA) give the 
same result as human test subjects would. But I think this sounds like a 
really good thing.

>
> 	Jean-Marc
>
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Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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Surely that argument extends to every possible codec.

If the codecs supported by both endpoints do not supply the characteristics=
 needed for the communication, then you will get interoperability failure.

If I want more bandwidth or quality than OPUS supplies, then I will also ge=
t interoperability failure.

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Harald Alvestrand
> Sent: 30 August 2012 11:56
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>=20
> The counterpoint is that having G.711 as the only MTI means that you get
> interoperability failure for any application in which G.711 sound
> quality is not acceptable.
>=20
> This includes all applications involving music.
>=20
> On 08/29/2012 06:30 PM, Randall Gellens wrote:
> > At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
> >
> >>  I would argue G.711 should be the MTI codec. The rest can be left up
> >> to browser implementers.
> >
> > I agree.
> >
> >>  We can argue all we want, but the best royalty free low bitrate
> >> codec available will be the one everybody supports.
> >
> > Very good point, and why we should mandate only one audio codec.
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From neils@vipadia.com  Thu Aug 30 05:46:24 2012
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On Thu, Aug 16, 2012 at 6:15 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com> wrote:
> At the last meeting we took a hum on selecting Opus and G.711 as the mediatory to implement audio codecs. If there is any new opinions please send them to the list by August 30th, after which the chairs will make a determination of consensus.

I support both G.711 and Opus as mandatory to implement.

G.711 alone does not meet the rtcweb goal of producing a viable
alternative to browser plugins as it will be impossible for an
application developer to guarantee an acceptable audio quality
(defined as at least as good as when using plugins) for browser to
browser calls.

Neil

From harald@alvestrand.no  Thu Aug 30 05:51:28 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
> Surely that argument extends to every possible codec.
The set of MTI codecs should cover the set of use cases that have been 
identified.
Additional MTI codecs that don't cover new use cases shouldn't be added.

I believe the use case "distributed music band" in 
draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with G.711.

>
> If the codecs supported by both endpoints do not supply the characteristics needed for the communication, then you will get interoperability failure.
>
> If I want more bandwidth or quality than OPUS supplies, then I will also get interoperability failure.
>
> Keith
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Harald Alvestrand
>> Sent: 30 August 2012 11:56
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>
>> The counterpoint is that having G.711 as the only MTI means that you get
>> interoperability failure for any application in which G.711 sound
>> quality is not acceptable.
>>
>> This includes all applications involving music.
>>
>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>
>>>>   I would argue G.711 should be the MTI codec. The rest can be left up
>>>> to browser implementers.
>>> I agree.
>>>
>>>>   We can argue all we want, but the best royalty free low bitrate
>>>> codec available will be the one everybody supports.
>>> Very good point, and why we should mandate only one audio codec.
>>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From Markus.Isomaki@nokia.com  Thu Aug 30 06:14:21 2012
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From: <Markus.Isomaki@nokia.com>
To: <stefan.lk.hakansson@ericsson.com>, <jmvalin@mozilla.com>
Thread-Topic: Opus over the Internet?
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Subject: [rtcweb] Opus over the Internet?
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Hi,

Stefan Hakansson wrote:
>
>And there seems to be no tests whatsoever (at least not involving
>humans) with actual channels introducing jitter and losses.
>

Are there any tests that actually compare different codecs, say Opus vs. G.=
722 vs. AMR-WB, in typical Internet use, meaning some loss and jitter? I su=
ppose the performance is only partially related to the codec, and partially=
 to other implementation decisions, so it might be difficult to compare app=
les-with-apples. But since people are arguing that Opus is an "Internet cod=
ec", it would be actually nice to see some results in this sense. Or does "=
Internet codec" mean something else?=20

Markus

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Stefan Hakansson LK
>Sent: 30 August, 2012 14:58
>To: Jean-Marc Valin
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
>
>On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
>...
>>
>>> That is great, but sort of underlines that there would be no harm in
>>> delaying the decision until there are experiences made from real
>>> world use - 'cause it would not be that long till that experience has
>>> been made (Markus also brought up the IPR status as a reason for
>>> waiting - I have no idea how long it would take to know more about
>>> that).
>>
>> As Harald is pointing out, rtcweb implementations are going to ship
>> pretty soon.
>
>If that is the case (and I think and hope so), why would we need to make i=
t
>MTI before seeing it in action?
>
>[...]
>
>> Are you expecting *another* single, standardized, royalty-free codec
>> that operates over vast ranges of bitrates and operating conditions,
>> from narrowband speech to stereo music, all with low delay, coming out
>> in the next year? If not, what are you really waiting for?
>
>No, I'm not expecting that. I would just prefer us to see that Opus does
>indeed deliver as promised before making it MTI. If it does, fine. If not,=
 we'd
>have to discuss what to do then.
>
>To me it is like if you're going to place a bet, for an upcoming big race,=
 on a
>horse that has never been in an actual race, but shows great promise. If y=
ou
>know that the horse is going to participate in a few practice races soon, =
would
>you not prefer to wait and see how it fares in those before placing the be=
t
>(given that the odds would not change)?
>
>Anyway, that's my view. Let's see what the chairs say.
>>
>>> As Paul suggested, I was referring to the lack of formal, controlled,
>>> characterization tests. That is how other SDOs do it. I don't think
>>> that is the only way to do it, but I think we should at least have
>>> either such tests conducted or experience from deployment and use (in
>>> a wide range of conditions and device types) before making it MTI.
>>
>> Opus has had "ITU-style" testing on English (
>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin (
>> http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you
>> don't trust Google on the tests I linked to, it's also been tested by
>> Nokia:
>>
>http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>> Quality_Characterization_of_IETF_Opus_Codec.pdf
>Come on, those tests are very limited compared to a formal characterizatio=
n
>test. (Example:
>www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx). There is
>very little info on the material used, the environment, processing and scr=
ipts
>and so on. And there seems to be no tests whatsoever (at least not involvi=
ng
>humans) with actual channels introducing jitter and losses.
>
>Note: I am in no way proposing that a formal characterization test is need=
ed,
>or even the right thing to do. It is a costly and time consuming process, =
and
>alternative approaches could prove to be more efficient.
>What I am saying is that I think we should not mandate a codec that has
>neither gone through that kind of formal characterization testing nor has =
any
>field experience from actual use on a reasonable scale (covering different
>conditions and device types). It just seems wrong, especially given that w=
e will
>soon have field experience.
>
>>
>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs
>> generally end up being testing with something in the order of tens of
>> minutes worth of audio. In *addition* to that kind of testing, Opus
>> also had automated testing with hundreds of years worth of audio. If
>> anything, I think other SDOs should learn something here.
>
>This may very well be true. I guess this comes down to how much you trust
>that the quality assessment models (e.g. PEAQ, POLQA) give the same result
>as human test subjects would. But I think this sounds like a really good t=
hing.
>
>>
>> 	Jean-Marc
>>
>> -----BEGIN PGP SIGNATURE-----
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>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
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>>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From stephane.proust@orange.com  Thu Aug 30 07:12:00 2012
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From: <stephane.proust@orange.com>
To: Martin Taylor <Martin.Taylor@metaswitch.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
Thread-Index: AQHNhZg7VAle8mPeRUCAmEaNdk1WoZdwDLGAgAB45ACAAdmicA==
Date: Thu, 30 Aug 2012 14:11:56 +0000
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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>An Internet codec is one which was designed specifically to handle a signi=
ficant level of packet loss.  The G.xxx series codecs were designed for cir=
cuit-switched environments where packet loss is a non-issue.

Looking at the draft "Summary of Opus listening test results" which is supp=
osed to "examine listening test results obtained for the Opus codec and how=
 they relate to the requirements" I cannot see unfortunately any result on =
OPUS performance with packet losses and jitter nor any other result on any =
real measured usage over internet, especially for the intended usage in "fu=
ll band"... So, I don't know exactly to what extent and with respect to wha=
t test results and what related requirement OPUS would be more an "internet=
 codec" than other G.xxx codecs that have been already and successfully exp=
erienced over internet for years (at least with respect to this "packet los=
s" robustness criterion). Furthermore, some key features that enhance any v=
oice codec performance for usage over internet are anyway outside the codec=
 itself.

Stephane



De=A0: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part =
de Martin Taylor
Envoy=E9=A0: mercredi 29 ao=FBt 2012 13:27
=C0=A0: rtcweb@ietf.org
Objet=A0: Re: [rtcweb] RTCWEB needs an Internet Codec

An Internet codec is one which was designed specifically to handle a signif=
icant level of packet loss.=A0 The G.xxx series codecs were designed for ci=
rcuit-switched environments where packet loss is a non-issue.

If you are not experiencing voice quality issues with G.722, then you are l=
ucky enough to have a very low loss Internet connection.=A0 We see frequent=
 issues with G.722 when used for VoIP, especially over WiFi and upstream ov=
er ADSL.=A0 Data traffic that competes with voice for bandwidth frequently =
causes choppy voice on these types of connection.=A0 We have chosen to use =
SILK to deal with this issue, and this has resulted in major improvements i=
n subjective voice quality.=A0 SILK is one of the "ingredients" of Opus.

I believe that WebRTC applications which use Opus will deliver a far better=
 user experience in general than those that use G.711 or G.722.=A0 Whether =
that means Opus should be mandatory to implement is another matter.=A0 In m=
y view, it is only necessary to specify one MTI codec to ensure that there =
is a baseline for interop.=A0 The market can decide what codecs it wishes t=
o use to improve on this baseline interop.

Martin

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: 29 August 2012 05:14
To: Alan Johnston
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

Not that I have anything against Opus, but what exactly makes Opus an inter=
net codec? What is internet codec anyway? Makes this all kind of a meaningl=
ess argument.

I would argue that for my broadband internet connection G.722 is a perfect =
internet codec. I do not care about bandwidth savings of Opus, and quality =
wise, for the voice conversation, I cannot hear any difference.

I would argue G.711 should be the MTI codec. The rest can be left up to bro=
wser implementers. We can argue all we want, but the best royalty free low =
bitrate codec available will be the one everybody supports. We can force it=
 to be Opus, but even if we don't, it will still be Opus on its merit alone=
. G.722 will probably end up being supported as well, since it is free, pre=
tty good quality, and easy to implement.

P.S. Not that I am arguing for it, I am surprised no one made a case for iS=
AC, since it is also royalty free, low bit rate, and very high quality. It =
is event named "internet Speech Audio Codec".
_____________
Roman Shpount
On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston <alan.b.johnston@gmail.com>=
 wrote:
The RTCWEB effort needs an Internet codec. =A0This is why Opus is the
right choice. =A0RTCWEB also needs one codec for backwards compatibility
with the VoIP world. =A0This is why G.711 is also the right choice.

Any G.mumble codec is NOT an Internet codec and will not have the same
performance on the Internet as one that was designed for the Internet!
=A0If anyone doesn't understand what that means, go back and examine the
CODEC Working Group archives to get educated.

If someone wants another codec instead of Opus, then they need to
propose another Internet codec. =A0Otherwise, we are not serving the
Internet.

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


___________________________________________________________________________=
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/30/2012 02:51 PM, Harald Alvestrand wrote:
> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>> Surely that argument extends to every possible codec.
> The set of MTI codecs should cover the set of use cases that have
> been identified. Additional MTI codecs that don't cover new use cases
> shouldn't be added.
>
> I believe the use case "distributed music band" in
> draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with
> G.711.

I would argue that for most of the use cases (except those where the 
browser is used to interop with PSTN) the expected quality can't be 
satisfied with G.711. A wideband capable codec at the very least is needed.

Another story is that I think that this is not the right time to mandate 
a codec that meets the quality expectations (and I said why I think so 
in [1]).

[1] http://www.ietf.org/mail-archive/web/rtcweb/current/msg05148.html

From christer.holmberg@ericsson.com  Thu Aug 30 07:22:02 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Date: Thu, 30 Aug 2012 16:21:56 +0200
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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>> Surely that argument extends to every possible codec.
>The set of MTI codecs should cover the set of use cases that have been ide=
ntified.
>Additional MTI codecs that don't cover new use cases shouldn't be added.
>
>I believe the use case "distributed music band" in draft-ietf-rtcweb-use-c=
ases-and-requirements can't be satisified with G.711.

Maybe dubstep was invented that way ;)

Regards,

Christer




>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>> Behalf Of Harald Alvestrand
>> Sent: 30 August 2012 11:56
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>
>> The counterpoint is that having G.711 as the only MTI means that you=20
>> get interoperability failure for any application in which G.711 sound=20
>> quality is not acceptable.
>>
>> This includes all applications involving music.
>>
>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>
>>>>   I would argue G.711 should be the MTI codec. The rest can be left=20
>>>> up to browser implementers.
>>> I agree.
>>>
>>>>   We can argue all we want, but the best royalty free low bitrate=20
>>>> codec available will be the one everybody supports.
>>> Very good point, and why we should mandate only one audio codec.
>>>
>> _______________________________________________
>> 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 Martin.Taylor@metaswitch.com  Thu Aug 30 07:36:02 2012
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From: Martin Taylor <Martin.Taylor@metaswitch.com>
To: "stephane.proust@orange.com" <stephane.proust@orange.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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> I cannot see unfortunately any result on OPUS performance with packet los=
ses

I suggest you read section 2.1 of http://tools.ietf.org/html/draft-valin-co=
dec-results-00

Admittedly this test related to SILK rather than to Opus, but Opus inherits=
 the characteristics of SILK as it relates to performance under conditions =
of packet loss.

Martin


-----Original Message-----
From: stephane.proust@orange.com [mailto:stephane.proust@orange.com]=20
Sent: 30 August 2012 15:12
To: Martin Taylor; rtcweb@ietf.org
Subject: RE: [rtcweb] RTCWEB needs an Internet Codec

>An Internet codec is one which was designed specifically to handle a signi=
ficant level of packet loss.  The G.xxx series codecs were designed for cir=
cuit-switched environments where packet loss is a non-issue.

Looking at the draft "Summary of Opus listening test results" which is supp=
osed to "examine listening test results obtained for the Opus codec and how=
 they relate to the requirements" I cannot see unfortunately any result on =
OPUS performance with packet losses and jitter nor any other result on any =
real measured usage over internet, especially for the intended usage in "fu=
ll band"... So, I don't know exactly to what extent and with respect to wha=
t test results and what related requirement OPUS would be more an "internet=
 codec" than other G.xxx codecs that have been already and successfully exp=
erienced over internet for years (at least with respect to this "packet los=
s" robustness criterion). Furthermore, some key features that enhance any v=
oice codec performance for usage over internet are anyway outside the codec=
 itself.

Stephane



De=A0: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part =
de Martin Taylor Envoy=E9=A0: mercredi 29 ao=FBt 2012 13:27 =C0=A0: rtcweb@=
ietf.org Objet=A0: Re: [rtcweb] RTCWEB needs an Internet Codec

An Internet codec is one which was designed specifically to handle a signif=
icant level of packet loss.=A0 The G.xxx series codecs were designed for ci=
rcuit-switched environments where packet loss is a non-issue.

If you are not experiencing voice quality issues with G.722, then you are l=
ucky enough to have a very low loss Internet connection.=A0 We see frequent=
 issues with G.722 when used for VoIP, especially over WiFi and upstream ov=
er ADSL.=A0 Data traffic that competes with voice for bandwidth frequently =
causes choppy voice on these types of connection.=A0 We have chosen to use =
SILK to deal with this issue, and this has resulted in major improvements i=
n subjective voice quality.=A0 SILK is one of the "ingredients" of Opus.

I believe that WebRTC applications which use Opus will deliver a far better=
 user experience in general than those that use G.711 or G.722.=A0 Whether =
that means Opus should be mandatory to implement is another matter.=A0 In m=
y view, it is only necessary to specify one MTI codec to ensure that there =
is a baseline for interop.=A0 The market can decide what codecs it wishes t=
o use to improve on this baseline interop.

Martin

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: 29 August 2012 05:14
To: Alan Johnston
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

Not that I have anything against Opus, but what exactly makes Opus an inter=
net codec? What is internet codec anyway? Makes this all kind of a meaningl=
ess argument.

I would argue that for my broadband internet connection G.722 is a perfect =
internet codec. I do not care about bandwidth savings of Opus, and quality =
wise, for the voice conversation, I cannot hear any difference.

I would argue G.711 should be the MTI codec. The rest can be left up to bro=
wser implementers. We can argue all we want, but the best royalty free low =
bitrate codec available will be the one everybody supports. We can force it=
 to be Opus, but even if we don't, it will still be Opus on its merit alone=
. G.722 will probably end up being supported as well, since it is free, pre=
tty good quality, and easy to implement.

P.S. Not that I am arguing for it, I am surprised no one made a case for iS=
AC, since it is also royalty free, low bit rate, and very high quality. It =
is event named "internet Speech Audio Codec".
_____________
Roman Shpount
On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston <alan.b.johnston@gmail.com>=
 wrote:
The RTCWEB effort needs an Internet codec. =A0This is why Opus is the right=
 choice. =A0RTCWEB also needs one codec for backwards compatibility with th=
e VoIP world. =A0This is why G.711 is also the right choice.

Any G.mumble codec is NOT an Internet codec and will not have the same perf=
ormance on the Internet as one that was designed for the Internet!
=A0If anyone doesn't understand what that means, go back and examine the CO=
DEC Working Group archives to get educated.

If someone wants another codec instead of Opus, then they need to propose a=
nother Internet codec. =A0Otherwise, we are not serving the Internet.

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


___________________________________________________________________________=
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From: "Dan Wing" <dwing@cisco.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, <public-webrtc@w3.org>, <rtcweb@ietf.org>
References: <503E0B63.3020708@ericsson.com>, <E17CAD772E76C742B645BD4DC602CD8106A9810E@NAHALD.us.int.genesyslab.com> <7F2072F1E0DE894DA4B517B93C6A05853409FF2F4C@ESESSCMS0356.eemea.ericsson.se>, <035101cd8636$3b654160$b22fc420$@com> <7F2072F1E0DE894DA4B517B93C6A05853409FF2F5C@ESESSCMS0356.eemea.ericsson.se>
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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, August 30, 2012 12:51 AM
> To: Dan Wing; public-webrtc@w3.org; rtcweb@ietf.org
> Subject: RE: [rtcweb] H.264 SVC and BUNDLE
> 
> 
> Hi Dan,
> 
> Thanks for the information!
> 
> The issue you raise is of course not related to BUNDLE, but to
> multiplexing in general.

Yes, of course.  That's why I answered in-line, after your question
about general multiplexing.

-d


> Regards,
> 
> Christer
> ________________________________________
> From: Dan Wing [dwing@cisco.com]
> Sent: Thursday, August 30, 2012 1:32 AM
> To: Christer Holmberg; public-webrtc@w3.org; rtcweb@ietf.org
> Subject: RE: [rtcweb] H.264 SVC and BUNDLE
> 
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Christer Holmberg
> > Sent: Wednesday, August 29, 2012 11:50 AM
> > To: public-webrtc@w3.org; rtcweb@ietf.org
> > Subject: [rtcweb] H.264 SVC and BUNDLE
> >
> > Hi,
> >
> > At the telco yesterday one of the Microsoft guys claimed that H.264
> SVC
> > does not work with BUNDLE, but he didn't have the details.
> >
> > Could someone please explain the reason why he/she doesn't think
> H.264
> > SVC work with BUNDLE?
> >
> > (Note, that if using the same port for the different codec layers is
> a
> > problem, then it's not a BUNDLE problem - it's a generic multiplexing
> > problem.)
> 
> The general multiplexing of layers onto the same 5-tuple is a problem,
> *if* we want the layers to have different DSCP values.  This is because
> we cannot successfully and reliably set different DSCP values for
> packets
> belonging to a 5-tuple over the Internet -- too many things rewrite
> the DSCP field (to a different value) or simply clear the DSCP field.
> This makes it impossible to restore the per-packet DSCP values later
> (e.g., using RSVP, or using any other better/stronger/faster signaling
> method).  If, however, each 5-tuple has its own DSCP value then that
> DSCP value could be restored.
> 
> -d
> 
> > Thanks!
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Eggert, Lars" <lars@netapp.com>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [rtcweb] Opus over the Internet?
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On Aug 30, 2012, at 15:14, Markus.Isomaki@nokia.com wrote:
> Are there any tests that actually compare different codecs, say Opus =
vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and =
jitter?
...
> But since people are arguing that Opus is an "Internet codec", it =
would be actually nice to see some results in this sense.=20

+1

Lars=

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At 1:58 PM +0200 8/30/12, Stefan Hakansson LK wrote:

>>  As Harald is pointing out, rtcweb implementations are going to ship
>>  pretty soon.
>
>  If that is the case (and I think and hope so), why would we need to 
> make it MTI before seeing it in action?

Good point.  We can hold off on mandating codecs until we see some 
initial implementations.  We might not need to mandate anything.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
Why was I born with such contemporaries?
                           --Oscar Wilde

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At 2:51 PM +0200 8/30/12, Harald Alvestrand wrote:

>  The set of MTI codecs should cover the set of use cases that have 
> been identified.

That may be too many mandated codecs.  This business of mandating 
specific codecs is difficult and contentious, since there are so many 
competing interests involving different implementation platforms and 
operating environments.  It may also be unnecessary since, obviously, 
implementers want to interoperate and the spectrum of supported 
codecs will reflect this.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
To err is human, to moo bovine.

From ted.ietf@gmail.com  Thu Aug 30 10:47:21 2012
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On Thu, Aug 30, 2012 at 10:26 AM, Randall Gellens <randy@qualcomm.com> wrote:

>>  If that is the case (and I think and hope so), why would we need to make
>> it MTI before seeing it in action?
>
>
> Good point.  We can hold off on mandating codecs until we see some initial
> implementations.  We might not need to mandate anything.

As chair:  The decision to mandate codecs as a way of avoiding
interoperability failures in negotiation has already been taken by the
working group.

regards,

Ted Hardie

From jmvalin@mozilla.com  Thu Aug 30 12:07:22 2012
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Opus over the Internet?
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On 12-08-30 09:14 AM, Markus.Isomaki@nokia.com wrote:
> Are there any tests that actually compare different codecs, say Opus
> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and
> jitter? I suppose the performance is only partially related to the
> codec, and partially to other implementation decisions, so it might
> be difficult to compare apples-with-apples. But since people are
> arguing that Opus is an "Internet codec", it would be actually nice
> to see some results in this sense. Or does "Internet codec" mean
> something else?

The term "Internet codec" means different things to different people,
but to me this means more than "having good packet loss concealment".
Sure we have PLC (though it was never compared to G.722 or AMR-WB) and
we also have (optional) low bit-rate embedded redundancy (also called
FEC) and (also optional) independent frames, but that's just one aspect.

One of the first things we've been asked when designing Opus was to make
the rate *really* adaptable because we never know what kind of rates
will be available. This not only meant having a wide range of bitrates,
but also being able to vary in small increments. This is why Opus scales
from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with
20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in
(on average) ~2.5 kb/s increments. The reason Opus can have more than
1200 possible bitrates is because on the Internet, other layers in the
protocol stack provide octet granularity framing/sizing. We don't need
to spend 11 bits signalling the bitrate because UDP already encodes the
packet size.

There's also practical aspects. If you look at the rates supported by
AMR-WB, you see that *none* of them represents an integer number of
bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per
frame. It's definitely not the only cause, but the bottom line is that
the payload format for AMR-WB (rfc4867) is quite complex. Now compare
this with the Opus RTP payload
(http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although
it's not complete, it's not only much simpler, but it also makes it
possible to decode RTP packets without having even seen the SDP or any
out-of-band signalling.

People may have other definitions, but that's what *I* mean by Opus
being an "Internet codec".

Cheers,

	Jean-Marc

>> -----Original Message----- From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Stefan Hakansson
>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:
>> rtcweb@ietf.org Subject: Re: [rtcweb] Confirmation of consensus on
>> audio codecs
>> 
>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
>>> ...
> 
>>>>> That is great, but sort of underlines that there would be no
>>>>> harm in delaying the decision until there are experiences
>>>>> made from real world use - 'cause it would not be that long
>>>>> till that experience has been made (Markus also brought up
>>>>> the IPR status as a reason for waiting - I have no idea how
>>>>> long it would take to know more about that).
> 
> As Harald is pointing out, rtcweb implementations are going to ship 
> pretty soon.
>>> 
>>> If that is the case (and I think and hope so), why would we need
>>> to make it MTI before seeing it in action?
>>> 
>>> [...]
>>> 
> Are you expecting *another* single, standardized, royalty-free codec 
> that operates over vast ranges of bitrates and operating conditions, 
> from narrowband speech to stereo music, all with low delay, coming
> out in the next year? If not, what are you really waiting for?
>>> 
>>> No, I'm not expecting that. I would just prefer us to see that
>>> Opus does indeed deliver as promised before making it MTI. If it
>>> does, fine. If not, we'd have to discuss what to do then.
>>> 
>>> To me it is like if you're going to place a bet, for an upcoming
>>> big race, on a horse that has never been in an actual race, but
>>> shows great promise. If you know that the horse is going to
>>> participate in a few practice races soon, would you not prefer to
>>> wait and see how it fares in those before placing the bet (given
>>> that the odds would not change)?
>>> 
>>> Anyway, that's my view. Let's see what the chairs say.
> 
>>>>> As Paul suggested, I was referring to the lack of formal,
>>>>> controlled, characterization tests. That is how other SDOs do
>>>>> it. I don't think that is the only way to do it, but I think
>>>>> we should at least have either such tests conducted or
>>>>> experience from deployment and use (in a wide range of
>>>>> conditions and device types) before making it MTI.
> 
> Opus has had "ITU-style" testing on English ( 
> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin
> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you 
> don't trust Google on the tests I linked to, it's also been tested
> by Nokia:
> 
>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>
>>> 
Quality_Characterization_of_IETF_Opus_Codec.pdf
>>> Come on, those tests are very limited compared to a formal
>>> characterization test. (Example: 
>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx).
>>> There is very little info on the material used, the environment,
>>> processing and scripts and so on. And there seems to be no tests
>>> whatsoever (at least not involving humans) with actual channels
>>> introducing jitter and losses.
>>> 
>>> Note: I am in no way proposing that a formal characterization
>>> test is needed, or even the right thing to do. It is a costly and
>>> time consuming process, and alternative approaches could prove to
>>> be more efficient. What I am saying is that I think we should not
>>> mandate a codec that has neither gone through that kind of formal
>>> characterization testing nor has any field experience from actual
>>> use on a reasonable scale (covering different conditions and
>>> device types). It just seems wrong, especially given that we
>>> will soon have field experience.
>>> 
> 
> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs 
> generally end up being testing with something in the order of tens
> of minutes worth of audio. In *addition* to that kind of testing,
> Opus also had automated testing with hundreds of years worth of
> audio. If anything, I think other SDOs should learn something here.
>>> 
>>> This may very well be true. I guess this comes down to how much
>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)
>>> give the same result as human test subjects would. But I think
>>> this sounds like a really good thing.
>>> 
> 
> Jean-Marc
> 
>>> 
>> 
>> _______________________________________________ rtcweb mailing
>> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

From mandyam@quicinc.com  Thu Aug 30 12:13:11 2012
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Harald Alvestrand <harald@alvestrand.no>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Hello Harald,
Can you clarify this viewpoint?  If we use the criterion that any MTI codec=
 has to be suitable for all use cases as currently documented, then it seem=
s neither Opus nor G.711 can suffice. =20

For instance, use case 4.2.7 - "... the user's provider of cellular access =
has QoS support enabled.  The user is able to take advantage of the QoS sup=
port both when accessing via the residential router and when using cellular=
."  If the guaranteed bit rate (GBR, part of the QoS Class Indicator in 3GP=
P defined networks) falls outside of the operating range where Opus is the =
best performing codec (e.g. the sub-8kbps part of the 'Quality vs Bitrate' =
curve in http://www.opus-codec.org/comparison/), another codec would be req=
uired.

The study available at http://research.nokia.com/files/public/%5B16%5D_Inte=
rSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf clearly s=
hows a significant advantage in MOS scores at 5.9 kbps for  AMR-NB as compa=
red to Opus (Silk).  The initial conclusion from my end is that using Opus =
when cellular QoS support results in a guaranteed bit rate less than 8 kbps=
 would not result in the user being "able to take advantage of the QoS supp=
ort" as per the use case.

I think we shouldn't try to mandate codecs to try to cover every single use=
 case.  Rather we should mandate as few as necessary to avoid codec negotia=
tion failure.

-Giri

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Thursday, August 30, 2012 5:51 AM
To: DRAGE, Keith (Keith)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
> Surely that argument extends to every possible codec.
The set of MTI codecs should cover the set of use cases that have been iden=
tified.
Additional MTI codecs that don't cover new use cases shouldn't be added.

I believe the use case "distributed music band" in draft-ietf-rtcweb-use-ca=
ses-and-requirements can't be satisified with G.711.

>
> If the codecs supported by both endpoints do not supply the characteristi=
cs needed for the communication, then you will get interoperability failure=
.
>
> If I want more bandwidth or quality than OPUS supplies, then I will also =
get interoperability failure.
>
> Keith
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>> Behalf Of Harald Alvestrand
>> Sent: 30 August 2012 11:56
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>
>> The counterpoint is that having G.711 as the only MTI means that you=20
>> get interoperability failure for any application in which G.711 sound=20
>> quality is not acceptable.
>>
>> This includes all applications involving music.
>>
>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>
>>>>   I would argue G.711 should be the MTI codec. The rest can be left=20
>>>> up to browser implementers.
>>> I agree.
>>>
>>>>   We can argue all we want, but the best royalty free low bitrate=20
>>>> codec available will be the one everybody supports.
>>> Very good point, and why we should mandate only one audio codec.
>>>
>> _______________________________________________
>> 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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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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Neil Stratford wrote:
> G.711 alone does not meet the rtcweb goal of producing a viable
> alternative to browser plugins as it will be impossible for an
> application developer to guarantee an acceptable audio quality
> (defined as at least as good as when using plugins) for browser to
> browser calls.

And, for those who do not know, this is what plugins are capable of 
today: https://plus.google.com/u/0/117107600391040942348/posts/ZYw4j21RjUL

From jmvalin@mozilla.com  Thu Aug 30 12:40:16 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 12-08-30 03:13 PM, Mandyam, Giridhar wrote:
> The study available at
> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf
> clearly shows a significant advantage in MOS scores at 5.9 kbps for
> AMR-NB as compared to Opus (Silk).  The initial conclusion from my
> end is that using Opus when cellular QoS support results in a
> guaranteed bit rate less than 8 kbps would not result in the user
> being "able to take advantage of the QoS support" as per the use
> case.

So because Opus is better except around 6 kb/s means that we have
negotiation failure and that we should instead use G.711 at 64 kb/s?
Also, keep in mind that browser-to-browser means RTP, which typically
means 16 kb/s worth of overhead anyway (with 20 ms frames). Even though
Opus is not optimal at 6 kb/s, we still have something that works and we
avoid interop failure. And what are the alternatives anyway? If we only
decide on G.711, then we not only have embarrassingly bad quality, but
we also have interop failure below 64 kb/s. If you want guaranteed good
quality without Opus, you can also make AMR-NB, AMR-WB, G.722.1C,
AAC-LD, HE-AAC, and AAC-LC all MTI. Then you'll have (on average)
quality that's almost as good as Opus alone can do, you'll cover almost
as many use cases, and as a bonus, you get all the negotiation and
switching nightmares involved.

	Jean-Marc

> -----Original Message----- From: rtcweb-bounces@ietf.org
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand Sent:
> Thursday, August 30, 2012 5:51 AM To: DRAGE, Keith (Keith) Cc:
> rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
> 
> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>> Surely that argument extends to every possible codec.
> The set of MTI codecs should cover the set of use cases that have
> been identified. Additional MTI codecs that don't cover new use cases
> shouldn't be added.
> 
> I believe the use case "distributed music band" in
> draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with
> G.711.
> 
>> 
>> If the codecs supported by both endpoints do not supply the
>> characteristics needed for the communication, then you will get
>> interoperability failure.
>> 
>> If I want more bandwidth or quality than OPUS supplies, then I will
>> also get interoperability failure.
>> 
>> Keith
>> 
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand 
>>> Sent: 30 August 2012 11:56 To: rtcweb@ietf.org Subject: Re:
>>> [rtcweb] RTCWEB needs an Internet Codec
>>> 
>>> The counterpoint is that having G.711 as the only MTI means that
>>> you get interoperability failure for any application in which
>>> G.711 sound quality is not acceptable.
>>> 
>>> This includes all applications involving music.
>>> 
>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>> 
>>>>> I would argue G.711 should be the MTI codec. The rest can be
>>>>> left up to browser implementers.
>>>> I agree.
>>>> 
>>>>> We can argue all we want, but the best royalty free low
>>>>> bitrate codec available will be the one everybody supports.
>>>> Very good point, and why we should mandate only one audio
>>>> codec.
>>>> 
>>> _______________________________________________ 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 harald@alvestrand.no  Thu Aug 30 12:41:26 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/30/2012 09:13 PM, Mandyam, Giridhar wrote:
> Hello Harald,
> Can you clarify this viewpoint?  If we use the criterion that any MTI codec has to be suitable for all use cases as currently documented, then it seems neither Opus nor G.711 can suffice.
That's not what I said. I said that the *set* of MTI codecs should cover 
the use cases.
We have a requirement (F21) that explicitly mentions commonly supported 
codecs in telephony services; there's no way Opus can cover that case.
>    
>
> For instance, use case 4.2.7 - "... the user's provider of cellular access has QoS support enabled.  The user is able to take advantage of the QoS support both when accessing via the residential router and when using cellular."
You're not reading the latest draft. That section is now 4.2.6.
>    If the guaranteed bit rate (GBR, part of the QoS Class Indicator in 3GPP defined networks) falls outside of the operating range where Opus is the best performing codec (e.g. the sub-8kbps part of the 'Quality vs Bitrate' curve in http://www.opus-codec.org/comparison/), another codec would be required.
Section 4.1 of the draft defines "narrowband" to be "10s to 100s of 
Kbits/sec".
>
> The study available at http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf clearly shows a significant advantage in MOS scores at 5.9 kbps for  AMR-NB as compared to Opus (Silk).
Respectfully ..... the slight advantage in quality that AMR-NB seems to 
offer over Opus at 8 kbps is very different from the advantage that Opus 
enjoys over G.711 at 64 Kbits and above.
>    The initial conclusion from my end is that using Opus when cellular QoS support results in a guaranteed bit rate less than 8 kbps would not result in the user being "able to take advantage of the QoS support" as per the use case.
Neither does it cover the case where there's not enough bits to carry an 
IP+RTP header every 20 milliseconds. There are cases where WebRTC won't 
work.
>
> I think we shouldn't try to mandate codecs to try to cover every single use case.  Rather we should mandate as few as necessary to avoid codec negotiation failure.
Yes. And failure to negotiate an usable codec is a negotiation failure.
>
> -Giri
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: Thursday, August 30, 2012 5:51 AM
> To: DRAGE, Keith (Keith)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>
> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>> Surely that argument extends to every possible codec.
> The set of MTI codecs should cover the set of use cases that have been identified.
> Additional MTI codecs that don't cover new use cases shouldn't be added.
>
> I believe the use case "distributed music band" in draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with G.711.
>
>> If the codecs supported by both endpoints do not supply the characteristics needed for the communication, then you will get interoperability failure.
>>
>> If I want more bandwidth or quality than OPUS supplies, then I will also get interoperability failure.
>>
>> Keith
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Harald Alvestrand
>>> Sent: 30 August 2012 11:56
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>
>>> The counterpoint is that having G.711 as the only MTI means that you
>>> get interoperability failure for any application in which G.711 sound
>>> quality is not acceptable.
>>>
>>> This includes all applications involving music.
>>>
>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>>
>>>>>    I would argue G.711 should be the MTI codec. The rest can be left
>>>>> up to browser implementers.
>>>> I agree.
>>>>
>>>>>    We can argue all we want, but the best royalty free low bitrate
>>>>> codec available will be the one everybody supports.
>>>> Very good point, and why we should mandate only one audio codec.
>>>>
>>> _______________________________________________
>>> 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: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] H.264 SVC and BUNDLE
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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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Dan Wing wrote:
> The general multiplexing of layers onto the same 5-tuple is a problem,
> *if* we want the layers to have different DSCP values.  This is because
> we cannot successfully and reliably set different DSCP values for packets
> belonging to a 5-tuple over the Internet -- too many things rewrite
> the DSCP field (to a different value) or simply clear the DSCP field.
> This makes it impossible to restore the per-packet DSCP values later
> (e.g., using RSVP, or using any other better/stronger/faster signaling
> method).  If, however, each 5-tuple has its own DSCP value then that
> DSCP value could be restored.

Dan,
You make a claim that is not necessarily true.  If packets within a 5-tuple=
 have different DSCP values that are then erased by the network, why would =
it be "impossible" to restore those values later?  Within the context of mu=
ltiplexing of RTP streams, each RTP stream within a 5-tuple would of necess=
ity need to still use a single DSCP value.  If signaling can be used to inf=
orm a node of some means of identifying each RTP stream then the original D=
SCP markings could be restored.

While I agree that it would be easier to assign the same DSCP value to all =
packets in a 5-tuple, it would still be "possible" to filter on, for exampl=
e, the SSRC value used for each stream (in addition to the 5-tuple) to reas=
sign the original DSCP value (or a mapped value with appropriate meaning in=
 the target network) for each stream.

Richard

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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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My assumption has always been that if you were operating on a network 
that could only apply diffserv to separate 5-tuples, you would allocate 
the SSRCs comprising a multilayer encoding to different 5-tuples.

If you were operating on a network that preserved DSCP values, or on one 
that did stateful resource reservation based on SSRC values, you would 
be able to use the same 5-tuple, if desirable.

This choice seems to be a choice that the application writer should take 
before deciding whether or not to use BUNDLE, so it's a little hard to 
see what the relationship to BUNDLE is.

One size does not fit all. Sometimes bundling is not what people want to do.

On 08/30/2012 10:07 PM, Ejzak, Richard P (Richard) wrote:
> Dan Wing wrote:
>> The general multiplexing of layers onto the same 5-tuple is a problem,
>> *if* we want the layers to have different DSCP values.  This is because
>> we cannot successfully and reliably set different DSCP values for packets
>> belonging to a 5-tuple over the Internet -- too many things rewrite
>> the DSCP field (to a different value) or simply clear the DSCP field.
>> This makes it impossible to restore the per-packet DSCP values later
>> (e.g., using RSVP, or using any other better/stronger/faster signaling
>> method).  If, however, each 5-tuple has its own DSCP value then that
>> DSCP value could be restored.
> Dan,
> You make a claim that is not necessarily true.  If packets within a 5-tuple have different DSCP values that are then erased by the network, why would it be "impossible" to restore those values later?  Within the context of multiplexing of RTP streams, each RTP stream within a 5-tuple would of necessity need to still use a single DSCP value.  If signaling can be used to inform a node of some means of identifying each RTP stream then the original DSCP markings could be restored.
>
> While I agree that it would be easier to assign the same DSCP value to all packets in a 5-tuple, it would still be "possible" to filter on, for example, the SSRC value used for each stream (in addition to the 5-tuple) to reassign the original DSCP value (or a mapped value with appropriate meaning in the target network) for each stream.
>
> Richard
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From jmvalin@mozilla.com  Thu Aug 30 13:24:05 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 12-08-30 08:38 AM, DRAGE, Keith (Keith) wrote:
> If the codecs supported by both endpoints do not supply the
> characteristics needed for the communication, then you will get
> interoperability failure.
> 
> If I want more bandwidth or quality than OPUS supplies, then I will
> also get interoperability failure.

Care to provide an example of use case to which Opus does not scale? It
starts from about 6 kb/s, which is certainly low enough (and far less
than the RTP overhead anyway). From there, it can scale to 512 kb/s
stereo, which is more than twice as much as the highest bitrate at which
anyone was ever able to ABX the coded audio against the original.
There's also multi-channel (>2) mappings if needed. Well, we don't cover
dogs, but I think I can live with that.

	Jean-Marc

> Keith
> 
>> -----Original Message----- From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand 
>> Sent: 30 August 2012 11:56 To: rtcweb@ietf.org Subject: Re:
>> [rtcweb] RTCWEB needs an Internet Codec
>> 
>> The counterpoint is that having G.711 as the only MTI means that
>> you get interoperability failure for any application in which G.711
>> sound quality is not acceptable.
>> 
>> This includes all applications involving music.
>> 
>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>> 
>>>> I would argue G.711 should be the MTI codec. The rest can be
>>>> left up to browser implementers.
>>> 
>>> I agree.
>>> 
>>>> We can argue all we want, but the best royalty free low
>>>> bitrate codec available will be the one everybody supports.
>>> 
>>> Very good point, and why we should mandate only one audio codec.
>>> 
>> 
>> _______________________________________________ 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: Gregory Maxwell <gmaxwell@juniper.net>
To: Jean-Marc Valin <jmvalin@mozilla.com>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Date: Thu, 30 Aug 2012 13:58:41 -0700
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Jean-Marc Valin [jmvalin@mozilla.com] wrote:
> Well, we don't cover
> dogs, but I think I can live with that.

But on the internet no one can tell you're a dog.

Fortunately preserving the anonymity of dogs was not an enumerated applicat=
ion.

Including Opus as MTI should be the most straightforward way to ensure inte=
roperability for all the enumerated applications, and covering the space of=
 human perception is also important for all the ones the working group inev=
itably forgot to enumerate.

Codecs are boring. It's important to have good infrastructure uniformly ava=
ilable so that the authors of applications that will build on the browsers =
that implement rtcweb won't have to worry about codecs and can spend their =
attention on the real innovation that will happen on top of this low level =
infrastructure.=

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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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Harald said:=20

> My assumption has always been that if you were operating on a network=20
> that could only apply diffserv to separate 5-tuples=2C you would allocate=
=20
> the SSRCs comprising a multilayer encoding to different 5-tuples.

[BA] RFC 6190 Section 1 says:

   This memo defines two basic modes for transmission of SVC data=2C=0A=
   single-session transmission (SST) and multi-session transmission=0A=
   (MST).  In SST=2C a single RTP session is used for the transmission of=
=0A=
   all scalability layers comprising an SVC bitstream=3B in MST=2C the=0A=
   scalability layers are transported on different RTP sessions. =20
[BA] So it is possible either to use multiple 5-tuples (MST) or a single 5-=
tuple (SST).  However=2C my understanding=20
is that the SST approach is more popular than MST.=20

> This choice seems to be a choice that the application writer should take=
=20
> before deciding whether or not to use BUNDLE=2C so it's a little hard to=
=20
> see what the relationship to BUNDLE is.

[BA] The open question has been how to indicate the desire for both layered=
 coding and BUNDLE in SDP=2C=20
given the backward compatibility issues we have talked about.  For example=
=2C if it is desired to do SST and
BUNDLE=2C do you signal MST and BUNDLE on different ports in the offer and =
then if SST and BUNDLE are
supported by the answer=2C have the answer respond with BUNDLE as well as d=
ependency groups using=20
the same port?  That would seem to imply an assumption that every MST imple=
mentation can also do SST. =20
Or do do you signal SST and BUNDLE on the same port=2C and then if the resp=
onse indicates that the SDP
was rejected=2C formulate an alternative offer (MST with BUNDLE?  H.264/AVC=
 with BUNDLE?=20
SST without BUNDLE??) =20

> One size does not fit all. Sometimes bundling is not what people want to =
do.

[BA] Or sometimes it is=2C but implementations might not be clear how to ne=
gotiate it :)


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Harald said: <br><br><div>&gt=3B=
 My assumption has always been that if you were operating on a network <br>=
&gt=3B that could only apply diffserv to separate 5-tuples=2C you would all=
ocate <br>&gt=3B the SSRCs comprising a multilayer encoding to different 5-=
tuples.<br><br>[BA] RFC 6190 Section 1 says:<br><br><pre class=3D"newpage">=
   This memo defines two basic modes for transmission of SVC data=2C=0A=
   single-session transmission (SST) and multi-session transmission=0A=
   (MST).  In SST=2C a single RTP session is used for the transmission of=
=0A=
   all scalability layers comprising an SVC bitstream=3B in MST=2C the=0A=
   scalability layers are transported on different RTP sessions.  </pre><br=
>[BA] So it is possible either to use multiple 5-tuples (MST) or a single 5=
-tuple (SST).&nbsp=3B However=2C my understanding <br>is that the SST appro=
ach is more popular than MST. <br><br>&gt=3B This choice seems to be a choi=
ce that the application writer should take <br>&gt=3B before deciding wheth=
er or not to use BUNDLE=2C so it's a little hard to <br>&gt=3B see what the=
 relationship to BUNDLE is.<br><br>[BA] The open question has been how to i=
ndicate the desire for both layered coding and BUNDLE in SDP=2C <br>given t=
he backward compatibility issues we have talked about.&nbsp=3B For example=
=2C if it is desired to do SST and<br>BUNDLE=2C do you signal MST and BUNDL=
E on different ports in the offer and then if SST and BUNDLE are<br>support=
ed by the answer=2C have the answer respond with BUNDLE as well as dependen=
cy groups using <br>the same port?&nbsp=3B That would seem to imply an assu=
mption that every MST implementation can also do SST.&nbsp=3B <br>Or do do =
you signal SST and BUNDLE on the same port=2C and then if the response indi=
cates that the SDP<br>was rejected=2C formulate an alternative offer (MST w=
ith BUNDLE?&nbsp=3B H.264/AVC with BUNDLE? <br>SST without BUNDLE??)&nbsp=
=3B <br><br>&gt=3B One size does not fit all. Sometimes bundling is not wha=
t people want to do.<br><br>[BA] Or sometimes it is=2C but implementations =
might not be clear how to negotiate it :)<br><br><br></div> 		 	   		  </di=
v></body>
</html>=

--_4617ccfe-a700-48b7-989b-165a0b84ff40_--

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Subject: Re: [rtcweb] H.264 SVC and BUNDLE
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On 08/30/2012 11:19 PM, Bernard Aboba wrote:
> Harald said:
>
> > My assumption has always been that if you were operating on a network
> > that could only apply diffserv to separate 5-tuples, you would allocate
> > the SSRCs comprising a multilayer encoding to different 5-tuples.
>
> [BA] RFC 6190 Section 1 says:
>
>     This memo defines two basic modes for transmission of SVC data,
>     single-session transmission (SST) and multi-session transmission
>     (MST).  In SST, a single RTP session is used for the transmission of
>     all scalability layers comprising an SVC bitstream; in MST, the
>     scalability layers are transported on different RTP sessions.
>
> [BA] So it is possible either to use multiple 5-tuples (MST) or a 
> single 5-tuple (SST).  However, my understanding
> is that the SST approach is more popular than MST.
>
> > This choice seems to be a choice that the application writer should 
> take
> > before deciding whether or not to use BUNDLE, so it's a little hard to
> > see what the relationship to BUNDLE is.
>
> [BA] The open question has been how to indicate the desire for both 
> layered coding and BUNDLE in SDP,
> given the backward compatibility issues we have talked about. For 
> example, if it is desired to do SST and
> BUNDLE, do you signal MST and BUNDLE on different ports in the offer 
> and then if SST and BUNDLE are
> supported by the answer, have the answer respond with BUNDLE as well 
> as dependency groups using
> the same port?  That would seem to imply an assumption that every MST 
> implementation can also do SST.
> Or do do you signal SST and BUNDLE on the same port, and then if the 
> response indicates that the SDP
> was rejected, formulate an alternative offer (MST with BUNDLE?  
> H.264/AVC with BUNDLE?
> SST without BUNDLE??)
I lack the background to think about this... in existing equipment, if 
you want to do SST, but would be willing to do MST if the other end did 
not support SST, how would your offer look (without BUNDLE)?

My head hurts a little when trying to guess, but from your question it 
sounds like you've done it, so it's better if you just paste an example.

(BTW, this is really an MMUSIC discussion)


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    <div class="moz-cite-prefix">On 08/30/2012 11:19 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W2102260755F0099C2AA98E93A70@phx.gbl"
      type="cite">
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      <div dir="ltr">Harald said: <br>
        <br>
        <div>&gt; My assumption has always been that if you were
          operating on a network <br>
          &gt; that could only apply diffserv to separate 5-tuples, you
          would allocate <br>
          &gt; the SSRCs comprising a multilayer encoding to different
          5-tuples.<br>
          <br>
          [BA] RFC 6190 Section 1 says:<br>
          <br>
          <pre class="newpage">   This memo defines two basic modes for transmission of SVC data,
   single-session transmission (SST) and multi-session transmission
   (MST).  In SST, a single RTP session is used for the transmission of
   all scalability layers comprising an SVC bitstream; in MST, the
   scalability layers are transported on different RTP sessions.  </pre>
          <br>
          [BA] So it is possible either to use multiple 5-tuples (MST)
          or a single 5-tuple (SST).&nbsp; However, my understanding <br>
          is that the SST approach is more popular than MST. <br>
          <br>
          &gt; This choice seems to be a choice that the application
          writer should take <br>
          &gt; before deciding whether or not to use BUNDLE, so it's a
          little hard to <br>
          &gt; see what the relationship to BUNDLE is.<br>
          <br>
          [BA] The open question has been how to indicate the desire for
          both layered coding and BUNDLE in SDP, <br>
          given the backward compatibility issues we have talked about.&nbsp;
          For example, if it is desired to do SST and<br>
          BUNDLE, do you signal MST and BUNDLE on different ports in the
          offer and then if SST and BUNDLE are<br>
          supported by the answer, have the answer respond with BUNDLE
          as well as dependency groups using <br>
          the same port?&nbsp; That would seem to imply an assumption that
          every MST implementation can also do SST.&nbsp; <br>
          Or do do you signal SST and BUNDLE on the same port, and then
          if the response indicates that the SDP<br>
          was rejected, formulate an alternative offer (MST with
          BUNDLE?&nbsp; H.264/AVC with BUNDLE? <br>
          SST without BUNDLE??)&nbsp; <br>
        </div>
      </div>
    </blockquote>
    I lack the background to think about this... in existing equipment,
    if you want to do SST, but would be willing to do MST if the other
    end did not support SST, how would your offer look (without BUNDLE)?<br>
    <br>
    My head hurts a little when trying to guess, but from your question
    it sounds like you've done it, so it's better if you just paste an
    example.<br>
    <br>
    (BTW, this is really an MMUSIC discussion)<br>
    <br>
  </body>
</html>

--------------020601060207030902090407--

From mandyam@quicinc.com  Thu Aug 30 17:18:31 2012
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
Thread-Index: AQHNhgORyuEFfQzQ3keG1dHHPO9QYpdypWyAgAAchICAAAO0AP//n3uwgADSpQD//6kqkA==
Date: Fri, 31 Aug 2012 00:18:28 +0000
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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> So because Opus is better except around 6 kb/s means that we have negotia=
tion failure and that we should instead use G.711 at 64 kb/s?

I don't follow. If both parties in the call support a codec suitable for 6 =
kb/s (e.g. AMR-NB) then there should be no negotiation failure.  MTI still =
allows for non-MTI codecs to be used on a WebRTC call.  The point I was try=
ing to make was that Opus does not satisfy the use case as specified.  I ne=
ver said that G.711 satisfied this use case.  I believe it is unrealistic t=
o require that the set of MTI audio codecs (whatever they may be) satisfy e=
very single use case as currently outlined. =20

> Also, keep in mind that browser-to-browser means RTP, which typically mea=
ns 16 kb/s worth of overhead anyway (with 20 ms frames).

All the more reason to achieve toll quality at codec data rates below 8 kbp=
s when operating in a cellular environment.  Just because there is substant=
ial overhead for IP/UDP/RTP when compared to that for a circuit-switched ca=
ll does not mean we shouldn't strive for the best quality at the lowest dat=
a rate.  Keep in mind that there are codecs that exhibit suitable performan=
ce at even lower data rates than AMR-NB (e.g. EVRC-B at approximately 4 kbp=
s).  If we assume Opus has operate at 8 kbps to exhibit suitable voice qual=
ity (which still needs to be verified under codec performance testing as pe=
r industry standards), that is still a significant increase in required dat=
a rate even accounting for the IP/UDP/RTP header overhead.  Going from Opus=
 at approx.. 8 kbps to EVRC-B at approx. 4 kbps gives near 20% savings in r=
equired bitrate even when considering 16 kbps overhead. This is a significa=
nt savings for cellular access. =20

For any interested parties, EVRC-B characterization tests have been present=
ed previously in the 3GPP2, e.g. the document "Characterization Test Report=
 for EVRC-Release B" (A. Sharpley and I. Panzer, 3GPP2 Document 3GPP2- C11-=
20060327-015, March 27-31, 2006).

> If you want guaranteed good quality without Opus, you can also make AMR-N=
B, AMR-WB, G.722.1C, AAC-LD, HE-AAC, and AAC-LC all MTI. Then you'll have (=
on average) quality that's almost as good as Opus alone can do, you'll cove=
r almost as many use cases, and as a bonus, you get all the negotiation and=
 switching nightmares involved.

I've already stated my position on MTI audio codecs on this mailing list.  =
I never advocated making every codec that is optimal given a particular ope=
rating condition as MTI.  I also think the performance loss due to the use =
of Opus at sub 10 kbps is not trivial, and it is worth leveraging other cod=
ecs at these lower data rates.=20

-----Original Message-----
From: Jean-Marc Valin [mailto:jmvalin@mozilla.com]=20
Sent: Thursday, August 30, 2012 12:40 PM
To: Mandyam, Giridhar
Cc: Harald Alvestrand; DRAGE, Keith (Keith); rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

On 12-08-30 03:13 PM, Mandyam, Giridhar wrote:
> The study available at
> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
> Quality_Characterization_of_IETF_Opus_Codec.pdf
> clearly shows a significant advantage in MOS scores at 5.9 kbps for=20
> AMR-NB as compared to Opus (Silk).  The initial conclusion from my end=20
> is that using Opus when cellular QoS support results in a guaranteed=20
> bit rate less than 8 kbps would not result in the user being "able to=20
> take advantage of the QoS support" as per the use case.

So because Opus is better except around 6 kb/s means that we have negotiati=
on failure and that we should instead use G.711 at 64 kb/s?
Also, keep in mind that browser-to-browser means RTP, which typically means=
 16 kb/s worth of overhead anyway (with 20 ms frames). Even though Opus is =
not optimal at 6 kb/s, we still have something that works and we avoid inte=
rop failure. And what are the alternatives anyway? If we only decide on G.7=
11, then we not only have embarrassingly bad quality, but we also have inte=
rop failure below 64 kb/s. If you want guaranteed good quality without Opus=
, you can also make AMR-NB, AMR-WB, G.722.1C, AAC-LD, HE-AAC, and AAC-LC al=
l MTI. Then you'll have (on average) quality that's almost as good as Opus =
alone can do, you'll cover almost as many use cases, and as a bonus, you ge=
t all the negotiation and switching nightmares involved.

	Jean-Marc

> -----Original Message----- From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand Sent:
> Thursday, August 30, 2012 5:51 AM To: DRAGE, Keith (Keith) Cc:
> rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>=20
> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>> Surely that argument extends to every possible codec.
> The set of MTI codecs should cover the set of use cases that have been=20
> identified. Additional MTI codecs that don't cover new use cases=20
> shouldn't be added.
>=20
> I believe the use case "distributed music band" in=20
> draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with=20
> G.711.
>=20
>>=20
>> If the codecs supported by both endpoints do not supply the=20
>> characteristics needed for the communication, then you will get=20
>> interoperability failure.
>>=20
>> If I want more bandwidth or quality than OPUS supplies, then I will=20
>> also get interoperability failure.
>>=20
>> Keith
>>=20
>>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
>>> Sent: 30 August 2012 11:56 To: rtcweb@ietf.org Subject: Re:
>>> [rtcweb] RTCWEB needs an Internet Codec
>>>=20
>>> The counterpoint is that having G.711 as the only MTI means that you=20
>>> get interoperability failure for any application in which
>>> G.711 sound quality is not acceptable.
>>>=20
>>> This includes all applications involving music.
>>>=20
>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>>=20
>>>>> I would argue G.711 should be the MTI codec. The rest can be left=20
>>>>> up to browser implementers.
>>>> I agree.
>>>>=20
>>>>> We can argue all we want, but the best royalty free low bitrate=20
>>>>> codec available will be the one everybody supports.
>>>> Very good point, and why we should mandate only one audio codec.
>>>>=20
>>> _______________________________________________ rtcweb mailing list=20
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________ rtcweb mailing list=20
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________ rtcweb mailing list=20
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>=20

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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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>That's not what I said. I said that the *set* of MTI codecs should cover t=
he use cases.
>We have a requirement (F21) that explicitly mentions commonly supported co=
decs in telephony services; there's no way Opus can cover that case.

Apologies - inaccurate wording on my part.  I still think it is unrealistic=
 for any suite of MTI codecs to cover all use cases.  I think the QoS-based=
 use case still stands as an example of why this is unrealistic.

> You're not reading the latest draft. That section is now 4.2.6.

The version I linked to from the RTCWEB status pages is -09, which has the =
relevant use case as 4.2.7 (see http://tools.ietf.org/html/draft-ietf-rtcwe=
b-use-cases-and-requirements-09#section-4.2.7).  If the doc is rapidly evol=
ving, then the use case doc may not be suitable for evaluation of codecs fo=
r MTI at this point in time.

> Section 4.1 of the draft defines "narrowband" to be "10s to 100s of Kbits=
/sec".

The full text states  "Clients can be on narrowband (10s to 100s of Kbits/s=
ec)".  Does this definition refer to codecs or the underlying access techno=
logy?  I think in the context one could reasonably conclude that "narrowban=
d" refers to dial-up or low data rate cellular (e.g. GPRS, GSM HSCSD, 1X). =
 The text in 4.1 says nothing about bandlimiting of the input voice signal =
to a codec.  Moreover, the ensuing text states  "Clients can be on variable=
-media-quality networks (wireless)".  To me, this means that the sub-10 kbp=
s case for voice codecs is still relevant.

> Respectfully ..... the slight advantage in quality that AMR-NB seems to o=
ffer over Opus at 8 kbps is very different from the advantage that Opus enj=
oys over G.711 at 64 Kbits and above.

I don't view the quality advantage as 6 kbps as slight, and implementers st=
ill have the option of using Opus at any all data rates without designating=
 it as MTI.  Just as implementers can use AMR-WB, G.722 and so on.  Moreove=
r, the study I cited did not consider codecs such as EVRC-B that can operat=
e in the sub 5 kbps range. =20

>>    The initial conclusion from my end is that using Opus when cellular Q=
oS support results in a guaranteed bit rate less than 8 kbps would not resu=
lt in the user being "able to take advantage of the QoS support" as per the=
 use case.

>Neither does it cover the case where there's not enough bits to carry an I=
P+RTP header every 20 milliseconds. There are cases where WebRTC won't work=
.

I'm not sure what you are trying to say.  Do you believe the use of Opus wi=
ll not satisfy the QoS use case?  Or do you believe that the QoS use case i=
s one where WebRTC will simply not work?

-----Original Message-----
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Thursday, August 30, 2012 12:41 PM
To: Mandyam, Giridhar
Cc: DRAGE, Keith (Keith); rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

On 08/30/2012 09:13 PM, Mandyam, Giridhar wrote:
> Hello Harald,
> Can you clarify this viewpoint?  If we use the criterion that any MTI cod=
ec has to be suitable for all use cases as currently documented, then it se=
ems neither Opus nor G.711 can suffice.
That's not what I said. I said that the *set* of MTI codecs should cover th=
e use cases.
We have a requirement (F21) that explicitly mentions commonly supported cod=
ecs in telephony services; there's no way Opus can cover that case.
>   =20
>
> For instance, use case 4.2.7 - "... the user's provider of cellular acces=
s has QoS support enabled.  The user is able to take advantage of the QoS s=
upport both when accessing via the residential router and when using cellul=
ar."
You're not reading the latest draft. That section is now 4.2.6.
>    If the guaranteed bit rate (GBR, part of the QoS Class Indicator in 3G=
PP defined networks) falls outside of the operating range where Opus is the=
 best performing codec (e.g. the sub-8kbps part of the 'Quality vs Bitrate'=
 curve in http://www.opus-codec.org/comparison/), another codec would be re=
quired.
Section 4.1 of the draft defines "narrowband" to be "10s to 100s of=20
Kbits/sec".
>
> The study available at http://research.nokia.com/files/public/%5B16%5D_In=
terSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf clearly=
 shows a significant advantage in MOS scores at 5.9 kbps for  AMR-NB as com=
pared to Opus (Silk).
Respectfully ..... the slight advantage in quality that AMR-NB seems to=20
offer over Opus at 8 kbps is very different from the advantage that Opus=20
enjoys over G.711 at 64 Kbits and above.
>    The initial conclusion from my end is that using Opus when cellular Qo=
S support results in a guaranteed bit rate less than 8 kbps would not resul=
t in the user being "able to take advantage of the QoS support" as per the =
use case.
Neither does it cover the case where there's not enough bits to carry an=20
IP+RTP header every 20 milliseconds. There are cases where WebRTC won't=20
work.
>
> I think we shouldn't try to mandate codecs to try to cover every single u=
se case.  Rather we should mandate as few as necessary to avoid codec negot=
iation failure.
Yes. And failure to negotiate an usable codec is a negotiation failure.
>
> -Giri
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Harald Alvestrand
> Sent: Thursday, August 30, 2012 5:51 AM
> To: DRAGE, Keith (Keith)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>
> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>> Surely that argument extends to every possible codec.
> The set of MTI codecs should cover the set of use cases that have been id=
entified.
> Additional MTI codecs that don't cover new use cases shouldn't be added.
>
> I believe the use case "distributed music band" in draft-ietf-rtcweb-use-=
cases-and-requirements can't be satisified with G.711.
>
>> If the codecs supported by both endpoints do not supply the characterist=
ics needed for the communication, then you will get interoperability failur=
e.
>>
>> If I want more bandwidth or quality than OPUS supplies, then I will also=
 get interoperability failure.
>>
>> Keith
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Harald Alvestrand
>>> Sent: 30 August 2012 11:56
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>
>>> The counterpoint is that having G.711 as the only MTI means that you
>>> get interoperability failure for any application in which G.711 sound
>>> quality is not acceptable.
>>>
>>> This includes all applications involving music.
>>>
>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>>
>>>>>    I would argue G.711 should be the MTI codec. The rest can be left
>>>>> up to browser implementers.
>>>> I agree.
>>>>
>>>>>    We can argue all we want, but the best royalty free low bitrate
>>>>> codec available will be the one everybody supports.
>>>> Very good point, and why we should mandate only one audio codec.
>>>>
>>> _______________________________________________
>>> 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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On 08/30/2012 08:18 PM, Mandyam, Giridhar wrote:
> I don't follow. If both parties in the call support a codec
> suitable for 6 kb/s (e.g. AMR-NB) then there should be no
> negotiation failure. MTI still allows for non-MTI codecs to be used
> on a WebRTC call.  The point I was trying to make was that Opus
> does not satisfy the use case as specified.  I never said that
> G.711 satisfied this use case. I believe it is unrealistic to
> require that the set of MTI audio codecs (whatever they may be)
> satisfy every single use case as currently outlined.

Here's the thing. I'm not trying to prevent you from including AMR-NB
or any other codec. If all you have is 6 kb/s and you have AMR-NB on
both ends, then by all means use that. But what if that's not the case
and only one side supports it? If only G.711 is MTI, then the
negotiation *fails*. And it won't just fail at 6 kb/s. It'll fail all
the way up to 64 kb/s and that's unacceptable. OTOH, if you have Opus
as MTI, you'll still be able to talk, which is big improvement, even
if the quality isn't nearly as good as AMR-NB or EVRC-B at these
ridiculously low bitrates. Essentially, I've yet to understand your
reasoning that because Opus is "only" optimal for 99% of the use
cases, then we should use something (G.711) that's highly suboptimal
in all use cases.

The way I see this shaping up, it seems highly likely that

> All the more reason to achieve toll quality at codec data rates
> below 8 kbps when operating in a cellular environment.

Sure, go as low as you like. There's even codecs that work at 2 kb/s
and below if you like. If both sides support it, it's all good. But
when all we have in common are the MTI codecs, then please let's make
webrtc not suck or (worse) fail completely.

> I've already stated my position on MTI audio codecs on this
> mailing list.  I never advocated making every codec that is optimal
> given a particular operating condition as MTI.  I also think the
> performance loss due to the use of Opus at sub 10 kbps is not
> trivial, and it is worth leveraging other codecs at these lower
> data rates.

So here's the options we're looking at in the cases where all you have
in common are MTI codecs (and it seems that those will be common):
1) Optimal quality at almost all rates, but sub-optimal below 10 kb/s; or
2) Highly sub-optimal at 64 kb/s and above, with completely failure
below 64 kb/s.

I'm sure most people prefer 1) and I still don't understand why you
would prefer option 2). Option 2) essentially means "we have a really
crappy standard which can only work decently if you have extensions".

	Jean-Marc

> -----Original Message----- From: Jean-Marc Valin 
> [mailto:jmvalin@mozilla.com] Sent: Thursday, August 30, 2012 12:40 
> PM To: Mandyam, Giridhar Cc: Harald Alvestrand; DRAGE, Keith
> (Keith); rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an
> Internet Codec
> 
> On 12-08-30 03:13 PM, Mandyam, Giridhar wrote:
>> The study available at 
>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>>
>>
>> 
Quality_Characterization_of_IETF_Opus_Codec.pdf
>> clearly shows a significant advantage in MOS scores at 5.9 kbps
>> for AMR-NB as compared to Opus (Silk).  The initial conclusion
>> from my end is that using Opus when cellular QoS support results
>> in a guaranteed bit rate less than 8 kbps would not result in the
>> user being "able to take advantage of the QoS support" as per the
>> use case.
> 
> So because Opus is better except around 6 kb/s means that we have 
> negotiation failure and that we should instead use G.711 at 64
> kb/s? Also, keep in mind that browser-to-browser means RTP, which
> typically means 16 kb/s worth of overhead anyway (with 20 ms
> frames). Even though Opus is not optimal at 6 kb/s, we still have
> something that works and we avoid interop failure. And what are the
> alternatives anyway? If we only decide on G.711, then we not only
> have embarrassingly bad quality, but we also have interop failure
> below 64 kb/s. If you want guaranteed good quality without Opus,
> you can also make AMR-NB, AMR-WB, G.722.1C, AAC-LD, HE-AAC, and
> AAC-LC all MTI. Then you'll have (on average) quality that's almost
> as good as Opus alone can do, you'll cover almost as many use
> cases, and as a bonus, you get all the negotiation and switching
> nightmares involved.
> 
> Jean-Marc
> 
>> -----Original Message----- From: rtcweb-bounces@ietf.org 
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand 
>> Sent: Thursday, August 30, 2012 5:51 AM To: DRAGE, Keith (Keith) 
>> Cc: rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an
>> Internet Codec
>> 
>> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>>> Surely that argument extends to every possible codec.
>> The set of MTI codecs should cover the set of use cases that
>> have been identified. Additional MTI codecs that don't cover new
>> use cases shouldn't be added.
>> 
>> I believe the use case "distributed music band" in 
>> draft-ietf-rtcweb-use-cases-and-requirements can't be satisified 
>> with G.711.
>> 
>>> 
>>> If the codecs supported by both endpoints do not supply the 
>>> characteristics needed for the communication, then you will get
>>>  interoperability failure.
>>> 
>>> If I want more bandwidth or quality than OPUS supplies, then I 
>>> will also get interoperability failure.
>>> 
>>> Keith
>>> 
>>>> -----Original Message----- From: rtcweb-bounces@ietf.org 
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald 
>>>> Alvestrand Sent: 30 August 2012 11:56 To: rtcweb@ietf.org 
>>>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>> 
>>>> The counterpoint is that having G.711 as the only MTI means 
>>>> that you get interoperability failure for any application in 
>>>> which G.711 sound quality is not acceptable.
>>>> 
>>>> This includes all applications involving music.
>>>> 
>>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>>> 
>>>>>> I would argue G.711 should be the MTI codec. The rest
>>>>>> can be left up to browser implementers.
>>>>> I agree.
>>>>> 
>>>>>> We can argue all we want, but the best royalty free low 
>>>>>> bitrate codec available will be the one everybody 
>>>>>> supports.
>>>>> Very good point, and why we should mandate only one audio 
>>>>> codec.
>>>>> 
>>>> _______________________________________________ rtcweb
>>>> mailing list rtcweb@ietf.org 
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 
>> _______________________________________________ rtcweb mailing
>> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>  _______________________________________________ rtcweb mailing
>> list rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 

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On 08/31/2012 02:20 AM, Mandyam, Giridhar wrote:
>> That's not what I said. I said that the *set* of MTI codecs should cover the use cases.
>> We have a requirement (F21) that explicitly mentions commonly supported codecs in telephony services; there's no way Opus can cover that case.
> Apologies - inaccurate wording on my part.  I still think it is unrealistic for any suite of MTI codecs to cover all use cases.  I think the QoS-based use case still stands as an example of why this is unrealistic.
I still think you are constructing an unrealistic use case, and one that 
isn't in the document.
If necessary, we should make the document more explicit to say that we 
don't solve for use cases where less than 10s of kilobits of Internet 
bandwidth is available; I believe the speech quality of Opus is not your 
biggest problem in that scenario.

The purpose of an use case document is to describe the problems we're 
solving so that we can separate them from the problems we don't solve. 
I'd be fine with ruling sustained sub-10-Kbits/sec available Internet 
bandwidth explicitly out of scope.
>> You're not reading the latest draft. That section is now 4.2.6.
> The version I linked to from the RTCWEB status pages is -09, which has the relevant use case as 4.2.7 (see http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-09#section-4.2.7).  If the doc is rapidly evolving, then the use case doc may not be suitable for evaluation of codecs for MTI at this point in time.
Yes, apologies; I was the one with an unrefreshed archive.
>
>> Section 4.1 of the draft defines "narrowband" to be "10s to 100s of Kbits/sec".
> The full text states  "Clients can be on narrowband (10s to 100s of Kbits/sec)".  Does this definition refer to codecs or the underlying access technology?  I think in the context one could reasonably conclude that "narrowband" refers to dial-up or low data rate cellular (e.g. GPRS, GSM HSCSD, 1X).  The text in 4.1 says nothing about bandlimiting of the input voice signal to a codec.  Moreover, the ensuing text states  "Clients can be on variable-media-quality networks (wireless)".  To me, this means that the sub-10 kbps case for voice codecs is still relevant.
>
>> Respectfully ..... the slight advantage in quality that AMR-NB seems to offer over Opus at 8 kbps is very different from the advantage that Opus enjoys over G.711 at 64 Kbits and above.
> I don't view the quality advantage as 6 kbps as slight, and implementers still have the option of using Opus at any all data rates without designating it as MTI.  Just as implementers can use AMR-WB, G.722 and so on.  Moreover, the study I cited did not consider codecs such as EVRC-B that can operate in the sub 5 kbps range.
>
>>>     The initial conclusion from my end is that using Opus when cellular QoS support results in a guaranteed bit rate less than 8 kbps would not result in the user being "able to take advantage of the QoS support" as per the use case.
>> Neither does it cover the case where there's not enough bits to carry an IP+RTP header every 20 milliseconds. There are cases where WebRTC won't work.
> I'm not sure what you are trying to say.  Do you believe the use of Opus will not satisfy the QoS use case?  Or do you believe that the QoS use case is one where WebRTC will simply not work?
I believe we'll have a hard time getting the QoS use case working, not 
because of what you say, but because the QoS APIs on the Internet are 
still quite difficult to use. But this is entirely orthogonal to the 
codec question.

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Harald Alvestrand wrote:
> If necessary, we should make the document more explicit to say that we
> don't solve for use cases where less than 10s of kilobits of Internet
> bandwidth is available; I believe the speech quality of Opus is not your
> biggest problem in that scenario.

Agreed. Even if it was, as Jean-Marc points out elsewhere, making Opus 
non-MTI does not magically solve that problem. It just means that you 
may only be left with G.711 to solve that problem. That doesn't exactly 
seem like an improvement.

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From: "Eggert, Lars" <lars@netapp.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>
Thread-Topic: [rtcweb] Opus over the Internet?
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That's all great, but is there any qualitative comparison data of the differ=
ent codecs available?

--=20
Sent from a mobile device; please excuse typos.

On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin@mozilla.com> wrote:

> On 12-08-30 09:14 AM, Markus.Isomaki@nokia.com wrote:
>> Are there any tests that actually compare different codecs, say Opus
>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and
>> jitter? I suppose the performance is only partially related to the
>> codec, and partially to other implementation decisions, so it might
>> be difficult to compare apples-with-apples. But since people are
>> arguing that Opus is an "Internet codec", it would be actually nice
>> to see some results in this sense. Or does "Internet codec" mean
>> something else?
>=20
> The term "Internet codec" means different things to different people,
> but to me this means more than "having good packet loss concealment".
> Sure we have PLC (though it was never compared to G.722 or AMR-WB) and
> we also have (optional) low bit-rate embedded redundancy (also called
> FEC) and (also optional) independent frames, but that's just one aspect.
>=20
> One of the first things we've been asked when designing Opus was to make
> the rate *really* adaptable because we never know what kind of rates
> will be available. This not only meant having a wide range of bitrates,
> but also being able to vary in small increments. This is why Opus scales
> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with
> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in
> (on average) ~2.5 kb/s increments. The reason Opus can have more than
> 1200 possible bitrates is because on the Internet, other layers in the
> protocol stack provide octet granularity framing/sizing. We don't need
> to spend 11 bits signalling the bitrate because UDP already encodes the
> packet size.
>=20
> There's also practical aspects. If you look at the rates supported by
> AMR-WB, you see that *none* of them represents an integer number of
> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per
> frame. It's definitely not the only cause, but the bottom line is that
> the payload format for AMR-WB (rfc4867) is quite complex. Now compare
> this with the Opus RTP payload
> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although
> it's not complete, it's not only much simpler, but it also makes it
> possible to decode RTP packets without having even seen the SDP or any
> out-of-band signalling.
>=20
> People may have other definitions, but that's what *I* mean by Opus
> being an "Internet codec".
>=20
> Cheers,
>=20
>    Jean-Marc
>=20
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Stefan Hakansson
>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:
>>> rtcweb@ietf.org Subject: Re: [rtcweb] Confirmation of consensus on
>>> audio codecs
>>>=20
>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
>>>> ...
>>=20
>>>>>> That is great, but sort of underlines that there would be no
>>>>>> harm in delaying the decision until there are experiences
>>>>>> made from real world use - 'cause it would not be that long
>>>>>> till that experience has been made (Markus also brought up
>>>>>> the IPR status as a reason for waiting - I have no idea how
>>>>>> long it would take to know more about that).
>>=20
>> As Harald is pointing out, rtcweb implementations are going to ship=20
>> pretty soon.
>>>>=20
>>>> If that is the case (and I think and hope so), why would we need
>>>> to make it MTI before seeing it in action?
>>>>=20
>>>> [...]
>>>>=20
>> Are you expecting *another* single, standardized, royalty-free codec=20
>> that operates over vast ranges of bitrates and operating conditions,=20
>> from narrowband speech to stereo music, all with low delay, coming
>> out in the next year? If not, what are you really waiting for?
>>>>=20
>>>> No, I'm not expecting that. I would just prefer us to see that
>>>> Opus does indeed deliver as promised before making it MTI. If it
>>>> does, fine. If not, we'd have to discuss what to do then.
>>>>=20
>>>> To me it is like if you're going to place a bet, for an upcoming
>>>> big race, on a horse that has never been in an actual race, but
>>>> shows great promise. If you know that the horse is going to
>>>> participate in a few practice races soon, would you not prefer to
>>>> wait and see how it fares in those before placing the bet (given
>>>> that the odds would not change)?
>>>>=20
>>>> Anyway, that's my view. Let's see what the chairs say.
>>=20
>>>>>> As Paul suggested, I was referring to the lack of formal,
>>>>>> controlled, characterization tests. That is how other SDOs do
>>>>>> it. I don't think that is the only way to do it, but I think
>>>>>> we should at least have either such tests conducted or
>>>>>> experience from deployment and use (in a wide range of
>>>>>> conditions and device types) before making it MTI.
>>=20
>> Opus has had "ITU-style" testing on English (=20
>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin
>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you=20
>> don't trust Google on the tests I linked to, it's also been tested
>> by Nokia:
>>=20
>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>>=20
>>>>=20
> Quality_Characterization_of_IETF_Opus_Codec.pdf
>>>> Come on, those tests are very limited compared to a formal
>>>> characterization test. (Example:=20
>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx).
>>>> There is very little info on the material used, the environment,
>>>> processing and scripts and so on. And there seems to be no tests
>>>> whatsoever (at least not involving humans) with actual channels
>>>> introducing jitter and losses.
>>>>=20
>>>> Note: I am in no way proposing that a formal characterization
>>>> test is needed, or even the right thing to do. It is a costly and
>>>> time consuming process, and alternative approaches could prove to
>>>> be more efficient. What I am saying is that I think we should not
>>>> mandate a codec that has neither gone through that kind of formal
>>>> characterization testing nor has any field experience from actual
>>>> use on a reasonable scale (covering different conditions and
>>>> device types). It just seems wrong, especially given that we
>>>> will soon have field experience.
>>>>=20
>>=20
>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs=20
>> generally end up being testing with something in the order of tens
>> of minutes worth of audio. In *addition* to that kind of testing,
>> Opus also had automated testing with hundreds of years worth of
>> audio. If anything, I think other SDOs should learn something here.
>>>>=20
>>>> This may very well be true. I guess this comes down to how much
>>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)
>>>> give the same result as human test subjects would. But I think
>>>> this sounds like a really good thing.
>>>>=20
>>=20
>> Jean-Marc
>>=20
>>>>=20
>>>=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

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From: John Leslie <john@jlc.net>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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Harald Alvestrand <harald@alvestrand.no> wrote:
> 
> If necessary, we should make the document more explicit to say that we 
> don't solve for use cases where less than 10s of kilobits of Internet 
> bandwidth is available...

   +1

   There are today some use-cases with less than 56Kbps; but they're
dying, and the infrastructure for them isn't being built anywhere
(including third-world countries, if you must ask).

   I'm very glad there _are_ codecs that can operate in such an
environment; but that's not where our effort belongs. We want something
optimized for 5-10 years down the road.

--
John Leslie <john@jlc.net>

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From: Koen Vos <koen.vos@skype.net>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Opus over the Internet?
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Date: Fri, 31 Aug 2012 14:37:57 +0000
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Lars Eggert wrote:
> That's all great, but is there any qualitative comparison data of the dif=
ferent codecs available?

Please have a look at the test Dynastat did with SILK: http://developer.sky=
pe.com/resources/SILKDataSheet.pdf

Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
  All changes were thoroughly tested to not degrade performance, and in mos=
t cases improved it.  So the SILK Dynastat results give a lower bound on th=
e performance of a subset of the Opus modes.

While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.  According to the test results:
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.

The packet loss in that test was maximally random (ie not bursty).  While t=
hat may not perfectly mimic real networks, I can think of no reason why the=
 results would be reversed or very different in practice.

To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.

Hope this helps,
koen.



Lars Eggert wrote:
That's all great, but is there any qualitative comparison data of the diffe=
rent codecs available?

--
Sent from a mobile device; please excuse typos.

On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin at mozilla.com> wrote=
:

> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:
>> Are there any tests that actually compare different codecs, say Opus
>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and
>> jitter? I suppose the performance is only partially related to the
>> codec, and partially to other implementation decisions, so it might
>> be difficult to compare apples-with-apples. But since people are
>> arguing that Opus is an "Internet codec", it would be actually nice
>> to see some results in this sense. Or does "Internet codec" mean
>> something else?
>
> The term "Internet codec" means different things to different people,
> but to me this means more than "having good packet loss concealment".
> Sure we have PLC (though it was never compared to G.722 or AMR-WB) and
> we also have (optional) low bit-rate embedded redundancy (also called
> FEC) and (also optional) independent frames, but that's just one aspect.
>
> One of the first things we've been asked when designing Opus was to make
> the rate *really* adaptable because we never know what kind of rates
> will be available. This not only meant having a wide range of bitrates,
> but also being able to vary in small increments. This is why Opus scales
> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with
> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in
> (on average) ~2.5 kb/s increments. The reason Opus can have more than
> 1200 possible bitrates is because on the Internet, other layers in the
> protocol stack provide octet granularity framing/sizing. We don't need
> to spend 11 bits signalling the bitrate because UDP already encodes the
> packet size.
>
> There's also practical aspects. If you look at the rates supported by
> AMR-WB, you see that *none* of them represents an integer number of
> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per
> frame. It's definitely not the only cause, but the bottom line is that
> the payload format for AMR-WB (rfc4867) is quite complex. Now compare
> this with the Opus RTP payload
> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although
> it's not complete, it's not only much simpler, but it also makes it
> possible to decode RTP packets without having even seen the SDP or any
> out-of-band signalling.
>
> People may have other definitions, but that's what *I* mean by Opus
> being an "Internet codec".
>
> Cheers,
>
>    Jean-Marc
>
>>> -----Original Message----- From: rtcweb-bounces at ietf.org
>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan Hakansson
>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:
>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of consensus on
>>> audio codecs
>>>
>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
>>>> ...
>>
>>>>>> That is great, but sort of underlines that there would be no
>>>>>> harm in delaying the decision until there are experiences
>>>>>> made from real world use - 'cause it would not be that long
>>>>>> till that experience has been made (Markus also brought up
>>>>>> the IPR status as a reason for waiting - I have no idea how
>>>>>> long it would take to know more about that).
>>
>> As Harald is pointing out, rtcweb implementations are going to ship
>> pretty soon.
>>>>
>>>> If that is the case (and I think and hope so), why would we need
>>>> to make it MTI before seeing it in action?
>>>>
>>>> [...]
>>>>
>> Are you expecting *another* single, standardized, royalty-free codec
>> that operates over vast ranges of bitrates and operating conditions,
>> from narrowband speech to stereo music, all with low delay, coming
>> out in the next year? If not, what are you really waiting for?
>>>>
>>>> No, I'm not expecting that. I would just prefer us to see that
>>>> Opus does indeed deliver as promised before making it MTI. If it
>>>> does, fine. If not, we'd have to discuss what to do then.
>>>>
>>>> To me it is like if you're going to place a bet, for an upcoming
>>>> big race, on a horse that has never been in an actual race, but
>>>> shows great promise. If you know that the horse is going to
>>>> participate in a few practice races soon, would you not prefer to
>>>> wait and see how it fares in those before placing the bet (given
>>>> that the odds would not change)?
>>>>
>>>> Anyway, that's my view. Let's see what the chairs say.
>>
>>>>>> As Paul suggested, I was referring to the lack of formal,
>>>>>> controlled, characterization tests. That is how other SDOs do
>>>>>> it. I don't think that is the only way to do it, but I think
>>>>>> we should at least have either such tests conducted or
>>>>>> experience from deployment and use (in a wide range of
>>>>>> conditions and device types) before making it MTI.
>>
>> Opus has had "ITU-style" testing on English (
>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin
>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you
>> don't trust Google on the tests I linked to, it's also been tested
>> by Nokia:
>>
>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>>
>>>>
> Quality_Characterization_of_IETF_Opus_Codec.pdf
>>>> Come on, those tests are very limited compared to a formal
>>>> characterization test. (Example:
>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx).
>>>> There is very little info on the material used, the environment,
>>>> processing and scripts and so on. And there seems to be no tests
>>>> whatsoever (at least not involving humans) with actual channels
>>>> introducing jitter and losses.
>>>>
>>>> Note: I am in no way proposing that a formal characterization
>>>> test is needed, or even the right thing to do. It is a costly and
>>>> time consuming process, and alternative approaches could prove to
>>>> be more efficient. What I am saying is that I think we should not
>>>> mandate a codec that has neither gone through that kind of formal
>>>> characterization testing nor has any field experience from actual
>>>> use on a reasonable scale (covering different conditions and
>>>> device types). It just seems wrong, especially given that we
>>>> will soon have field experience.
>>>>
>>
>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs
>> generally end up being testing with something in the order of tens
>> of minutes worth of audio. In *addition* to that kind of testing,
>> Opus also had automated testing with hundreds of years worth of
>> audio. If anything, I think other SDOs should learn something here.
>>>>
>>>> This may very well be true. I guess this comes down to how much
>>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)
>>>> give the same result as human test subjects would. But I think
>>>> this sounds like a really good thing.
>>>>
>>
>> Jean-Marc
>>
>>>>
>>>
>>> _______________________________________________ rtcweb mailing
>>> list rtcweb at ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb at ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--_000_D79146E3783B6942A3E8BC43352BBB460579F16DTK5EX14MBXC254r_
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<html dir=3D"ltr">
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1">
<style id=3D"owaParaStyle" type=3D"text/css">=0A=
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<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;"><font face=3D"Courier New">Lars Eggert wrote:<br>
</font>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<div>
<div style=3D"direction:ltr; font-family:Tahoma; color:#000000; font-size:1=
0pt"><font face=3D"Courier New">&gt; That's all great, but is there any qua=
litative comparison data of the different codecs available?
<br>
<br>
Please have a look at the test Dynastat did with SILK: </font><font face=3D=
"Courier New"><a href=3D"http://developer.skype.com/resources/SILKDataSheet=
.pdf" target=3D"_blank">http://developer.skype.com/resources/SILKDataSheet.=
pdf</a><br>
<br>
Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
&nbsp; All changes were thoroughly tested to not degrade performance, and i=
n most cases improved it.&nbsp; So the SILK Dynastat results give a lower b=
ound</font><font face=3D"Courier New"> on the
 performance of a subset of the Opus modes.<br>
<br>
While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.&nbsp; According to the test resul=
ts:
</font><font face=3D"Courier New"><br>
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance<br>
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps <br>
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.<br>
<br>
The packet loss in that test was maximally random (ie not bursty).&nbsp; Wh=
ile that may not perfectly mimic real networks, I can think of no reason wh=
y the results would be reversed or very different in practice.
</font><font face=3D"Courier New"><br>
<br>
To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.
</font><font face=3D"Courier New"><br>
<br>
Hope this helps, </font><font face=3D"Courier New"><br>
koen.<br>
<br>
<br>
</font>
<pre><font face=3D"Courier New">Lars Eggert wrote:<br></font><font face=3D"=
Courier New">That's all great, but is there any qualitative comparison data=
 of the different codecs available?=0A=
=0A=
-- =0A=
Sent from a mobile device; please excuse typos.=0A=
=0A=
On Aug 30, 2012, at 21:07, &quot;Jean-Marc Valin&quot; &lt;jmvalin at mozil=
la.com&gt; wrote:=0A=
=0A=
&gt; On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:=0A=
&gt;&gt; Are there any tests that actually compare different codecs, say Op=
us=0A=
&gt;&gt; vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss a=
nd=0A=
&gt;&gt; jitter? I suppose the performance is only partially related to the=
=0A=
&gt;&gt; codec, and partially to other implementation decisions, so it migh=
t=0A=
&gt;&gt; be difficult to compare apples-with-apples. But since people are=
=0A=
&gt;&gt; arguing that Opus is an &quot;Internet codec&quot;, it would be ac=
tually nice=0A=
&gt;&gt; to see some results in this sense. Or does &quot;Internet codec&qu=
ot; mean=0A=
&gt;&gt; something else?=0A=
&gt; =0A=
&gt; The term &quot;Internet codec&quot; means different things to differen=
t people,=0A=
&gt; but to me this means more than &quot;having good packet loss concealme=
nt&quot;.=0A=
&gt; Sure we have PLC (though it was never compared to G.722 or AMR-WB) and=
=0A=
&gt; we also have (optional) low bit-rate embedded redundancy (also called=
=0A=
&gt; FEC) and (also optional) independent frames, but that's just one aspec=
t.=0A=
&gt; =0A=
&gt; One of the first things we've been asked when designing Opus was to ma=
ke=0A=
&gt; the rate *really* adaptable because we never know what kind of rates=
=0A=
&gt; will be available. This not only meant having a wide range of bitrates=
,=0A=
&gt; but also being able to vary in small increments. This is why Opus scal=
es=0A=
&gt; from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte wit=
h=0A=
&gt; 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 =
in=0A=
&gt; (on average) ~2.5 kb/s increments. The reason Opus can have more than=
=0A=
&gt; 1200 possible bitrates is because on the Internet, other layers in the=
=0A=
&gt; protocol stack provide octet granularity framing/sizing. We don't need=
=0A=
&gt; to spend 11 bits signalling the bitrate because UDP already encodes th=
e=0A=
&gt; packet size.=0A=
&gt; =0A=
&gt; There's also practical aspects. If you look at the rates supported by=
=0A=
&gt; AMR-WB, you see that *none* of them represents an integer number of=0A=
&gt; bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per=
=0A=
&gt; frame. It's definitely not the only cause, but the bottom line is that=
=0A=
&gt; the payload format for AMR-WB (rfc4867) is quite complex. Now compare=
=0A=
&gt; this with the Opus RTP payload=0A=
&gt; (<a rel=3D"nofollow" href=3D"http://tools.ietf.org/html/draft-spittka-=
payload-rtp-opus" target=3D"_blank">http://tools.ietf.org/html/draft-spittk=
a-payload-rtp-opus</a>). Although=0A=
&gt; it's not complete, it's not only much simpler, but it also makes it=0A=
&gt; possible to decode RTP packets without having even seen the SDP or any=
=0A=
&gt; out-of-band signalling.=0A=
&gt; =0A=
&gt; People may have other definitions, but that's what *I* mean by Opus=0A=
&gt; being an &quot;Internet codec&quot;.=0A=
&gt; =0A=
&gt; Cheers,=0A=
&gt; =0A=
&gt;    Jean-Marc=0A=
&gt; =0A=
&gt;&gt;&gt; -----Original Message----- From: rtcweb-bounces at ietf.org=0A=
&gt;&gt;&gt; [<a rel=3D"nofollow" href=3D"mailto:rtcweb-bounces" target=3D"=
_blank">mailto:rtcweb-bounces</a> at ietf.org] On Behalf Of ext Stefan Haka=
nsson=0A=
&gt;&gt;&gt; LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:=0A=
&gt;&gt;&gt; rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of conse=
nsus on=0A=
&gt;&gt;&gt; audio codecs=0A=
&gt;&gt;&gt; =0A=
&gt;&gt;&gt; On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:=0A=
&gt;&gt; On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:=0A=
&gt;&gt;&gt;&gt; ...=0A=
&gt;&gt; =0A=
&gt;&gt;&gt;&gt;&gt;&gt; That is great, but sort of underlines that there w=
ould be no=0A=
&gt;&gt;&gt;&gt;&gt;&gt; harm in delaying the decision until there are expe=
riences=0A=
&gt;&gt;&gt;&gt;&gt;&gt; made from real world use - 'cause it would not be =
that long=0A=
&gt;&gt;&gt;&gt;&gt;&gt; till that experience has been made (Markus also br=
ought up=0A=
&gt;&gt;&gt;&gt;&gt;&gt; the IPR status as a reason for waiting - I have no=
 idea how=0A=
&gt;&gt;&gt;&gt;&gt;&gt; long it would take to know more about that).=0A=
&gt;&gt; =0A=
&gt;&gt; As Harald is pointing out, rtcweb implementations are going to shi=
p =0A=
&gt;&gt; pretty soon.=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; If that is the case (and I think and hope so), why would w=
e need=0A=
&gt;&gt;&gt;&gt; to make it MTI before seeing it in action?=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; [...]=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt; Are you expecting *another* single, standardized, royalty-free cod=
ec =0A=
&gt;&gt; that operates over vast ranges of bitrates and operating condition=
s, =0A=
&gt;&gt; from narrowband speech to stereo music, all with low delay, coming=
=0A=
&gt;&gt; out in the next year? If not, what are you really waiting for?=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; No, I'm not expecting that. I would just prefer us to see =
that=0A=
&gt;&gt;&gt;&gt; Opus does indeed deliver as promised before making it MTI.=
 If it=0A=
&gt;&gt;&gt;&gt; does, fine. If not, we'd have to discuss what to do then.=
=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; To me it is like if you're going to place a bet, for an up=
coming=0A=
&gt;&gt;&gt;&gt; big race, on a horse that has never been in an actual race=
, but=0A=
&gt;&gt;&gt;&gt; shows great promise. If you know that the horse is going t=
o=0A=
&gt;&gt;&gt;&gt; participate in a few practice races soon, would you not pr=
efer to=0A=
&gt;&gt;&gt;&gt; wait and see how it fares in those before placing the bet =
(given=0A=
&gt;&gt;&gt;&gt; that the odds would not change)?=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; Anyway, that's my view. Let's see what the chairs say.=0A=
&gt;&gt; =0A=
&gt;&gt;&gt;&gt;&gt;&gt; As Paul suggested, I was referring to the lack of =
formal,=0A=
&gt;&gt;&gt;&gt;&gt;&gt; controlled, characterization tests. That is how ot=
her SDOs do=0A=
&gt;&gt;&gt;&gt;&gt;&gt; it. I don't think that is the only way to do it, b=
ut I think=0A=
&gt;&gt;&gt;&gt;&gt;&gt; we should at least have either such tests conducte=
d or=0A=
&gt;&gt;&gt;&gt;&gt;&gt; experience from deployment and use (in a wide rang=
e of=0A=
&gt;&gt;&gt;&gt;&gt;&gt; conditions and device types) before making it MTI.=
=0A=
&gt;&gt; =0A=
&gt;&gt; Opus has had &quot;ITU-style&quot; testing on English ( =0A=
&gt;&gt; <a rel=3D"nofollow" href=3D"http://www.opus-codec.org/comparison/G=
oogleTest1.pdf" target=3D"_blank">http://www.opus-codec.org/comparison/Goog=
leTest1.pdf</a> ) and Mandarin=0A=
&gt;&gt; ( <a rel=3D"nofollow" href=3D"http://www.opus-codec.org/comparison=
/GoogleTest2.pdf" target=3D"_blank">http://www.opus-codec.org/comparison/Go=
ogleTest2.pdf</a> ). And if you =0A=
&gt;&gt; don't trust Google on the tests I linked to, it's also been tested=
=0A=
&gt;&gt; by Nokia:=0A=
&gt;&gt; =0A=
&gt;&gt;&gt;&gt; <a rel=3D"nofollow" href=3D"http://research.nokia.com/file=
s/public/%5B16%5D_InterSpeech2011_Voice_" target=3D"_blank">http://research=
.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_</a>=0A=
&gt;&gt; =0A=
&gt;&gt;&gt;&gt; =0A=
&gt; Quality_Characterization_of_IETF_Opus_Codec.pdf=0A=
&gt;&gt;&gt;&gt; Come on, those tests are very limited compared to a formal=
=0A=
&gt;&gt;&gt;&gt; characterization test. (Example: =0A=
&gt;&gt;&gt;&gt; www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.doc=
x).=0A=
&gt;&gt;&gt;&gt; There is very little info on the material used, the enviro=
nment,=0A=
&gt;&gt;&gt;&gt; processing and scripts and so on. And there seems to be no=
 tests=0A=
&gt;&gt;&gt;&gt; whatsoever (at least not involving humans) with actual cha=
nnels=0A=
&gt;&gt;&gt;&gt; introducing jitter and losses.=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; Note: I am in no way proposing that a formal characterizat=
ion=0A=
&gt;&gt;&gt;&gt; test is needed, or even the right thing to do. It is a cos=
tly and=0A=
&gt;&gt;&gt;&gt; time consuming process, and alternative approaches could p=
rove to=0A=
&gt;&gt;&gt;&gt; be more efficient. What I am saying is that I think we sho=
uld not=0A=
&gt;&gt;&gt;&gt; mandate a codec that has neither gone through that kind of=
 formal=0A=
&gt;&gt;&gt;&gt; characterization testing nor has any field experience from=
 actual=0A=
&gt;&gt;&gt;&gt; use on a reasonable scale (covering different conditions a=
nd=0A=
&gt;&gt;&gt;&gt; device types). It just seems wrong, especially given that =
we=0A=
&gt;&gt;&gt;&gt; will soon have field experience.=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt; =0A=
&gt;&gt; Now, unlike other SDOs, the testing did not stop there. ITU-T code=
cs =0A=
&gt;&gt; generally end up being testing with something in the order of tens=
=0A=
&gt;&gt; of minutes worth of audio. In *addition* to that kind of testing,=
=0A=
&gt;&gt; Opus also had automated testing with hundreds of years worth of=0A=
&gt;&gt; audio. If anything, I think other SDOs should learn something here=
.=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt;&gt; This may very well be true. I guess this comes down to how=
 much=0A=
&gt;&gt;&gt;&gt; you trust that the quality assessment models (e.g. PEAQ, P=
OLQA)=0A=
&gt;&gt;&gt;&gt; give the same result as human test subjects would. But I t=
hink=0A=
&gt;&gt;&gt;&gt; this sounds like a really good thing.=0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt; =0A=
&gt;&gt; Jean-Marc=0A=
&gt;&gt; =0A=
&gt;&gt;&gt;&gt; =0A=
&gt;&gt;&gt; =0A=
&gt;&gt;&gt; _______________________________________________ rtcweb mailing=
=0A=
&gt;&gt;&gt; list rtcweb at ietf.org <a rel=3D"nofollow" href=3D"https://ww=
w.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/=
mailman/listinfo/rtcweb</a>=0A=
&gt; _______________________________________________=0A=
&gt; rtcweb mailing list=0A=
&gt; rtcweb at ietf.org=0A=
&gt; <a rel=3D"nofollow" href=3D"https://www.ietf.org/mailman/listinfo/rtcw=
eb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></fon=
t></pre>
<font face=3D"Courier New"><br>
</font></div>
</div>
</div>
</div>
</body>
</html>

--_000_D79146E3783B6942A3E8BC43352BBB460579F16DTK5EX14MBXC254r_--

From randell-ietf@jesup.org  Fri Aug 31 07:46:47 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 8/31/2012 9:38 AM, John Leslie wrote:
> Harald Alvestrand <harald@alvestrand.no> wrote:
>> If necessary, we should make the document more explicit to say that we
>> don't solve for use cases where less than 10s of kilobits of Internet
>> bandwidth is available...
>     +1

Agreed.

>     There are today some use-cases with less than 56Kbps; but they're
> dying, and the infrastructure for them isn't being built anywhere
> (including third-world countries, if you must ask).
>
>     I'm very glad there _are_ codecs that can operate in such an
> environment; but that's not where our effort belongs. We want something
> optimized for 5-10 years down John Leslie

If you're talking audio-only calls, I agree - but for audio+video, you 
want a lot of bandwidth available for video.  If the user has 128Kbps, I 
don't want 56K of that devoted to audio; I probably want 15-25K for 
audio (which can give good wideband or better performance with Opus), 
and the rest for video.  I probably wouldn't open audio up to 56 or 64K 
(using Opus) below 300-500K of total bandwidth, because of diminishing 
rate of improvement over ~25K.

Having a *good* low-bitrate (and adaptable) audio codec is *critical* to 
having a good solution for video.  Without that all the cases that 
involve video are blown below 100's of Kbps; with a good low-bandwidth 
codec you can do video calls with 100Kbps or even less.  This is why I 
support Opus (+G.711 for legacy interop and testing) for MTI.

-- 
Randell Jesup
randell-ietf@jesup.org


From mandyam@quicinc.com  Fri Aug 31 08:04:27 2012
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Harald,=20
I don't think your suggestion is going to work.  As Jean-Marc pointed out i=
n a separate email, RTP transmission imposes a floor of 16 kbps assuming 20=
 ms framing.  I agree with him.  Moreover even though the transmission rate=
 has to be greater than 10 kbps, the need to use a bandwidth-efficient code=
c does not go away.

If you want to eliminate the cellular QoS use case altogether, I think you =
should explicitly state so.  I think this "moves the goalposts" for the Web=
RTC effort however, and needs significantly more discussion among the group=
.

-Giri

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Thursday, August 30, 2012 11:05 PM
To: Mandyam, Giridhar
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

On 08/31/2012 02:20 AM, Mandyam, Giridhar wrote:
>> That's not what I said. I said that the *set* of MTI codecs should cover=
 the use cases.
>> We have a requirement (F21) that explicitly mentions commonly supported =
codecs in telephony services; there's no way Opus can cover that case.
> Apologies - inaccurate wording on my part.  I still think it is unrealist=
ic for any suite of MTI codecs to cover all use cases.  I think the QoS-bas=
ed use case still stands as an example of why this is unrealistic.
I still think you are constructing an unrealistic use case, and one that is=
n't in the document.
If necessary, we should make the document more explicit to say that we don'=
t solve for use cases where less than 10s of kilobits of Internet bandwidth=
 is available; I believe the speech quality of Opus is not your biggest pro=
blem in that scenario.

The purpose of an use case document is to describe the problems we're solvi=
ng so that we can separate them from the problems we don't solve.=20
I'd be fine with ruling sustained sub-10-Kbits/sec available Internet bandw=
idth explicitly out of scope.
>> You're not reading the latest draft. That section is now 4.2.6.
> The version I linked to from the RTCWEB status pages is -09, which has th=
e relevant use case as 4.2.7 (see http://tools.ietf.org/html/draft-ietf-rtc=
web-use-cases-and-requirements-09#section-4.2.7).  If the doc is rapidly ev=
olving, then the use case doc may not be suitable for evaluation of codecs =
for MTI at this point in time.
Yes, apologies; I was the one with an unrefreshed archive.
>
>> Section 4.1 of the draft defines "narrowband" to be "10s to 100s of Kbit=
s/sec".
> The full text states  "Clients can be on narrowband (10s to 100s of Kbits=
/sec)".  Does this definition refer to codecs or the underlying access tech=
nology?  I think in the context one could reasonably conclude that "narrowb=
and" refers to dial-up or low data rate cellular (e.g. GPRS, GSM HSCSD, 1X)=
.  The text in 4.1 says nothing about bandlimiting of the input voice signa=
l to a codec.  Moreover, the ensuing text states  "Clients can be on variab=
le-media-quality networks (wireless)".  To me, this means that the sub-10 k=
bps case for voice codecs is still relevant.
>
>> Respectfully ..... the slight advantage in quality that AMR-NB seems to =
offer over Opus at 8 kbps is very different from the advantage that Opus en=
joys over G.711 at 64 Kbits and above.
> I don't view the quality advantage as 6 kbps as slight, and implementers =
still have the option of using Opus at any all data rates without designati=
ng it as MTI.  Just as implementers can use AMR-WB, G.722 and so on.  Moreo=
ver, the study I cited did not consider codecs such as EVRC-B that can oper=
ate in the sub 5 kbps range.
>
>>>     The initial conclusion from my end is that using Opus when cellular=
 QoS support results in a guaranteed bit rate less than 8 kbps would not re=
sult in the user being "able to take advantage of the QoS support" as per t=
he use case.
>> Neither does it cover the case where there's not enough bits to carry an=
 IP+RTP header every 20 milliseconds. There are cases where WebRTC won't wo=
rk.
> I'm not sure what you are trying to say.  Do you believe the use of Opus =
will not satisfy the QoS use case?  Or do you believe that the QoS use case=
 is one where WebRTC will simply not work?
I believe we'll have a hard time getting the QoS use case working, not beca=
use of what you say, but because the QoS APIs on the Internet are still qui=
te difficult to use. But this is entirely orthogonal to the codec question.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From john@jlc.net  Fri Aug 31 08:12:48 2012
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From: John Leslie <john@jlc.net>
To: Randell Jesup <randell-ietf@jesup.org>
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Randell Jesup <randell-ietf@jesup.org> wrote:
> 
> If you're talking audio-only calls, I agree - but for audio+video, you 
> want a lot of bandwidth available for video.  If the user has 128Kbps, I 
> don't want 56K of that devoted to audio; I probably want 15-25K for 
> audio (which can give good wideband or better performance with Opus), 
> and the rest for video.  I probably wouldn't open audio up to 56 or 64K 
> (using Opus) below 300-500K of total bandwidth, because of diminishing 
> rate of improvement over ~25K.

   128Kbps is marginal for audio+video, yes.

   However, I don't believe it wise for us to try to standardize anything
about the audio-vs-video tradeoff.

   I will state my opinion that good audio plus one frame per second is
better in a conferencing environment than bad audio plus 128Kbps of
video; but I'd never impose that choice on anyone else.

   The point is: G.711 is suboptimal, yes, but not unusable.

   Our issue here is Mandatory-to-Implement. It is very important to
have at least one MTI audio codec. I could live with that being G.711,
because I trust the market to _actually_ implement others.

> Having a *good* low-bitrate (and adaptable) audio codec is *critical*
> to having a good solution for video.

   I'm not sure I agree, though it is certainly _very_ helpful.

> Without that all the cases that involve video are blown below 100's
> of Kbps; with a good low-bandwidth codec you can do video calls
> with 100Kbps or even less.

   I'm not sure even this will remain true for five years.

   But I am sure that the market will implement Opus or something
like it within five years without our needing to call it MTI.

> This is why I support Opus (+G.711 for legacy interop and testing)
> for MTI.

   This _is_ a good argument: I don't mean to ridicule it.

   But we need to reach closure on this. If enough people are scared
of the IPR questions of Opus, I'm willing to retreat to a SHOULD
implement Opus.

--
John Leslie <john@jlc.net>

From mandyam@quicinc.com  Fri Aug 31 08:16:42 2012
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Thread-Topic: [rtcweb] Opus over the Internet?
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Date: Fri, 31 Aug 2012 15:16:30 +0000
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Subject: Re: [rtcweb] Opus over the Internet?
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Thank you for pointing out these results.

I think these results represent a very good start, but the results as provi=
ded may not be sufficient to characterize Opus.  I've already cited the Dyn=
astat study in 3GPP2 for EVRC-B characterization in a separate email.  In a=
ddition to packet loss (in the 3GPP2 study, the analog would be frame error=
 rate), background noise conditions were varied (based on MNRU, street nois=
e, car noise).  This is the kind of testing that is typical for codecs stan=
dardized in 3GPP and 3GPP2.

There are several details missing.  For instance,

Was only office noise tested? If so, why?
How many listeners were used?
What was the mix of talkers (e.g. no. of female, no. of male)?

If you can point to more details of the testing, that would be very helpful=
 for members of the group.

-Giri Mandyam

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Koen Vos
Sent: Friday, August 31, 2012 7:38 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Opus over the Internet?

Lars Eggert wrote:
> That's all great, but is there any qualitative comparison data of the dif=
ferent codecs available?

Please have a look at the test Dynastat did with SILK: http://developer.sky=
pe.com/resources/SILKDataSheet.pdf

Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
  All changes were thoroughly tested to not degrade performance, and in mos=
t cases improved it.  So the SILK Dynastat results give a lower bound on th=
e performance of a subset of the Opus modes.

While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.  According to the test results:
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.

The packet loss in that test was maximally random (ie not bursty).  While t=
hat may not perfectly mimic real networks, I can think of no reason why the=
 results would be reversed or very different in practice.

To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.

Hope this helps,
koen.


Lars Eggert wrote:
That's all great, but is there any qualitative comparison data of the diffe=
rent codecs available?



--

Sent from a mobile device; please excuse typos.



On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin at mozilla.com> wrote=
:



> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:

>> Are there any tests that actually compare different codecs, say Opus

>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and

>> jitter? I suppose the performance is only partially related to the

>> codec, and partially to other implementation decisions, so it might

>> be difficult to compare apples-with-apples. But since people are

>> arguing that Opus is an "Internet codec", it would be actually nice

>> to see some results in this sense. Or does "Internet codec" mean

>> something else?

>

> The term "Internet codec" means different things to different people,

> but to me this means more than "having good packet loss concealment".

> Sure we have PLC (though it was never compared to G.722 or AMR-WB) and

> we also have (optional) low bit-rate embedded redundancy (also called

> FEC) and (also optional) independent frames, but that's just one aspect.

>

> One of the first things we've been asked when designing Opus was to make

> the rate *really* adaptable because we never know what kind of rates

> will be available. This not only meant having a wide range of bitrates,

> but also being able to vary in small increments. This is why Opus scales

> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with

> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in

> (on average) ~2.5 kb/s increments. The reason Opus can have more than

> 1200 possible bitrates is because on the Internet, other layers in the

> protocol stack provide octet granularity framing/sizing. We don't need

> to spend 11 bits signalling the bitrate because UDP already encodes the

> packet size.

>

> There's also practical aspects. If you look at the rates supported by

> AMR-WB, you see that *none* of them represents an integer number of

> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per

> frame. It's definitely not the only cause, but the bottom line is that

> the payload format for AMR-WB (rfc4867) is quite complex. Now compare

> this with the Opus RTP payload

> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although

> it's not complete, it's not only much simpler, but it also makes it

> possible to decode RTP packets without having even seen the SDP or any

> out-of-band signalling.

>

> People may have other definitions, but that's what *I* mean by Opus

> being an "Internet codec".

>

> Cheers,

>

>    Jean-Marc

>

>>> -----Original Message----- From: rtcweb-bounces at ietf.org

>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan Hakansson

>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:

>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of consensus on

>>> audio codecs

>>>

>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:

>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:

>>>> ...

>>

>>>>>> That is great, but sort of underlines that there would be no

>>>>>> harm in delaying the decision until there are experiences

>>>>>> made from real world use - 'cause it would not be that long

>>>>>> till that experience has been made (Markus also brought up

>>>>>> the IPR status as a reason for waiting - I have no idea how

>>>>>> long it would take to know more about that).

>>

>> As Harald is pointing out, rtcweb implementations are going to ship

>> pretty soon.

>>>>

>>>> If that is the case (and I think and hope so), why would we need

>>>> to make it MTI before seeing it in action?

>>>>

>>>> [...]

>>>>

>> Are you expecting *another* single, standardized, royalty-free codec

>> that operates over vast ranges of bitrates and operating conditions,

>> from narrowband speech to stereo music, all with low delay, coming

>> out in the next year? If not, what are you really waiting for?

>>>>

>>>> No, I'm not expecting that. I would just prefer us to see that

>>>> Opus does indeed deliver as promised before making it MTI. If it

>>>> does, fine. If not, we'd have to discuss what to do then.

>>>>

>>>> To me it is like if you're going to place a bet, for an upcoming

>>>> big race, on a horse that has never been in an actual race, but

>>>> shows great promise. If you know that the horse is going to

>>>> participate in a few practice races soon, would you not prefer to

>>>> wait and see how it fares in those before placing the bet (given

>>>> that the odds would not change)?

>>>>

>>>> Anyway, that's my view. Let's see what the chairs say.

>>

>>>>>> As Paul suggested, I was referring to the lack of formal,

>>>>>> controlled, characterization tests. That is how other SDOs do

>>>>>> it. I don't think that is the only way to do it, but I think

>>>>>> we should at least have either such tests conducted or

>>>>>> experience from deployment and use (in a wide range of

>>>>>> conditions and device types) before making it MTI.

>>

>> Opus has had "ITU-style" testing on English (

>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin

>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you

>> don't trust Google on the tests I linked to, it's also been tested

>> by Nokia:

>>

>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_

>>

>>>>

> Quality_Characterization_of_IETF_Opus_Codec.pdf

>>>> Come on, those tests are very limited compared to a formal

>>>> characterization test. (Example:

>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx<http://www=
.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx>).

>>>> There is very little info on the material used, the environment,

>>>> processing and scripts and so on. And there seems to be no tests

>>>> whatsoever (at least not involving humans) with actual channels

>>>> introducing jitter and losses.

>>>>

>>>> Note: I am in no way proposing that a formal characterization

>>>> test is needed, or even the right thing to do. It is a costly and

>>>> time consuming process, and alternative approaches could prove to

>>>> be more efficient. What I am saying is that I think we should not

>>>> mandate a codec that has neither gone through that kind of formal

>>>> characterization testing nor has any field experience from actual

>>>> use on a reasonable scale (covering different conditions and

>>>> device types). It just seems wrong, especially given that we

>>>> will soon have field experience.

>>>>

>>

>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs

>> generally end up being testing with something in the order of tens

>> of minutes worth of audio. In *addition* to that kind of testing,

>> Opus also had automated testing with hundreds of years worth of

>> audio. If anything, I think other SDOs should learn something here.

>>>>

>>>> This may very well be true. I guess this comes down to how much

>>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)

>>>> give the same result as human test subjects would. But I think

>>>> this sounds like a really good thing.

>>>>

>>

>> Jean-Marc

>>

>>>>

>>>

>>> _______________________________________________ rtcweb mailing

>>> list rtcweb at ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

> _______________________________________________

> rtcweb mailing list

> rtcweb at ietf.org

> https://www.ietf.org/mailman/listinfo/rtcweb


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thank you for pointing ou=
t these results.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I think these results rep=
resent a very good start, but the results as provided may not be sufficient=
 to characterize Opus.&nbsp; I&#8217;ve already cited the Dynastat
 study in 3GPP2 for EVRC-B characterization in a separate email.&nbsp; In a=
ddition to packet loss (in the 3GPP2 study, the analog would be frame error=
 rate), background noise conditions were varied (based on MNRU, street nois=
e, car noise).&nbsp; This is the kind of testing
 that is typical for codecs standardized in 3GPP and 3GPP2.<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">There are several details=
 missing.&nbsp; For instance,<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">Was only office noise tes=
ted? If so, why? &nbsp;&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">How many listeners were u=
sed?&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">What was the mix of talke=
rs (e.g. no. of female, no. of male)?
<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">If you can point to more =
details of the testing, that would be very helpful for members of the group=
.<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">-Giri Mandyam<o:p></o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Koen Vos<br>
<b>Sent:</b> Friday, August 31, 2012 7:38 AM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Opus over the Internet?<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">Lars Eggert wrote:</span><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:blac=
k"><o:p></o:p></span></p>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Courier New&quot;;color:black">&gt; That's all=
 great, but is there any qualitative comparison data of the different codec=
s available?
<br>
<br>
Please have a look at the test Dynastat did with SILK: <a href=3D"http://de=
veloper.skype.com/resources/SILKDataSheet.pdf" target=3D"_blank">
http://developer.skype.com/resources/SILKDataSheet.pdf</a><br>
<br>
Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
&nbsp; All changes were thoroughly tested to not degrade performance, and i=
n most cases improved it.&nbsp; So the SILK Dynastat results give a lower b=
ound on the performance of a subset of the
 Opus modes.<br>
<br>
While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.&nbsp; According to the test resul=
ts:
<br>
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance<br>
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps <br>
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.<br>
<br>
The packet loss in that test was maximally random (ie not bursty).&nbsp; Wh=
ile that may not perfectly mimic real networks, I can think of no reason wh=
y the results would be reversed or very different in practice.
<br>
<br>
To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.
<br>
<br>
Hope this helps, <br>
koen.<br>
<br>
</span><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot=
;sans-serif&quot;;color:black"><o:p></o:p></span></p>
<pre><span style=3D"color:black">Lars Eggert wrote:<br>That's all great, bu=
t is there any qualitative comparison data of the different codecs availabl=
e?<o:p></o:p></span></pre>
<pre><span style=3D"color:black"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"color:black">-- <o:p></o:p></span></pre>
<pre><span style=3D"color:black">Sent from a mobile device; please excuse t=
ypos.<o:p></o:p></span></pre>
<pre><span style=3D"color:black"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"color:black">On Aug 30, 2012, at 21:07, &quot;Jean-Marc=
 Valin&quot; &lt;jmvalin at mozilla.com&gt; wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"color:black">&gt; On 12-08-30 09:14 AM, Markus.Isomaki =
at nokia.com wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are there any tests that actually=
 compare different codecs, say Opus<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; vs. G.722 vs. AMR-WB, in typical =
Internet use, meaning some loss and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; jitter? I suppose the performance=
 is only partially related to the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; codec, and partially to other imp=
lementation decisions, so it might<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; be difficult to compare apples-wi=
th-apples. But since people are<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; arguing that Opus is an &quot;Int=
ernet codec&quot;, it would be actually nice<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; to see some results in this sense=
. Or does &quot;Internet codec&quot; mean<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; something else?<o:p></o:p></span>=
</pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; The term &quot;Internet codec&quot; m=
eans different things to different people,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; but to me this means more than &quot;=
having good packet loss concealment&quot;.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Sure we have PLC (though it was never=
 compared to G.722 or AMR-WB) and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; we also have (optional) low bit-rate =
embedded redundancy (also called<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; FEC) and (also optional) independent =
frames, but that's just one aspect.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; One of the first things we've been as=
ked when designing Opus was to make<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; the rate *really* adaptable because w=
e never know what kind of rates<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; will be available. This not only mean=
t having a wide range of bitrates,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; but also being able to vary in small =
increments. This is why Opus scales<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; from about 6 kb/s to 512 kb/s, in inc=
rements of 0.4 kb/s (one byte with<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; 20 ms frames). Compare that to AMR-WB=
, which scales from 6.6 to 23.85 in<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; (on average) ~2.5 kb/s increments. Th=
e reason Opus can have more than<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; 1200 possible bitrates is because on =
the Internet, other layers in the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; protocol stack provide octet granular=
ity framing/sizing. We don't need<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; to spend 11 bits signalling the bitra=
te because UDP already encodes the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; packet size.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; There's also practical aspects. If yo=
u look at the rates supported by<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; AMR-WB, you see that *none* of them r=
epresents an integer number of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; bytes per frame. For example, 23.85 k=
b/s is 59 bytes plus 5 bits per<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; frame. It's definitely not the only c=
ause, but the bottom line is that<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; the payload format for AMR-WB (rfc486=
7) is quite complex. Now compare<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; this with the Opus RTP payload<o:p></=
o:p></span></pre>
<pre><span style=3D"color:black">&gt; (<a href=3D"http://tools.ietf.org/htm=
l/draft-spittka-payload-rtp-opus" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-spittka-payload-rtp-opus</a>). Although<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; it's not complete, it's not only much=
 simpler, but it also makes it<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; possible to decode RTP packets withou=
t having even seen the SDP or any<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; out-of-band signalling.<o:p></o:p></s=
pan></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; People may have other definitions, bu=
t that's what *I* mean by Opus<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; being an &quot;Internet codec&quot;.<=
o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Cheers,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&nbsp;&nbsp;&nbsp; Jean-Marc<o:p></o:p=
></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; -----Original Message----- Fr=
om: rtcweb-bounces at ietf.org<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; [<a href=3D"mailto:rtcweb-bou=
nces" target=3D"_blank">mailto:rtcweb-bounces</a> at ietf.org] On Behalf Of=
 ext Stefan Hakansson<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; LK Sent: 30 August, 2012 14:5=
8 To: Jean-Marc Valin Cc:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; rtcweb at ietf.org Subject: R=
e: [rtcweb] Confirmation of consensus on<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; audio codecs<o:p></o:p></span=
></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; On 08/29/2012 03:10 PM, Jean-=
Marc Valin wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; On 08/29/2012 05:51 AM, Stefan Ha=
kansson LK wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; ...<o:p></o:p></span></pr=
e>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; That is great, bu=
t sort of underlines that there would be no<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; harm in delaying =
the decision until there are experiences<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; made from real wo=
rld use - 'cause it would not be that long<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; till that experie=
nce has been made (Markus also brought up<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; the IPR status as=
 a reason for waiting - I have no idea how<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; long it would tak=
e to know more about that).<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; As Harald is pointing out, rtcweb=
 implementations are going to ship <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; pretty soon.<o:p></o:p></span></p=
re>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; If that is the case (and =
I think and hope so), why would we need<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; to make it MTI before see=
ing it in action?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; [...]<o:p></o:p></span></=
pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are you expecting *another* singl=
e, standardized, royalty-free codec <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; that operates over vast ranges of=
 bitrates and operating conditions, <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; from narrowband speech to stereo =
music, all with low delay, coming<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; out in the next year? If not, wha=
t are you really waiting for?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; No, I'm not expecting tha=
t. I would just prefer us to see that<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Opus does indeed deliver =
as promised before making it MTI. If it<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; does, fine. If not, we'd =
have to discuss what to do then.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; To me it is like if you'r=
e going to place a bet, for an upcoming<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; big race, on a horse that=
 has never been in an actual race, but<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; shows great promise. If y=
ou know that the horse is going to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; participate in a few prac=
tice races soon, would you not prefer to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; wait and see how it fares=
 in those before placing the bet (given<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; that the odds would not c=
hange)?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Anyway, that's my view. L=
et's see what the chairs say.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; As Paul suggested=
, I was referring to the lack of formal,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; controlled, chara=
cterization tests. That is how other SDOs do<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; it. I don't think=
 that is the only way to do it, but I think<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; we should at leas=
t have either such tests conducted or<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; experience from d=
eployment and use (in a wide range of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; conditions and de=
vice types) before making it MTI.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus has had &quot;ITU-style&quot=
; testing on English ( <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <a href=3D"http://www.opus-codec.=
org/comparison/GoogleTest1.pdf" target=3D"_blank">http://www.opus-codec.org=
/comparison/GoogleTest1.pdf</a> ) and Mandarin<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; ( <a href=3D"http://www.opus-code=
c.org/comparison/GoogleTest2.pdf" target=3D"_blank">http://www.opus-codec.o=
rg/comparison/GoogleTest2.pdf</a> ). And if you <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; don't trust Google on the tests I=
 linked to, it's also been tested<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; by Nokia:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://researc=
h.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_" target=3D"_blank"=
>http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_</a>=
<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Quality_Characterization_of_IETF_Opus=
_Codec.pdf<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Come on, those tests are =
very limited compared to a formal<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization test. (E=
xample: <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://www.itu=
.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx">www.itu.int/dms_pub/i=
tu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx</a>).<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; There is very little info=
 on the material used, the environment,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; processing and scripts an=
d so on. And there seems to be no tests<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; whatsoever (at least not =
involving humans) with actual channels<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; introducing jitter and lo=
sses.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Note: I am in no way prop=
osing that a formal characterization<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; test is needed, or even t=
he right thing to do. It is a costly and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; time consuming process, a=
nd alternative approaches could prove to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; be more efficient. What I=
 am saying is that I think we should not<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; mandate a codec that has =
neither gone through that kind of formal<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization testing =
nor has any field experience from actual<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; use on a reasonable scale=
 (covering different conditions and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; device types). It just se=
ems wrong, especially given that we<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; will soon have field expe=
rience.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Now, unlike other SDOs, the testi=
ng did not stop there. ITU-T codecs <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; generally end up being testing wi=
th something in the order of tens<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; of minutes worth of audio. In *ad=
dition* to that kind of testing,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus also had automated testing w=
ith hundreds of years worth of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; audio. If anything, I think other=
 SDOs should learn something here.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; This may very well be tru=
e. I guess this comes down to how much<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; you trust that the qualit=
y assessment models (e.g. PEAQ, POLQA)<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; give the same result as h=
uman test subjects would. But I think<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; this sounds like a really=
 good thing.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Jean-Marc<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; _____________________________=
__________________ rtcweb mailing<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; list rtcweb at ietf.org <a hr=
ef=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https=
://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; _____________________________________=
__________<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; rtcweb mailing list<o:p></o:p></span>=
</pre>
<pre><span style=3D"color:black">&gt; rtcweb at ietf.org<o:p></o:p></span><=
/pre>
<pre><span style=3D"color:black">&gt; <a href=3D"https://www.ietf.org/mailm=
an/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo=
/rtcweb</a><o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/31/2012 05:03 PM, Mandyam, Giridhar wrote:
> Harald,
> I don't think your suggestion is going to work.  As Jean-Marc pointed out in a separate email, RTP transmission imposes a floor of 16 kbps assuming 20 ms framing.  I agree with him.  Moreover even though the transmission rate has to be greater than 10 kbps, the need to use a bandwidth-efficient codec does not go away.
>
> If you want to eliminate the cellular QoS use case altogether, I think you should explicitly state so.  I think this "moves the goalposts" for the WebRTC effort however, and needs significantly more discussion among the group.
>
I think you're the one moving the goalposts.... if you want to introduce 
an use case with audio communcation with an available Internet bit rate 
significantly below 10 Kbits/second, be my guest.

The QoS case is about interacting with QoS mechanisms (bandwidth 
reservation, DSCP, what have you). Reading it as mandating support for 
extremely low bandwidths is a stretch.


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Cc: Randell Jesup <randell-ietf@jesup.org>, rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On Aug 31, 2012 8:13 AM, "John Leslie" <john@jlc.net> wrote:
>
> Randell Jesup <randell-ietf@jesup.org> wrote:
> >
> > If you're talking audio-only calls, I agree - but for audio+video, you
> > want a lot of bandwidth available for video.  If the user has 128Kbps, I
> > don't want 56K of that devoted to audio; I probably want 15-25K for
> > audio (which can give good wideband or better performance with Opus),
> > and the rest for video.  I probably wouldn't open audio up to 56 or 64K
> > (using Opus) below 300-500K of total bandwidth, because of diminishing
> > rate of improvement over ~25K.
>
>    128Kbps is marginal for audio+video, yes.
>
>    However, I don't believe it wise for us to try to standardize anything
> about the audio-vs-video tradeoff.
>
>    I will state my opinion that good audio plus one frame per second is
> better in a conferencing environment than bad audio plus 128Kbps of
> video; but I'd never impose that choice on anyone else.
>
>    The point is: G.711 is suboptimal, yes, but not unusable.
>
>    Our issue here is Mandatory-to-Implement. It is very important to
> have at least one MTI audio codec. I could live with that being G.711,
> because I trust the market to _actually_ implement others.
>

Agree with the above. G.711 for mti only.

I want to push Opus and promote its use, but MTI is not the right method
for that.

CB

> > Having a *good* low-bitrate (and adaptable) audio codec is *critical*
> > to having a good solution for video.
>
>    I'm not sure I agree, though it is certainly _very_ helpful.
>
> > Without that all the cases that involve video are blown below 100's
> > of Kbps; with a good low-bandwidth codec you can do video calls
> > with 100Kbps or even less.
>
>    I'm not sure even this will remain true for five years.
>
>    But I am sure that the market will implement Opus or something
> like it within five years without our needing to call it MTI.
>
> > This is why I support Opus (+G.711 for legacy interop and testing)
> > for MTI.
>
>    This _is_ a good argument: I don't mean to ridicule it.
>
>    But we need to reach closure on this. If enough people are scared
> of the IPR questions of Opus, I'm willing to retreat to a SHOULD
> implement Opus.
>
> --
> John Leslie <john@jlc.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--f46d040838d37be4b104c8919dcf
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<p><br>
On Aug 31, 2012 8:13 AM, &quot;John Leslie&quot; &lt;<a href=3D"mailto:john=
@jlc.net">john@jlc.net</a>&gt; wrote:<br>
&gt;<br>
&gt; Randell Jesup &lt;<a href=3D"mailto:randell-ietf@jesup.org">randell-ie=
tf@jesup.org</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; If you&#39;re talking audio-only calls, I agree - but for audio+v=
ideo, you<br>
&gt; &gt; want a lot of bandwidth available for video. =A0If the user has 1=
28Kbps, I<br>
&gt; &gt; don&#39;t want 56K of that devoted to audio; I probably want 15-2=
5K for<br>
&gt; &gt; audio (which can give good wideband or better performance with Op=
us),<br>
&gt; &gt; and the rest for video. =A0I probably wouldn&#39;t open audio up =
to 56 or 64K<br>
&gt; &gt; (using Opus) below 300-500K of total bandwidth, because of dimini=
shing<br>
&gt; &gt; rate of improvement over ~25K.<br>
&gt;<br>
&gt; =A0 =A0128Kbps is marginal for audio+video, yes.<br>
&gt;<br>
&gt; =A0 =A0However, I don&#39;t believe it wise for us to try to standardi=
ze anything<br>
&gt; about the audio-vs-video tradeoff.<br>
&gt;<br>
&gt; =A0 =A0I will state my opinion that good audio plus one frame per seco=
nd is<br>
&gt; better in a conferencing environment than bad audio plus 128Kbps of<br=
>
&gt; video; but I&#39;d never impose that choice on anyone else.<br>
&gt;<br>
&gt; =A0 =A0The point is: G.711 is suboptimal, yes, but not unusable.<br>
&gt;<br>
&gt; =A0 =A0Our issue here is Mandatory-to-Implement. It is very important =
to<br>
&gt; have at least one MTI audio codec. I could live with that being G.711,=
<br>
&gt; because I trust the market to _actually_ implement others.<br>
&gt;</p>
<p>Agree with the above. G.711 for mti only.</p>
<p>I want to push Opus and promote its use, but MTI is not the right method=
 for that.</p>
<p>CB</p>
<p>&gt; &gt; Having a *good* low-bitrate (and adaptable) audio codec is *cr=
itical*<br>
&gt; &gt; to having a good solution for video.<br>
&gt;<br>
&gt; =A0 =A0I&#39;m not sure I agree, though it is certainly _very_ helpful=
.<br>
&gt;<br>
&gt; &gt; Without that all the cases that involve video are blown below 100=
&#39;s<br>
&gt; &gt; of Kbps; with a good low-bandwidth codec you can do video calls<b=
r>
&gt; &gt; with 100Kbps or even less.<br>
&gt;<br>
&gt; =A0 =A0I&#39;m not sure even this will remain true for five years.<br>
&gt;<br>
&gt; =A0 =A0But I am sure that the market will implement Opus or something<=
br>
&gt; like it within five years without our needing to call it MTI.<br>
&gt;<br>
&gt; &gt; This is why I support Opus (+G.711 for legacy interop and testing=
)<br>
&gt; &gt; for MTI.<br>
&gt;<br>
&gt; =A0 =A0This _is_ a good argument: I don&#39;t mean to ridicule it.<br>
&gt;<br>
&gt; =A0 =A0But we need to reach closure on this. If enough people are scar=
ed<br>
&gt; of the IPR questions of Opus, I&#39;m willing to retreat to a SHOULD<b=
r>
&gt; implement Opus.<br>
&gt;<br>
&gt; --<br>
&gt; John Leslie &lt;<a href=3D"mailto:john@jlc.net">john@jlc.net</a>&gt;<b=
r>
&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>
</p>

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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Hi Harald,

> I think you're the one moving the goalposts.... if you want to introduce =
an use case with audio communcation with an available Internet bit rate sig=
nificantly below 10 Kbits/second, be my guest.

What are you defining as "available internet bit rate"?  Is it the bps prov=
ided by the access technology to IP?  If so, I concede that an IP/UDP/RTP s=
ession will require more than 10 kbps from the underlying access technology=
.  There can be an explicit mention in the document as such if it helps.  I=
f you are referring to the data rate available to a WebRTC application over=
 RTP (i.e. the codec rate), then I don't agree that we should limit ourselv=
es to only 10 kbps codec settings or greater. =20

> The QoS case is about interacting with QoS mechanisms (bandwidth reservat=
ion, DSCP, what have you). Reading it as mandating support for extremely lo=
w bandwidths is a stretch.

The use case text states "In addition, the user's provider of cellular acce=
ss has QoS support enabled." QoS with respect to cellular access is well-un=
derstood in the industry, and is not restricted to reservation or DSCP sett=
ings.  This includes access-network settings for guaranteed bit rate, maxim=
um packet delay, acceptable packet loss rate.=20

To clarify, when we mention "low bandwidth" do we mean low bandwidth from a=
 codec perspective or from an access perspective?
 =20
-----Original Message-----
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Friday, August 31, 2012 8:33 AM
To: Mandyam, Giridhar
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

On 08/31/2012 05:03 PM, Mandyam, Giridhar wrote:
> Harald,
> I don't think your suggestion is going to work.  As Jean-Marc pointed out=
 in a separate email, RTP transmission imposes a floor of 16 kbps assuming =
20 ms framing.  I agree with him.  Moreover even though the transmission ra=
te has to be greater than 10 kbps, the need to use a bandwidth-efficient co=
dec does not go away.
>
> If you want to eliminate the cellular QoS use case altogether, I think yo=
u should explicitly state so.  I think this "moves the goalposts" for the W=
ebRTC effort however, and needs significantly more discussion among the gro=
up.
>
I think you're the one moving the goalposts.... if you want to introduce an=
 use case with audio communcation with an available Internet bit rate signi=
ficantly below 10 Kbits/second, be my guest.

The QoS case is about interacting with QoS mechanisms (bandwidth reservatio=
n, DSCP, what have you). Reading it as mandating support for extremely low =
bandwidths is a stretch.


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On 08/31/2012 11:12 AM, John Leslie wrote:
> The point is: G.711 is suboptimal, yes, but not unusable.

Not necessarily unusable, but it'll pretty much guarantee that nobody
will *want* to use webrtc with that kind of quality. Not to mention
*actually* unusable below 64 kb/s (+overhead).

> Our issue here is Mandatory-to-Implement. It is very important to 
> have at least one MTI audio codec. I could live with that being
> G.711, because I trust the market to _actually_ implement others.

Oh, I'm absolutely certain they'd implement other codecs. But how do
you know they'll all implement the *same* codec? That's the whole
point of MTI. Otherwise we might as well say that the standard is "tin
cans with string" and then tell people to add whatever extension is
needed to get decent quality. That's not a standard and that's no
better than the current situation with plugins.

The current situation is this: there's existing platforms from Skype,
Google, and others that provide wideband and super-wideband speech at
low enough rates for pretty much everyone. And if you use (e.g.)
Skype, you know that you can always get super-wideband quality
(assuming enough bandwidth). What some here are proposing is a huge
step backwards. Why would I use WebRTC over Skype/Google if I'm likely
to end up with high bit-rate narrowband just because the person I'm
talking to uses a different vendor. Let's at least make sure that
WebRTC will be better or equal to other offerings at all time.
Otherwise what's the point? And it's not like this is really hard, is
it? We have Opus here that's standardized by the IETF (well, in a few
days anyway), and it's an improvement on top of what Skype currently
ships. I've yet to see a valid reason to settle for less.

	Jean-Marc
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From roman@telurix.com  Fri Aug 31 10:23:22 2012
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Date: Fri, 31 Aug 2012 20:23:15 +0300
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From: Roman Shpount <roman@telurix.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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--047d7b10cdd52644f904c8930cde
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One issue I want to bring into consideration is computation complexity of
Opus. It is normally acceptable for end user devices in point-to-point
calls (I do not want to start all the mobile device CPU vs battery live
discussion. It is irrelevant), but in cases where you need to process all
the calls on some sort of central server (such as conferencing, or virtual
reality, or massive multi-player games) the additional processing cost of
supporting Opus can be significant. In such cases G.722 (ideally in
combination with  PLC and CN to save bandwidth) can be a valid compromise,
since it can provide quality better then G.711 with CPU processing
requirements order of magnitude less then Opus.

My point is I see no downside of making G.722 an MTI codec in addition to
G.711 and Opus. Additional code is minimal and well understood, there are
no royalties. On the upside there are at least two significant use cases
(conferencing or other central audio processing scenarious and interop with
enterprise equipment) that will justify this.
_____________
Roman Shpount


On Fri, Aug 31, 2012 at 7:34 PM, Jean-Marc Valin <jmvalin@mozilla.com>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 08/31/2012 11:12 AM, John Leslie wrote:
> > The point is: G.711 is suboptimal, yes, but not unusable.
>
> Not necessarily unusable, but it'll pretty much guarantee that nobody
> will *want* to use webrtc with that kind of quality. Not to mention
> *actually* unusable below 64 kb/s (+overhead).
>
> > Our issue here is Mandatory-to-Implement. It is very important to
> > have at least one MTI audio codec. I could live with that being
> > G.711, because I trust the market to _actually_ implement others.
>
> Oh, I'm absolutely certain they'd implement other codecs. But how do
> you know they'll all implement the *same* codec? That's the whole
> point of MTI. Otherwise we might as well say that the standard is "tin
> cans with string" and then tell people to add whatever extension is
> needed to get decent quality. That's not a standard and that's no
> better than the current situation with plugins.
>
> The current situation is this: there's existing platforms from Skype,
> Google, and others that provide wideband and super-wideband speech at
> low enough rates for pretty much everyone. And if you use (e.g.)
> Skype, you know that you can always get super-wideband quality
> (assuming enough bandwidth). What some here are proposing is a huge
> step backwards. Why would I use WebRTC over Skype/Google if I'm likely
> to end up with high bit-rate narrowband just because the person I'm
> talking to uses a different vendor. Let's at least make sure that
> WebRTC will be better or equal to other offerings at all time.
> Otherwise what's the point? And it's not like this is really hard, is
> it? We have Opus here that's standardized by the IETF (well, in a few
> days anyway), and it's an improvement on top of what Skype currently
> ships. I've yet to see a valid reason to settle for less.
>
>         Jean-Marc
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> _______________________________________________
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>

--047d7b10cdd52644f904c8930cde
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

One issue I want to bring into consideration is computation complexity of O=
pus. It is normally acceptable for end user devices in point-to-point calls=
 (I do not want to start all the mobile device CPU vs battery live discussi=
on. It is irrelevant), but in cases where you need to process all the calls=
 on some sort of central server (such as conferencing, or virtual reality, =
or massive multi-player games) the additional processing cost of supporting=
 Opus can be significant. In such cases G.722 (ideally in combination with=
=A0
PLC=A0and=A0CN to save bandwidth) can be a valid compromise, since it can p=
rovide quality better then G.711 with CPU processing requirements order of =
magnitude less then Opus.<div><br></div><div>My point is I see no downside =
of making G.722 an MTI codec in addition to G.711 and Opus. Additional code=
 is minimal and well understood, there are no royalties. On the upside ther=
e are at least two significant use cases (conferencing or other central aud=
io processing scenarious and interop with enterprise equipment) that will j=
ustify this.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Fri, Aug 31, 2012 at 7:34 PM, Jean-Ma=
rc Valin <span dir=3D"ltr">&lt;<a href=3D"mailto:jmvalin@mozilla.com" targe=
t=3D"_blank">jmvalin@mozilla.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<div class=3D"im">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
</div><div class=3D"im">On 08/31/2012 11:12 AM, John Leslie wrote:<br>
&gt; The point is: G.711 is suboptimal, yes, but not unusable.<br>
<br>
</div>Not necessarily unusable, but it&#39;ll pretty much guarantee that no=
body<br>
will *want* to use webrtc with that kind of quality. Not to mention<br>
*actually* unusable below 64 kb/s (+overhead).<br>
<div class=3D"im"><br>
&gt; Our issue here is Mandatory-to-Implement. It is very important to<br>
&gt; have at least one MTI audio codec. I could live with that being<br>
&gt; G.711, because I trust the market to _actually_ implement others.<br>
<br>
</div>Oh, I&#39;m absolutely certain they&#39;d implement other codecs. But=
 how do<br>
you know they&#39;ll all implement the *same* codec? That&#39;s the whole<b=
r>
point of MTI. Otherwise we might as well say that the standard is &quot;tin=
<br>
cans with string&quot; and then tell people to add whatever extension is<br=
>
needed to get decent quality. That&#39;s not a standard and that&#39;s no<b=
r>
better than the current situation with plugins.<br>
<br>
The current situation is this: there&#39;s existing platforms from Skype,<b=
r>
Google, and others that provide wideband and super-wideband speech at<br>
low enough rates for pretty much everyone. And if you use (e.g.)<br>
Skype, you know that you can always get super-wideband quality<br>
(assuming enough bandwidth). What some here are proposing is a huge<br>
step backwards. Why would I use WebRTC over Skype/Google if I&#39;m likely<=
br>
to end up with high bit-rate narrowband just because the person I&#39;m<br>
talking to uses a different vendor. Let&#39;s at least make sure that<br>
WebRTC will be better or equal to other offerings at all time.<br>
Otherwise what&#39;s the point? And it&#39;s not like this is really hard, =
is<br>
it? We have Opus here that&#39;s standardized by the IETF (well, in a few<b=
r>
days anyway), and it&#39;s an improvement on top of what Skype currently<br=
>
ships. I&#39;ve yet to see a valid reason to settle for less.<br>
<br>
=A0 =A0 =A0 =A0 Jean-Marc<br>
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-----END PGP SIGNATURE-----<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>

--047d7b10cdd52644f904c8930cde--

From mandyam@quicinc.com  Fri Aug 31 10:25:36 2012
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From: "Mandyam, Giridhar" <mandyam@quicinc.com>
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Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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>Here's the thing. I'm not trying to prevent you from including AMR-NB or a=
ny other codec. If all you have is 6 kb/s and you have AMR-NB on both ends,=
 then by all means use that. But what if that's not the case and only one s=
ide supports it? If only G.711 is >MTI, then the negotiation *fails*. And i=
t won't just fail at 6 kb/s. It'll fail all the way up to 64 kb/s and that'=
s unacceptable. OTOH, if you have Opus as MTI, you'll still be able to talk=
, which is big improvement, even if the quality isn't nearly as good as AMR=
->NB or EVRC-B at these ridiculously low bitrates. Essentially, I've yet to=
 understand your reasoning that because Opus is "only" optimal for 99% of t=
he use cases, then we should use something (G.711) that's highly suboptimal=
 in all use cases.

I'm not sure I understand your logic either.  You could make the same argum=
ent in favor of having AMR-NB or EVRC-B as MTI.  Why not do so and cover 10=
0% of the use cases?

In addition, I don't view all use cases as being of equal importance (at le=
ast with initial implementations).  Conversational voice over cellular must=
 work for WebRTC. =20

Moreover, I don't believe the standardization process for any codec suitabl=
e for conversational voice is complete without a characterization under los=
sy conditions and with human listeners (true MOS score).  This is the appro=
ach that standards bodies such as 3GPP and 3GPP2 have taken towards codec s=
tandardization.  I think the studies you pointed to in a separate email (fr=
om Google and Nokia) are a good start, but until we see testing like the Dy=
nastat study I referenced in my previous email then to say Opus satisfies a=
ny of the conversational voice use cases is premature.

Implementers that choose not to use the best possible codec under common op=
erating conditions (e.g. ridiculously low bitrates, which happens frequentl=
y in cellular environments) will suffer in the marketplace as a result.  Bu=
t this is not an issue for standardization.

> Sure, go as low as you like. There's even codecs that work at 2 kb/s and =
below if you like. If both sides support it, it's all good. But when all we=
 have in common are the MTI codecs, then please let's make webrtc not suck =
or (worse) fail completely.

No, it's not good.  2 kb/s codecs such as MELP are not toll quality.  Intel=
ligibility alone may be OK for military applications, but toll quality is a=
 requirement for consumer apps.

> I'm sure most people prefer 1) and I still don't understand why you would=
 prefer option 2). Option 2) essentially means "we have a really crappy sta=
ndard which can only work decently if you have extensions".

Implementers are expected to support more codecs than just whatever is MTI.=
  As Ted mentioned in another email, one of the goals of RTCWeb is promotin=
g support for native codecs:

At 8:39 PM -0700 8/21/12, Ted Hardie <ted.ietf at gmail.com> wrote:
>  the fundamental design of RTCWEB allows for the  negotiation of any=20
> codec mutually supported.  The decision to chose a =20
> Mandatory-to-implement was not made to eliminate other choices, but to =20
> eliminate interoperability failures by ensuring that at least one =20
> common codec is always available.

Codecs should be MTI only to prevent negotiation failure, not to guarantee =
ideal performance, especially in light of Ted's next statement:

At 8:39 PM -0700 8/21/12, Ted Hardie <ted.ietf at gmail.com> wrote:
>  the group presumes that codec support does not  come from the=20
> downloadable Javascript application, but from the  application=20
> environment into which it is downloaded (commonly a  browser or mobile=20
> environment using similar technology).  Promoting  support for those=20
> environments to have access to codecs supported by  the underlying=20
> hardware is the best way to further this goal, at least  in my=20
> personal opinion.

Moreover, claiming that access to codecs such as AMR is only possible with =
software extensions ignores the installed base in the handheld market.  Dev=
elopers can certainly have access to such implementations, and shouldn't ha=
ve to burden themselves with implementing more codecs than minimally necess=
ary because of MTI requirements.

-----Original Message-----
From: Jean-Marc Valin [mailto:jmvalin@mozilla.com]=20
Sent: Thursday, August 30, 2012 6:43 PM
To: Mandyam, Giridhar
Cc: Harald Alvestrand; DRAGE, Keith (Keith); rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 08/30/2012 08:18 PM, Mandyam, Giridhar wrote:
> I don't follow. If both parties in the call support a codec suitable=20
> for 6 kb/s (e.g. AMR-NB) then there should be no negotiation failure.=20
> MTI still allows for non-MTI codecs to be used on a WebRTC call.  The=20
> point I was trying to make was that Opus does not satisfy the use case=20
> as specified.  I never said that
> G.711 satisfied this use case. I believe it is unrealistic to require=20
> that the set of MTI audio codecs (whatever they may be) satisfy every=20
> single use case as currently outlined.

Here's the thing. I'm not trying to prevent you from including AMR-NB or an=
y other codec. If all you have is 6 kb/s and you have AMR-NB on both ends, =
then by all means use that. But what if that's not the case and only one si=
de supports it? If only G.711 is MTI, then the negotiation *fails*. And it =
won't just fail at 6 kb/s. It'll fail all the way up to 64 kb/s and that's =
unacceptable. OTOH, if you have Opus as MTI, you'll still be able to talk, =
which is big improvement, even if the quality isn't nearly as good as AMR-N=
B or EVRC-B at these ridiculously low bitrates. Essentially, I've yet to un=
derstand your reasoning that because Opus is "only" optimal for 99% of the =
use cases, then we should use something (G.711) that's highly suboptimal in=
 all use cases.

The way I see this shaping up, it seems highly likely that

> All the more reason to achieve toll quality at codec data rates below=20
> 8 kbps when operating in a cellular environment.

Sure, go as low as you like. There's even codecs that work at 2 kb/s and be=
low if you like. If both sides support it, it's all good. But when all we h=
ave in common are the MTI codecs, then please let's make webrtc not suck or=
 (worse) fail completely.

> I've already stated my position on MTI audio codecs on this mailing=20
> list.  I never advocated making every codec that is optimal given a=20
> particular operating condition as MTI.  I also think the performance=20
> loss due to the use of Opus at sub 10 kbps is not trivial, and it is=20
> worth leveraging other codecs at these lower data rates.

So here's the options we're looking at in the cases where all you have in c=
ommon are MTI codecs (and it seems that those will be common):
1) Optimal quality at almost all rates, but sub-optimal below 10 kb/s; or
2) Highly sub-optimal at 64 kb/s and above, with completely failure below 6=
4 kb/s.

I'm sure most people prefer 1) and I still don't understand why you would p=
refer option 2). Option 2) essentially means "we have a really crappy stand=
ard which can only work decently if you have extensions".

	Jean-Marc

> -----Original Message----- From: Jean-Marc Valin=20
> [mailto:jmvalin@mozilla.com] Sent: Thursday, August 30, 2012 12:40 PM=20
> To: Mandyam, Giridhar Cc: Harald Alvestrand; DRAGE, Keith (Keith);=20
> rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>=20
> On 12-08-30 03:13 PM, Mandyam, Giridhar wrote:
>> The study available at
>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice
>> _
>>
>>
>>=20
Quality_Characterization_of_IETF_Opus_Codec.pdf
>> clearly shows a significant advantage in MOS scores at 5.9 kbps for=20
>> AMR-NB as compared to Opus (Silk).  The initial conclusion from my=20
>> end is that using Opus when cellular QoS support results in a=20
>> guaranteed bit rate less than 8 kbps would not result in the user=20
>> being "able to take advantage of the QoS support" as per the use=20
>> case.
>=20
> So because Opus is better except around 6 kb/s means that we have=20
> negotiation failure and that we should instead use G.711 at 64 kb/s?=20
> Also, keep in mind that browser-to-browser means RTP, which typically=20
> means 16 kb/s worth of overhead anyway (with 20 ms frames). Even=20
> though Opus is not optimal at 6 kb/s, we still have something that=20
> works and we avoid interop failure. And what are the alternatives=20
> anyway? If we only decide on G.711, then we not only have=20
> embarrassingly bad quality, but we also have interop failure below 64=20
> kb/s. If you want guaranteed good quality without Opus, you can also=20
> make AMR-NB, AMR-WB, G.722.1C, AAC-LD, HE-AAC, and AAC-LC all MTI.=20
> Then you'll have (on average) quality that's almost as good as Opus=20
> alone can do, you'll cover almost as many use cases, and as a bonus,=20
> you get all the negotiation and switching nightmares involved.
>=20
> Jean-Marc
>=20
>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
>> Sent: Thursday, August 30, 2012 5:51 AM To: DRAGE, Keith (Keith)
>> Cc: rtcweb@ietf.org Subject: Re: [rtcweb] RTCWEB needs an Internet=20
>> Codec
>>=20
>> On 08/30/2012 02:38 PM, DRAGE, Keith (Keith) wrote:
>>> Surely that argument extends to every possible codec.
>> The set of MTI codecs should cover the set of use cases that have=20
>> been identified. Additional MTI codecs that don't cover new use cases=20
>> shouldn't be added.
>>=20
>> I believe the use case "distributed music band" in=20
>> draft-ietf-rtcweb-use-cases-and-requirements can't be satisified with=20
>> G.711.
>>=20
>>>=20
>>> If the codecs supported by both endpoints do not supply the=20
>>> characteristics needed for the communication, then you will get =20
>>> interoperability failure.
>>>=20
>>> If I want more bandwidth or quality than OPUS supplies, then I will=20
>>> also get interoperability failure.
>>>=20
>>> Keith
>>>=20
>>>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand=20
>>>> Sent: 30 August 2012 11:56 To: rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>>=20
>>>> The counterpoint is that having G.711 as the only MTI means that=20
>>>> you get interoperability failure for any application in which G.711=20
>>>> sound quality is not acceptable.
>>>>=20
>>>> This includes all applications involving music.
>>>>=20
>>>> On 08/29/2012 06:30 PM, Randall Gellens wrote:
>>>>> At 12:13 AM -0400 8/29/12, Roman Shpount wrote:
>>>>>=20
>>>>>> I would argue G.711 should be the MTI codec. The rest can be left=20
>>>>>> up to browser implementers.
>>>>> I agree.
>>>>>=20
>>>>>> We can argue all we want, but the best royalty free low bitrate=20
>>>>>> codec available will be the one everybody supports.
>>>>> Very good point, and why we should mandate only one audio codec.
>>>>>=20
>>>> _______________________________________________ rtcweb mailing list=20
>>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>> _______________________________________________ rtcweb mailing list=20
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>  _______________________________________________ rtcweb mailing list=20
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>=20

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On 31 Aug 2012, at 10:25, Mandyam, Giridhar wrote:

> In addition, I don't view all use cases as being of equal importance =
(at least with initial implementations).  Conversational voice over =
cellular must work for WebRTC.=20
All use cases are equal, but some are more equal than others ?

I think it is a _huge_ mistake to assume we know what will be important =
use cases for webRTC.=20
The use case you cite is _very_ well served by adding a tel: url to a =
mobile web page and placing a 2g gsm call.

Tim.


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 31 Aug 2012, at 10:25, Mandyam, Giridhar =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-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: 0px; font-size: medium; ">In addition, =
I don't view all use cases as being of equal importance (at least with =
initial implementations). &nbsp;Conversational voice over cellular must =
work for WebRTC.<span =
class=3D"Apple-converted-space">&nbsp;</span></span></blockquote></div>All=
 use cases are equal, but some are more equal than others =
?<div><br></div><div>I think it is a _huge_ mistake to assume we know =
what will be important use cases for webRTC.&nbsp;</div><div>The use =
case you cite is _very_ well served by adding a tel: url to a mobile web =
page and placing a 2g gsm =
call.</div><div><br></div><div>Tim.</div><div><br></div></body></html>=

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From koen.vos@skype.net  Fri Aug 31 13:30:23 2012
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From: Koen Vos <koen.vos@skype.net>
To: "Mandyam, Giridhar" <mandyam@quicinc.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Opus over the Internet?
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Hi Giri,


Dynastat's subjective test involved the following design parameters:

- subjects: 32, four panels of eight subjects, native speakers of North Ame=
rican English

- randomizations: one per listening panel
- talkers: eight (four males, four females), native speakers of North Ameri=
can English
- speech samples: four 8 sec. sentence-pairs per talker
- experimental design: partially-balanced, randomized-blocks
Dynastat constructed the randomizations (i.e., pseudo-randomized presentati=
on sequences) for each listening panel. Each randomization included 256 tri=
als (i.e., 8 talkers x 32 test conditions).

We tested only office noise at 15 dB SNR because we didn't have more condit=
ions available.

As web browsers are commonly used in an office, knowledge of a codec's perf=
ormance in that kind of environment would seem helpful to this working grou=
p.  Do I understand correctly that the 3GPP and 3GPP2 tests did not include=
 any office noise conditions?

best,
koen.

________________________________
From: Mandyam, Giridhar [mandyam@quicinc.com]
Sent: Friday, August 31, 2012 8:16 AM
To: Koen Vos; rtcweb@ietf.org
Subject: RE: [rtcweb] Opus over the Internet?

Thank you for pointing out these results.

I think these results represent a very good start, but the results as provi=
ded may not be sufficient to characterize Opus.  I=92ve already cited the D=
ynastat study in 3GPP2 for EVRC-B characterization in a separate email.  In=
 addition to packet loss (in the 3GPP2 study, the analog would be frame err=
or rate), background noise conditions were varied (based on MNRU, street no=
ise, car noise).  This is the kind of testing that is typical for codecs st=
andardized in 3GPP and 3GPP2.

There are several details missing.  For instance,

Was only office noise tested? If so, why?
How many listeners were used?
What was the mix of talkers (e.g. no. of female, no. of male)?

If you can point to more details of the testing, that would be very helpful=
 for members of the group.

-Giri Mandyam

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Koen Vos
Sent: Friday, August 31, 2012 7:38 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Opus over the Internet?

Lars Eggert wrote:
> That's all great, but is there any qualitative comparison data of the dif=
ferent codecs available?

Please have a look at the test Dynastat did with SILK: http://developer.sky=
pe.com/resources/SILKDataSheet.pdf

Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
  All changes were thoroughly tested to not degrade performance, and in mos=
t cases improved it.  So the SILK Dynastat results give a lower bound on th=
e performance of a subset of the Opus modes.

While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.  According to the test results:
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.

The packet loss in that test was maximally random (ie not bursty).  While t=
hat may not perfectly mimic real networks, I can think of no reason why the=
 results would be reversed or very different in practice.

To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.

Hope this helps,
koen.


Lars Eggert wrote:
That's all great, but is there any qualitative comparison data of the diffe=
rent codecs available?



--

Sent from a mobile device; please excuse typos.



On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin at mozilla.com> wrote=
:



> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:

>> Are there any tests that actually compare different codecs, say Opus

>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and

>> jitter? I suppose the performance is only partially related to the

>> codec, and partially to other implementation decisions, so it might

>> be difficult to compare apples-with-apples. But since people are

>> arguing that Opus is an "Internet codec", it would be actually nice

>> to see some results in this sense. Or does "Internet codec" mean

>> something else?

>

> The term "Internet codec" means different things to different people,

> but to me this means more than "having good packet loss concealment".

> Sure we have PLC (though it was never compared to G.722 or AMR-WB) and

> we also have (optional) low bit-rate embedded redundancy (also called

> FEC) and (also optional) independent frames, but that's just one aspect.

>

> One of the first things we've been asked when designing Opus was to make

> the rate *really* adaptable because we never know what kind of rates

> will be available. This not only meant having a wide range of bitrates,

> but also being able to vary in small increments. This is why Opus scales

> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with

> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in

> (on average) ~2.5 kb/s increments. The reason Opus can have more than

> 1200 possible bitrates is because on the Internet, other layers in the

> protocol stack provide octet granularity framing/sizing. We don't need

> to spend 11 bits signalling the bitrate because UDP already encodes the

> packet size.

>

> There's also practical aspects. If you look at the rates supported by

> AMR-WB, you see that *none* of them represents an integer number of

> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per

> frame. It's definitely not the only cause, but the bottom line is that

> the payload format for AMR-WB (rfc4867) is quite complex. Now compare

> this with the Opus RTP payload

> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although

> it's not complete, it's not only much simpler, but it also makes it

> possible to decode RTP packets without having even seen the SDP or any

> out-of-band signalling.

>

> People may have other definitions, but that's what *I* mean by Opus

> being an "Internet codec".

>

> Cheers,

>

>    Jean-Marc

>

>>> -----Original Message----- From: rtcweb-bounces at ietf.org

>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan Hakansson

>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:

>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of consensus on

>>> audio codecs

>>>

>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:

>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:

>>>> ...

>>

>>>>>> That is great, but sort of underlines that there would be no

>>>>>> harm in delaying the decision until there are experiences

>>>>>> made from real world use - 'cause it would not be that long

>>>>>> till that experience has been made (Markus also brought up

>>>>>> the IPR status as a reason for waiting - I have no idea how

>>>>>> long it would take to know more about that).

>>

>> As Harald is pointing out, rtcweb implementations are going to ship

>> pretty soon.

>>>>

>>>> If that is the case (and I think and hope so), why would we need

>>>> to make it MTI before seeing it in action?

>>>>

>>>> [...]

>>>>

>> Are you expecting *another* single, standardized, royalty-free codec

>> that operates over vast ranges of bitrates and operating conditions,

>> from narrowband speech to stereo music, all with low delay, coming

>> out in the next year? If not, what are you really waiting for?

>>>>

>>>> No, I'm not expecting that. I would just prefer us to see that

>>>> Opus does indeed deliver as promised before making it MTI. If it

>>>> does, fine. If not, we'd have to discuss what to do then.

>>>>

>>>> To me it is like if you're going to place a bet, for an upcoming

>>>> big race, on a horse that has never been in an actual race, but

>>>> shows great promise. If you know that the horse is going to

>>>> participate in a few practice races soon, would you not prefer to

>>>> wait and see how it fares in those before placing the bet (given

>>>> that the odds would not change)?

>>>>

>>>> Anyway, that's my view. Let's see what the chairs say.

>>

>>>>>> As Paul suggested, I was referring to the lack of formal,

>>>>>> controlled, characterization tests. That is how other SDOs do

>>>>>> it. I don't think that is the only way to do it, but I think

>>>>>> we should at least have either such tests conducted or

>>>>>> experience from deployment and use (in a wide range of

>>>>>> conditions and device types) before making it MTI.

>>

>> Opus has had "ITU-style" testing on English (

>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin

>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you

>> don't trust Google on the tests I linked to, it's also been tested

>> by Nokia:

>>

>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_

>>

>>>>

> Quality_Characterization_of_IETF_Opus_Codec.pdf

>>>> Come on, those tests are very limited compared to a formal

>>>> characterization test. (Example:

>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx<http://www=
.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx>).

>>>> There is very little info on the material used, the environment,

>>>> processing and scripts and so on. And there seems to be no tests

>>>> whatsoever (at least not involving humans) with actual channels

>>>> introducing jitter and losses.

>>>>

>>>> Note: I am in no way proposing that a formal characterization

>>>> test is needed, or even the right thing to do. It is a costly and

>>>> time consuming process, and alternative approaches could prove to

>>>> be more efficient. What I am saying is that I think we should not

>>>> mandate a codec that has neither gone through that kind of formal

>>>> characterization testing nor has any field experience from actual

>>>> use on a reasonable scale (covering different conditions and

>>>> device types). It just seems wrong, especially given that we

>>>> will soon have field experience.

>>>>

>>

>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs

>> generally end up being testing with something in the order of tens

>> of minutes worth of audio. In *addition* to that kind of testing,

>> Opus also had automated testing with hundreds of years worth of

>> audio. If anything, I think other SDOs should learn something here.

>>>>

>>>> This may very well be true. I guess this comes down to how much

>>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)

>>>> give the same result as human test subjects would. But I think

>>>> this sounds like a really good thing.

>>>>

>>

>> Jean-Marc

>>

>>>>

>>>

>>> _______________________________________________ rtcweb mailing

>>> list rtcweb at ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

> _______________________________________________

> rtcweb mailing list

> rtcweb at ietf.org

> https://www.ietf.org/mailman/listinfo/rtcweb


--_000_D79146E3783B6942A3E8BC43352BBB460579F1EDTK5EX14MBXC254r_
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<body ocsi=3D"0" fpstyle=3D"1" vlink=3D"purple" lang=3D"EN-US" link=3D"blue=
">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">Hi Giri,<br>
<br>
<p style=3D"margin-bottom: 0in" align=3D"JUSTIFY">Dynastat's subjective tes=
t involved the following design parameters:</p>
<p style=3D"margin-bottom: 0in" align=3D"JUSTIFY">- subjects: 32, four pane=
ls of eight subjects, native speakers of North American English</p>
- randomizations: one per listening panel<br>
- talkers: eight (four males, four females), native speakers of North Ameri=
can English<br>
- speech samples: four 8 sec. sentence-pairs per talker<br>
- experimental design: partially-balanced, randomized-blocks<br>
Dynastat constructed the randomizations (i.e., pseudo-randomized presentati=
on sequences) for each listening panel. Each randomization included 256 tri=
als (i.e., 8 talkers x 32 test conditions).
<br>
<br>
We tested only office noise at 15 dB SNR because we didn't have more condit=
ions available.&nbsp;
<br>
<br>
As web browsers are commonly used in an office, knowledge of a codec's perf=
ormance in that kind of environment would seem helpful to this working grou=
p.&nbsp; Do I understand correctly that the 3GPP and 3GPP2 tests did not in=
clude any office noise conditions?&nbsp;
<br>
<br>
best,<br>
koen.<br>
<br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div style=3D"direction: ltr;" id=3D"divRpF668266"><font color=3D"#000000" =
face=3D"Tahoma" size=3D"2"><b>From:</b> Mandyam, Giridhar [mandyam@quicinc.=
com]<br>
<b>Sent:</b> Friday, August 31, 2012 8:16 AM<br>
<b>To:</b> Koen Vos; rtcweb@ietf.org<br>
<b>Subject:</b> RE: [rtcweb] Opus over the Internet?<br>
</font><br>
</div>
<div></div>
<div>
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Thank you for pointing =
out these results.</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">I think these results r=
epresent a very good start, but the results as provided may not be sufficie=
nt to characterize Opus.&nbsp; I=92ve already cited the Dynastat
 study in 3GPP2 for EVRC-B characterization in a separate email.&nbsp; In a=
ddition to packet loss (in the 3GPP2 study, the analog would be frame error=
 rate), background noise conditions were varied (based on MNRU, street nois=
e, car noise).&nbsp; This is the kind of testing
 that is typical for codecs standardized in 3GPP and 3GPP2.</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">There are several detai=
ls missing.&nbsp; For instance,</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Was only office noise t=
ested? If so, why? &nbsp;&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">How many listeners were=
 used?&nbsp;
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">What was the mix of tal=
kers (e.g. no. of female, no. of male)?
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">If you can point to mor=
e details of the testing, that would be very helpful for members of the gro=
up.</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">-Giri Mandyam</span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<div>
<div style=3D"border:none; border-top:solid #B5C4DF 1.0pt; padding:3.0pt 0i=
n 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt; font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:10.0pt; font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb=
-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Koen Vos<br>
<b>Sent:</b> Friday, August 31, 2012 7:38 AM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Opus over the Internet?</span></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;</p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt; font-family:&quot;C=
ourier New&quot;; color:black">Lars Eggert wrote:</span><span style=3D"font=
-size:10.0pt; font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;; color:=
black"></span></p>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt; font-family:&quot;Courier New&quot;; color:black">&gt; That's a=
ll great, but is there any qualitative comparison data of the different cod=
ecs available?
<br>
<br>
Please have a look at the test Dynastat did with SILK: <a href=3D"http://de=
veloper.skype.com/resources/SILKDataSheet.pdf" target=3D"_blank">
http://developer.skype.com/resources/SILKDataSheet.pdf</a><br>
<br>
Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
&nbsp; All changes were thoroughly tested to not degrade performance, and i=
n most cases improved it.&nbsp; So the SILK Dynastat results give a lower b=
ound on the performance of a subset of the
 Opus modes.<br>
<br>
While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.&nbsp; According to the test resul=
ts:
<br>
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance<br>
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps <br>
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.<br>
<br>
The packet loss in that test was maximally random (ie not bursty).&nbsp; Wh=
ile that may not perfectly mimic real networks, I can think of no reason wh=
y the results would be reversed or very different in practice.
<br>
<br>
To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.
<br>
<br>
Hope this helps, <br>
koen.<br>
<br>
</span><span style=3D"font-size:10.0pt; font-family:&quot;Tahoma&quot;,&quo=
t;sans-serif&quot;; color:black"></span></p>
<pre><span style=3D"color:black">Lars Eggert wrote:<br>That's all great, bu=
t is there any qualitative comparison data of the different codecs availabl=
e?</span></pre>
<pre><span style=3D"color:black">&nbsp;</span></pre>
<pre><span style=3D"color:black">-- </span></pre>
<pre><span style=3D"color:black">Sent from a mobile device; please excuse t=
ypos.</span></pre>
<pre><span style=3D"color:black">&nbsp;</span></pre>
<pre><span style=3D"color:black">On Aug 30, 2012, at 21:07, &quot;Jean-Marc=
 Valin&quot; &lt;jmvalin at mozilla.com&gt; wrote:</span></pre>
<pre><span style=3D"color:black">&nbsp;</span></pre>
<pre><span style=3D"color:black">&gt; On 12-08-30 09:14 AM, Markus.Isomaki =
at nokia.com wrote:</span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are there any tests that actually=
 compare different codecs, say Opus</span></pre>
<pre><span style=3D"color:black">&gt;&gt; vs. G.722 vs. AMR-WB, in typical =
Internet use, meaning some loss and</span></pre>
<pre><span style=3D"color:black">&gt;&gt; jitter? I suppose the performance=
 is only partially related to the</span></pre>
<pre><span style=3D"color:black">&gt;&gt; codec, and partially to other imp=
lementation decisions, so it might</span></pre>
<pre><span style=3D"color:black">&gt;&gt; be difficult to compare apples-wi=
th-apples. But since people are</span></pre>
<pre><span style=3D"color:black">&gt;&gt; arguing that Opus is an &quot;Int=
ernet codec&quot;, it would be actually nice</span></pre>
<pre><span style=3D"color:black">&gt;&gt; to see some results in this sense=
. Or does &quot;Internet codec&quot; mean</span></pre>
<pre><span style=3D"color:black">&gt;&gt; something else?</span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt; The term &quot;Internet codec&quot; m=
eans different things to different people,</span></pre>
<pre><span style=3D"color:black">&gt; but to me this means more than &quot;=
having good packet loss concealment&quot;.</span></pre>
<pre><span style=3D"color:black">&gt; Sure we have PLC (though it was never=
 compared to G.722 or AMR-WB) and</span></pre>
<pre><span style=3D"color:black">&gt; we also have (optional) low bit-rate =
embedded redundancy (also called</span></pre>
<pre><span style=3D"color:black">&gt; FEC) and (also optional) independent =
frames, but that's just one aspect.</span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt; One of the first things we've been as=
ked when designing Opus was to make</span></pre>
<pre><span style=3D"color:black">&gt; the rate *really* adaptable because w=
e never know what kind of rates</span></pre>
<pre><span style=3D"color:black">&gt; will be available. This not only mean=
t having a wide range of bitrates,</span></pre>
<pre><span style=3D"color:black">&gt; but also being able to vary in small =
increments. This is why Opus scales</span></pre>
<pre><span style=3D"color:black">&gt; from about 6 kb/s to 512 kb/s, in inc=
rements of 0.4 kb/s (one byte with</span></pre>
<pre><span style=3D"color:black">&gt; 20 ms frames). Compare that to AMR-WB=
, which scales from 6.6 to 23.85 in</span></pre>
<pre><span style=3D"color:black">&gt; (on average) ~2.5 kb/s increments. Th=
e reason Opus can have more than</span></pre>
<pre><span style=3D"color:black">&gt; 1200 possible bitrates is because on =
the Internet, other layers in the</span></pre>
<pre><span style=3D"color:black">&gt; protocol stack provide octet granular=
ity framing/sizing. We don't need</span></pre>
<pre><span style=3D"color:black">&gt; to spend 11 bits signalling the bitra=
te because UDP already encodes the</span></pre>
<pre><span style=3D"color:black">&gt; packet size.</span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt; There's also practical aspects. If yo=
u look at the rates supported by</span></pre>
<pre><span style=3D"color:black">&gt; AMR-WB, you see that *none* of them r=
epresents an integer number of</span></pre>
<pre><span style=3D"color:black">&gt; bytes per frame. For example, 23.85 k=
b/s is 59 bytes plus 5 bits per</span></pre>
<pre><span style=3D"color:black">&gt; frame. It's definitely not the only c=
ause, but the bottom line is that</span></pre>
<pre><span style=3D"color:black">&gt; the payload format for AMR-WB (rfc486=
7) is quite complex. Now compare</span></pre>
<pre><span style=3D"color:black">&gt; this with the Opus RTP payload</span>=
</pre>
<pre><span style=3D"color:black">&gt; (<a href=3D"http://tools.ietf.org/htm=
l/draft-spittka-payload-rtp-opus" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-spittka-payload-rtp-opus</a>). Although</span></pre>
<pre><span style=3D"color:black">&gt; it's not complete, it's not only much=
 simpler, but it also makes it</span></pre>
<pre><span style=3D"color:black">&gt; possible to decode RTP packets withou=
t having even seen the SDP or any</span></pre>
<pre><span style=3D"color:black">&gt; out-of-band signalling.</span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt; People may have other definitions, bu=
t that's what *I* mean by Opus</span></pre>
<pre><span style=3D"color:black">&gt; being an &quot;Internet codec&quot;.<=
/span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt; Cheers,</span></pre>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&nbsp;&nbsp;&nbsp; Jean-Marc</span></p=
re>
<pre><span style=3D"color:black">&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; -----Original Message----- Fr=
om: rtcweb-bounces at ietf.org</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; [<a href=3D"mailto:rtcweb-bou=
nces" target=3D"_blank">mailto:rtcweb-bounces</a> at ietf.org] On Behalf Of=
 ext Stefan Hakansson</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; LK Sent: 30 August, 2012 14:5=
8 To: Jean-Marc Valin Cc:</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; rtcweb at ietf.org Subject: R=
e: [rtcweb] Confirmation of consensus on</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; audio codecs</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; On 08/29/2012 03:10 PM, Jean-=
Marc Valin wrote:</span></pre>
<pre><span style=3D"color:black">&gt;&gt; On 08/29/2012 05:51 AM, Stefan Ha=
kansson LK wrote:</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; ...</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; That is great, bu=
t sort of underlines that there would be no</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; harm in delaying =
the decision until there are experiences</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; made from real wo=
rld use - 'cause it would not be that long</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; till that experie=
nce has been made (Markus also brought up</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; the IPR status as=
 a reason for waiting - I have no idea how</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; long it would tak=
e to know more about that).</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; As Harald is pointing out, rtcweb=
 implementations are going to ship </span></pre>
<pre><span style=3D"color:black">&gt;&gt; pretty soon.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; If that is the case (and =
I think and hope so), why would we need</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; to make it MTI before see=
ing it in action?</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; [...]</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are you expecting *another* singl=
e, standardized, royalty-free codec </span></pre>
<pre><span style=3D"color:black">&gt;&gt; that operates over vast ranges of=
 bitrates and operating conditions, </span></pre>
<pre><span style=3D"color:black">&gt;&gt; from narrowband speech to stereo =
music, all with low delay, coming</span></pre>
<pre><span style=3D"color:black">&gt;&gt; out in the next year? If not, wha=
t are you really waiting for?</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; No, I'm not expecting tha=
t. I would just prefer us to see that</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Opus does indeed deliver =
as promised before making it MTI. If it</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; does, fine. If not, we'd =
have to discuss what to do then.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; To me it is like if you'r=
e going to place a bet, for an upcoming</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; big race, on a horse that=
 has never been in an actual race, but</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; shows great promise. If y=
ou know that the horse is going to</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; participate in a few prac=
tice races soon, would you not prefer to</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; wait and see how it fares=
 in those before placing the bet (given</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; that the odds would not c=
hange)?</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Anyway, that's my view. L=
et's see what the chairs say.</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; As Paul suggested=
, I was referring to the lack of formal,</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; controlled, chara=
cterization tests. That is how other SDOs do</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; it. I don't think=
 that is the only way to do it, but I think</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; we should at leas=
t have either such tests conducted or</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; experience from d=
eployment and use (in a wide range of</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; conditions and de=
vice types) before making it MTI.</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus has had &quot;ITU-style&quot=
; testing on English ( </span></pre>
<pre><span style=3D"color:black">&gt;&gt; <a href=3D"http://www.opus-codec.=
org/comparison/GoogleTest1.pdf" target=3D"_blank">http://www.opus-codec.org=
/comparison/GoogleTest1.pdf</a> ) and Mandarin</span></pre>
<pre><span style=3D"color:black">&gt;&gt; ( <a href=3D"http://www.opus-code=
c.org/comparison/GoogleTest2.pdf" target=3D"_blank">http://www.opus-codec.o=
rg/comparison/GoogleTest2.pdf</a> ). And if you </span></pre>
<pre><span style=3D"color:black">&gt;&gt; don't trust Google on the tests I=
 linked to, it's also been tested</span></pre>
<pre><span style=3D"color:black">&gt;&gt; by Nokia:</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://researc=
h.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_" target=3D"_blank"=
>http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_</a>=
</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt; Quality_Characterization_of_IETF_Opus=
_Codec.pdf</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Come on, those tests are =
very limited compared to a formal</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization test. (E=
xample: </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://www.itu=
.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx" target=3D"_blank">www=
.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx</a>).</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; There is very little info=
 on the material used, the environment,</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; processing and scripts an=
d so on. And there seems to be no tests</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; whatsoever (at least not =
involving humans) with actual channels</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; introducing jitter and lo=
sses.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Note: I am in no way prop=
osing that a formal characterization</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; test is needed, or even t=
he right thing to do. It is a costly and</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; time consuming process, a=
nd alternative approaches could prove to</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; be more efficient. What I=
 am saying is that I think we should not</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; mandate a codec that has =
neither gone through that kind of formal</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization testing =
nor has any field experience from actual</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; use on a reasonable scale=
 (covering different conditions and</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; device types). It just se=
ems wrong, especially given that we</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; will soon have field expe=
rience.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; Now, unlike other SDOs, the testi=
ng did not stop there. ITU-T codecs </span></pre>
<pre><span style=3D"color:black">&gt;&gt; generally end up being testing wi=
th something in the order of tens</span></pre>
<pre><span style=3D"color:black">&gt;&gt; of minutes worth of audio. In *ad=
dition* to that kind of testing,</span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus also had automated testing w=
ith hundreds of years worth of</span></pre>
<pre><span style=3D"color:black">&gt;&gt; audio. If anything, I think other=
 SDOs should learn something here.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; This may very well be tru=
e. I guess this comes down to how much</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; you trust that the qualit=
y assessment models (e.g. PEAQ, POLQA)</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; give the same result as h=
uman test subjects would. But I think</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; this sounds like a really=
 good thing.</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt; Jean-Marc</span></pre>
<pre><span style=3D"color:black">&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; </span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; _____________________________=
__________________ rtcweb mailing</span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; list rtcweb at ietf.org <a hr=
ef=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https=
://www.ietf.org/mailman/listinfo/rtcweb</a></span></pre>
<pre><span style=3D"color:black">&gt; _____________________________________=
__________</span></pre>
<pre><span style=3D"color:black">&gt; rtcweb mailing list</span></pre>
<pre><span style=3D"color:black">&gt; rtcweb at ietf.org</span></pre>
<pre><span style=3D"color:black">&gt; <a href=3D"https://www.ietf.org/mailm=
an/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo=
/rtcweb</a></span></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt; font-family:&quot;T=
ahoma&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From jmvalin@mozilla.com  Fri Aug 31 13:33:12 2012
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Cc: Randell Jesup <randell-ietf@jesup.org>, rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/31/2012 01:23 PM, Roman Shpount wrote:
> One issue I want to bring into consideration is computation
> complexity of Opus. It is normally acceptable for end user devices
> in point-to-point calls (I do not want to start all the mobile
> device CPU vs battery live discussion. It is irrelevant), but in
> cases where you need to process all the calls on some sort of
> central server (such as conferencing, or virtual reality, or
> massive multi-player games) the additional processing cost of
> supporting Opus can be significant. In such cases G.722 (ideally in
> combination with  PLC and CN to save bandwidth) can be a valid
> compromise, since it can provide quality better then G.711 with CPU
> processing requirements order of magnitude less then Opus.

Actually, you're making a really good point about complexity on the
server side. One important feature in Opus is scalable complexity.
Sure you can use it in ways that eat a lot of CPU to fully optimize
quality, but you can also get good enough quality at much lower
complexity. I just ran a quick test on my two year-old laptop. Using
the unoptimized C reference implementation with a single core, I can
encode and decode at 40 kb/s (higher quality than G.722) 150 channels
in real-time. Using architecture-specific optimizations and both
cores, we should be able to handle about 500 channels on my poor
laptop. And that's assuming that we're actually decoding and
re-encoding all streams, which we'd never do. In a mixing case where
we're decoding all channels, but only encoding one, we end up with
1000 channels.

Oh, one feature I forgot to mention about Opus is that (unlike G722)
it's *really* cheap to figure out which channels are active without
having to decode everything. That alone should provide at least a 2x
saving (more for larger conferences). So we're now at 2000 channels on
my poor old laptop. I think a recent multi-core server should have
about 10x that power. Conclusion, the audio codec complexity will just
*not* be a problem.

	Jean-Marc
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Subject: Re: [rtcweb] Opus vs AMR-WB for packet loss
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OK, here's something concrete for comparing Opus to AMR-WB in packet
loss conditions. I've simulated 30% bursty packet loss in the
following conditions:
1) AMR-WB mode 8 (23.85 kb/s)
2) Opus at 24 kb/s (no FEC)
3) Opus at 24 kb/s with FEC (24 kb/s is the total including FEC)

I've uploaded the results at: http://jmvalin.ca/plc/

I think everyone will agree that even without FEC, Opus sounds much
better than AMR-WB. And when you turn FEC on, then you barely even
notice the losses.

	Jean-Marc

On 08/31/2012 11:16 AM, Mandyam, Giridhar wrote:
> Thank you for pointing out these results.
> 
> 
> 
> I think these results represent a very good start, but the results
> as provided may not be sufficient to characterize Opus.  I’ve
> already cited the Dynastat study in 3GPP2 for EVRC-B
> characterization in a separate email.  In addition to packet loss
> (in the 3GPP2 study, the analog would be frame error rate),
> background noise conditions were varied (based on MNRU, street
> noise, car noise).  This is the kind of testing that is typical for
> codecs standardized in 3GPP and 3GPP2.
> 
> 
> 
> There are several details missing.  For instance,
> 
> 
> 
> Was only office noise tested? If so, why?
> 
> How many listeners were used?
> 
> What was the mix of talkers (e.g. no. of female, no. of male)?
> 
> 
> 
> If you can point to more details of the testing, that would be
> very helpful for members of the group.
> 
> 
> 
> -Giri Mandyam
> 
> 
> 
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
> *On Behalf Of *Koen Vos *Sent:* Friday, August 31, 2012 7:38 AM 
> *To:* rtcweb@ietf.org *Subject:* Re: [rtcweb] Opus over the
> Internet?
> 
> 
> 
> Lars Eggert wrote:
> 
>> That's all great, but is there any qualitative comparison data of
>> the different codecs available?
> 
> Please have a look at the test Dynastat did with SILK: 
> http://developer.skype.com/resources/SILKDataSheet.pdf
> 
> Note that SILK in Opus is quite similar to the old SILK tested by 
> Dynastat.  All changes were thoroughly tested to not degrade 
> performance, and in most cases improved it.  So the SILK Dynastat 
> results give a lower bound on the performance of a subset of the
> Opus modes.
> 
> While these plots don't show G.722, it was in fact included in the
> test for clean input signals without packet loss.  According to the
> test results: - SILK at 8.85 kbps is outperformed by G.722 at 64
> kbps with statistical significance - SILK at 12.65 kbps is a
> statistical tie with G.722 at 64 kbps - SILK at 18.85 kbps and
> above outperforms G.722 at 64 kbps with statistical significance.
> 
> The packet loss in that test was maximally random (ie not bursty).
>  While that may not perfectly mimic real networks, I can think of
> no reason why the results would be reversed or very different in
> practice.
> 
> To give one more measure of suitability for the Internet: Skype has
> used SILK to deliver 100s of billions of Skype-to-Skype voice
> minutes over the Internet, and our users are very satisfied with
> it.
> 
> Hope this helps, koen.
> 
> Lars Eggert wrote: That's all great, but is there any qualitative
> comparison data of the different codecs available?
> 
> 
> 
> --
> 
> Sent from a mobile device; please excuse typos.
> 
> 
> 
> On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin at
> mozilla.com> wrote:
> 
> 
> 
>> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:
> 
>>> Are there any tests that actually compare different codecs, say
>>> Opus
> 
>>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some
>>> loss and
> 
>>> jitter? I suppose the performance is only partially related to
>>> the
> 
>>> codec, and partially to other implementation decisions, so it
>>> might
> 
>>> be difficult to compare apples-with-apples. But since people
>>> are
> 
>>> arguing that Opus is an "Internet codec", it would be actually
>>> nice
> 
>>> to see some results in this sense. Or does "Internet codec"
>>> mean
> 
>>> something else?
> 
>> 
> 
>> The term "Internet codec" means different things to different
>> people,
> 
>> but to me this means more than "having good packet loss
>> concealment".
> 
>> Sure we have PLC (though it was never compared to G.722 or
>> AMR-WB) and
> 
>> we also have (optional) low bit-rate embedded redundancy (also
>> called
> 
>> FEC) and (also optional) independent frames, but that's just one
>> aspect.
> 
>> 
> 
>> One of the first things we've been asked when designing Opus was
>> to make
> 
>> the rate *really* adaptable because we never know what kind of
>> rates
> 
>> will be available. This not only meant having a wide range of
>> bitrates,
> 
>> but also being able to vary in small increments. This is why Opus
>> scales
> 
>> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one
>> byte with
> 
>> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to
>> 23.85 in
> 
>> (on average) ~2.5 kb/s increments. The reason Opus can have more
>> than
> 
>> 1200 possible bitrates is because on the Internet, other layers
>> in the
> 
>> protocol stack provide octet granularity framing/sizing. We don't
>> need
> 
>> to spend 11 bits signalling the bitrate because UDP already
>> encodes the
> 
>> packet size.
> 
>> 
> 
>> There's also practical aspects. If you look at the rates
>> supported by
> 
>> AMR-WB, you see that *none* of them represents an integer number
>> of
> 
>> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits
>> per
> 
>> frame. It's definitely not the only cause, but the bottom line is
>> that
> 
>> the payload format for AMR-WB (rfc4867) is quite complex. Now
>> compare
> 
>> this with the Opus RTP payload
> 
>> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus).
>> Although
> 
>> it's not complete, it's not only much simpler, but it also makes
>> it
> 
>> possible to decode RTP packets without having even seen the SDP
>> or any
> 
>> out-of-band signalling.
> 
>> 
> 
>> People may have other definitions, but that's what *I* mean by
>> Opus
> 
>> being an "Internet codec".
> 
>> 
> 
>> Cheers,
> 
>> 
> 
>> Jean-Marc
> 
>> 
> 
>>>> -----Original Message----- From: rtcweb-bounces at ietf.org
> 
>>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan
>>>> Hakansson
> 
>>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:
> 
>>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of
>>>> consensus on
> 
>>>> audio codecs
> 
>>>> 
> 
>>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
> 
>>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
> 
>>>>> ...
> 
>>> 
> 
>>>>>>> That is great, but sort of underlines that there would
>>>>>>> be no
> 
>>>>>>> harm in delaying the decision until there are
>>>>>>> experiences
> 
>>>>>>> made from real world use - 'cause it would not be that
>>>>>>> long
> 
>>>>>>> till that experience has been made (Markus also brought
>>>>>>> up
> 
>>>>>>> the IPR status as a reason for waiting - I have no idea
>>>>>>> how
> 
>>>>>>> long it would take to know more about that).
> 
>>> 
> 
>>> As Harald is pointing out, rtcweb implementations are going to
>>> ship
> 
>>> pretty soon.
> 
>>>>> 
> 
>>>>> If that is the case (and I think and hope so), why would we
>>>>> need
> 
>>>>> to make it MTI before seeing it in action?
> 
>>>>> 
> 
>>>>> [...]
> 
>>>>> 
> 
>>> Are you expecting *another* single, standardized, royalty-free
>>> codec
> 
>>> that operates over vast ranges of bitrates and operating
>>> conditions,
> 
>>> from narrowband speech to stereo music, all with low delay,
>>> coming
> 
>>> out in the next year? If not, what are you really waiting for?
> 
>>>>> 
> 
>>>>> No, I'm not expecting that. I would just prefer us to see
>>>>> that
> 
>>>>> Opus does indeed deliver as promised before making it MTI.
>>>>> If it
> 
>>>>> does, fine. If not, we'd have to discuss what to do then.
> 
>>>>> 
> 
>>>>> To me it is like if you're going to place a bet, for an
>>>>> upcoming
> 
>>>>> big race, on a horse that has never been in an actual race,
>>>>> but
> 
>>>>> shows great promise. If you know that the horse is going
>>>>> to
> 
>>>>> participate in a few practice races soon, would you not
>>>>> prefer to
> 
>>>>> wait and see how it fares in those before placing the bet
>>>>> (given
> 
>>>>> that the odds would not change)?
> 
>>>>> 
> 
>>>>> Anyway, that's my view. Let's see what the chairs say.
> 
>>> 
> 
>>>>>>> As Paul suggested, I was referring to the lack of
>>>>>>> formal,
> 
>>>>>>> controlled, characterization tests. That is how other
>>>>>>> SDOs do
> 
>>>>>>> it. I don't think that is the only way to do it, but I
>>>>>>> think
> 
>>>>>>> we should at least have either such tests conducted or
> 
>>>>>>> experience from deployment and use (in a wide range of
> 
>>>>>>> conditions and device types) before making it MTI.
> 
>>> 
> 
>>> Opus has had "ITU-style" testing on English (
> 
>>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and
>>> Mandarin
> 
>>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And
>>> if you
> 
>>> don't trust Google on the tests I linked to, it's also been
>>> tested
> 
>>> by Nokia:
> 
>>> 
> 
>>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>
>>>>> 
>>> 
> 
>>>>> 
> 
>> Quality_Characterization_of_IETF_Opus_Codec.pdf
> 
>>>>> Come on, those tests are very limited compared to a formal
> 
>>>>> characterization test. (Example:
> 
>>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx
>>>>> <http://www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx>).
>
>>>>>  There is very little info on the material used, the
>>>>> environment,
> 
>>>>> processing and scripts and so on. And there seems to be no
>>>>> tests
> 
>>>>> whatsoever (at least not involving humans) with actual
>>>>> channels
> 
>>>>> introducing jitter and losses.
> 
>>>>> 
> 
>>>>> Note: I am in no way proposing that a formal
>>>>> characterization
> 
>>>>> test is needed, or even the right thing to do. It is a
>>>>> costly and
> 
>>>>> time consuming process, and alternative approaches could
>>>>> prove to
> 
>>>>> be more efficient. What I am saying is that I think we
>>>>> should not
> 
>>>>> mandate a codec that has neither gone through that kind of
>>>>> formal
> 
>>>>> characterization testing nor has any field experience from
>>>>> actual
> 
>>>>> use on a reasonable scale (covering different conditions
>>>>> and
> 
>>>>> device types). It just seems wrong, especially given that
>>>>> we
> 
>>>>> will soon have field experience.
> 
>>>>> 
> 
>>> 
> 
>>> Now, unlike other SDOs, the testing did not stop there. ITU-T
>>> codecs
> 
>>> generally end up being testing with something in the order of
>>> tens
> 
>>> of minutes worth of audio. In *addition* to that kind of
>>> testing,
> 
>>> Opus also had automated testing with hundreds of years worth
>>> of
> 
>>> audio. If anything, I think other SDOs should learn something
>>> here.
> 
>>>>> 
> 
>>>>> This may very well be true. I guess this comes down to how
>>>>> much
> 
>>>>> you trust that the quality assessment models (e.g. PEAQ,
>>>>> POLQA)
> 
>>>>> give the same result as human test subjects would. But I
>>>>> think
> 
>>>>> this sounds like a really good thing.
> 
>>>>> 
> 
>>> 
> 
>>> Jean-Marc
> 
>>> 
> 
>>>>> 
> 
>>>> 
> 
>>>> _______________________________________________ rtcweb
>>>> mailing
> 
>>>> list rtcweb at ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
> 
>> _______________________________________________
> 
>> rtcweb mailing list
> 
>> rtcweb at 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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Thanks for getting back to me.  It looks like the assembled speaker panels =
and the mix of talkers followed a similar methodology to PP and PP2.

3GPP TR 26.935 (Packet Switched (PS) conversational multimedia applications=
; Performance characterisation of default codecs) lists in Table 12 three t=
ypes of background noise conditions for AMR-WB:  car, street and cafeteria.=
  Cafeteria is the closest to office IMO -  it includes background babble i=
n an open environment.

-Giri


From: Koen Vos [mailto:koen.vos@skype.net]
Sent: Friday, August 31, 2012 1:30 PM
To: Mandyam, Giridhar; rtcweb@ietf.org
Subject: RE: [rtcweb] Opus over the Internet?

Hi Giri,

Dynastat's subjective test involved the following design parameters:

- subjects: 32, four panels of eight subjects, native speakers of North Ame=
rican English
- randomizations: one per listening panel
- talkers: eight (four males, four females), native speakers of North Ameri=
can English
- speech samples: four 8 sec. sentence-pairs per talker
- experimental design: partially-balanced, randomized-blocks
Dynastat constructed the randomizations (i.e., pseudo-randomized presentati=
on sequences) for each listening panel. Each randomization included 256 tri=
als (i.e., 8 talkers x 32 test conditions).

We tested only office noise at 15 dB SNR because we didn't have more condit=
ions available.

As web browsers are commonly used in an office, knowledge of a codec's perf=
ormance in that kind of environment would seem helpful to this working grou=
p.  Do I understand correctly that the 3GPP and 3GPP2 tests did not include=
 any office noise conditions?

best,
koen.
________________________________
From: Mandyam, Giridhar [mandyam@quicinc.com]
Sent: Friday, August 31, 2012 8:16 AM
To: Koen Vos; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: RE: [rtcweb] Opus over the Internet?
Thank you for pointing out these results.

I think these results represent a very good start, but the results as provi=
ded may not be sufficient to characterize Opus.  I've already cited the Dyn=
astat study in 3GPP2 for EVRC-B characterization in a separate email.  In a=
ddition to packet loss (in the 3GPP2 study, the analog would be frame error=
 rate), background noise conditions were varied (based on MNRU, street nois=
e, car noise).  This is the kind of testing that is typical for codecs stan=
dardized in 3GPP and 3GPP2.

There are several details missing.  For instance,

Was only office noise tested? If so, why?
How many listeners were used?
What was the mix of talkers (e.g. no. of female, no. of male)?

If you can point to more details of the testing, that would be very helpful=
 for members of the group.

-Giri Mandyam

From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org]<mailto:[mailto:rtcweb-bounces@ietf.org]> On Behalf Of K=
oen Vos
Sent: Friday, August 31, 2012 7:38 AM
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Opus over the Internet?

Lars Eggert wrote:
> That's all great, but is there any qualitative comparison data of the dif=
ferent codecs available?

Please have a look at the test Dynastat did with SILK: http://developer.sky=
pe.com/resources/SILKDataSheet.pdf

Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
  All changes were thoroughly tested to not degrade performance, and in mos=
t cases improved it.  So the SILK Dynastat results give a lower bound on th=
e performance of a subset of the Opus modes.

While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.  According to the test results:
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.

The packet loss in that test was maximally random (ie not bursty).  While t=
hat may not perfectly mimic real networks, I can think of no reason why the=
 results would be reversed or very different in practice.

To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.

Hope this helps,
koen.

Lars Eggert wrote:
That's all great, but is there any qualitative comparison data of the diffe=
rent codecs available?



--

Sent from a mobile device; please excuse typos.



On Aug 30, 2012, at 21:07, "Jean-Marc Valin" <jmvalin at mozilla.com> wrote=
:



> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:

>> Are there any tests that actually compare different codecs, say Opus

>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some loss and

>> jitter? I suppose the performance is only partially related to the

>> codec, and partially to other implementation decisions, so it might

>> be difficult to compare apples-with-apples. But since people are

>> arguing that Opus is an "Internet codec", it would be actually nice

>> to see some results in this sense. Or does "Internet codec" mean

>> something else?

>

> The term "Internet codec" means different things to different people,

> but to me this means more than "having good packet loss concealment".

> Sure we have PLC (though it was never compared to G.722 or AMR-WB) and

> we also have (optional) low bit-rate embedded redundancy (also called

> FEC) and (also optional) independent frames, but that's just one aspect.

>

> One of the first things we've been asked when designing Opus was to make

> the rate *really* adaptable because we never know what kind of rates

> will be available. This not only meant having a wide range of bitrates,

> but also being able to vary in small increments. This is why Opus scales

> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one byte with

> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to 23.85 in

> (on average) ~2.5 kb/s increments. The reason Opus can have more than

> 1200 possible bitrates is because on the Internet, other layers in the

> protocol stack provide octet granularity framing/sizing. We don't need

> to spend 11 bits signalling the bitrate because UDP already encodes the

> packet size.

>

> There's also practical aspects. If you look at the rates supported by

> AMR-WB, you see that *none* of them represents an integer number of

> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits per

> frame. It's definitely not the only cause, but the bottom line is that

> the payload format for AMR-WB (rfc4867) is quite complex. Now compare

> this with the Opus RTP payload

> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus). Although

> it's not complete, it's not only much simpler, but it also makes it

> possible to decode RTP packets without having even seen the SDP or any

> out-of-band signalling.

>

> People may have other definitions, but that's what *I* mean by Opus

> being an "Internet codec".

>

> Cheers,

>

>    Jean-Marc

>

>>> -----Original Message----- From: rtcweb-bounces at ietf.org

>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan Hakansson

>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:

>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of consensus on

>>> audio codecs

>>>

>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:

>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:

>>>> ...

>>

>>>>>> That is great, but sort of underlines that there would be no

>>>>>> harm in delaying the decision until there are experiences

>>>>>> made from real world use - 'cause it would not be that long

>>>>>> till that experience has been made (Markus also brought up

>>>>>> the IPR status as a reason for waiting - I have no idea how

>>>>>> long it would take to know more about that).

>>

>> As Harald is pointing out, rtcweb implementations are going to ship

>> pretty soon.

>>>>

>>>> If that is the case (and I think and hope so), why would we need

>>>> to make it MTI before seeing it in action?

>>>>

>>>> [...]

>>>>

>> Are you expecting *another* single, standardized, royalty-free codec

>> that operates over vast ranges of bitrates and operating conditions,

>> from narrowband speech to stereo music, all with low delay, coming

>> out in the next year? If not, what are you really waiting for?

>>>>

>>>> No, I'm not expecting that. I would just prefer us to see that

>>>> Opus does indeed deliver as promised before making it MTI. If it

>>>> does, fine. If not, we'd have to discuss what to do then.

>>>>

>>>> To me it is like if you're going to place a bet, for an upcoming

>>>> big race, on a horse that has never been in an actual race, but

>>>> shows great promise. If you know that the horse is going to

>>>> participate in a few practice races soon, would you not prefer to

>>>> wait and see how it fares in those before placing the bet (given

>>>> that the odds would not change)?

>>>>

>>>> Anyway, that's my view. Let's see what the chairs say.

>>

>>>>>> As Paul suggested, I was referring to the lack of formal,

>>>>>> controlled, characterization tests. That is how other SDOs do

>>>>>> it. I don't think that is the only way to do it, but I think

>>>>>> we should at least have either such tests conducted or

>>>>>> experience from deployment and use (in a wide range of

>>>>>> conditions and device types) before making it MTI.

>>

>> Opus has had "ITU-style" testing on English (

>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and Mandarin

>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And if you

>> don't trust Google on the tests I linked to, it's also been tested

>> by Nokia:

>>

>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_

>>

>>>>

> Quality_Characterization_of_IETF_Opus_Codec.pdf

>>>> Come on, those tests are very limited compared to a formal

>>>> characterization test. (Example:

>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx<http://www=
.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx>).

>>>> There is very little info on the material used, the environment,

>>>> processing and scripts and so on. And there seems to be no tests

>>>> whatsoever (at least not involving humans) with actual channels

>>>> introducing jitter and losses.

>>>>

>>>> Note: I am in no way proposing that a formal characterization

>>>> test is needed, or even the right thing to do. It is a costly and

>>>> time consuming process, and alternative approaches could prove to

>>>> be more efficient. What I am saying is that I think we should not

>>>> mandate a codec that has neither gone through that kind of formal

>>>> characterization testing nor has any field experience from actual

>>>> use on a reasonable scale (covering different conditions and

>>>> device types). It just seems wrong, especially given that we

>>>> will soon have field experience.

>>>>

>>

>> Now, unlike other SDOs, the testing did not stop there. ITU-T codecs

>> generally end up being testing with something in the order of tens

>> of minutes worth of audio. In *addition* to that kind of testing,

>> Opus also had automated testing with hundreds of years worth of

>> audio. If anything, I think other SDOs should learn something here.

>>>>

>>>> This may very well be true. I guess this comes down to how much

>>>> you trust that the quality assessment models (e.g. PEAQ, POLQA)

>>>> give the same result as human test subjects would. But I think

>>>> this sounds like a really good thing.

>>>>

>>

>> Jean-Marc

>>

>>>>

>>>

>>> _______________________________________________ rtcweb mailing

>>> list rtcweb at ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

> _______________________________________________

> rtcweb mailing list

> rtcweb at ietf.org

> https://www.ietf.org/mailman/listinfo/rtcweb


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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for getting back t=
o me.&nbsp; It looks like the assembled speaker panels and the mix of talke=
rs followed a similar methodology to PP and PP2.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">3GPP TR 26.935 (Packet Sw=
itched (PS) conversational multimedia applications; Performance characteris=
ation of default codecs) lists in Table 12 three types of
 background noise conditions for AMR-WB:&nbsp; car, street and cafeteria.&n=
bsp; Cafeteria is the closest to office IMO - &nbsp;it includes background =
babble in an open environment.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">-Giri<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>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Koen Vos=
 [mailto:koen.vos@skype.net]
<br>
<b>Sent:</b> Friday, August 31, 2012 1:30 PM<br>
<b>To:</b> Mandyam, Giridhar; rtcweb@ietf.org<br>
<b>Subject:</b> RE: [rtcweb] Opus over the Internet?<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:blac=
k">Hi Giri,<o:p></o:p></span></p>
<p><span style=3D"color:black">Dynastat's subjective test involved the foll=
owing design parameters:<o:p></o:p></span></p>
<p><span style=3D"color:black">- subjects: 32, four panels of eight subject=
s, native speakers of North American English<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:blac=
k">- randomizations: one per listening panel<br>
- talkers: eight (four males, four females), native speakers of North Ameri=
can English<br>
- speech samples: four 8 sec. sentence-pairs per talker<br>
- experimental design: partially-balanced, randomized-blocks<br>
Dynastat constructed the randomizations (i.e., pseudo-randomized presentati=
on sequences) for each listening panel. Each randomization included 256 tri=
als (i.e., 8 talkers x 32 test conditions).
<br>
<br>
We tested only office noise at 15 dB SNR because we didn't have more condit=
ions available.&nbsp;
<br>
<br>
As web browsers are commonly used in an office, knowledge of a codec's perf=
ormance in that kind of environment would seem helpful to this working grou=
p.&nbsp; Do I understand correctly that the 3GPP and 3GPP2 tests did not in=
clude any office noise conditions?&nbsp;
<br>
<br>
best,<br>
koen.<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"color:black">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<div id=3D"divRpF668266">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><span style=3D"fon=
t-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:b=
lack">From:</span></b><span style=3D"font-size:10.0pt;font-family:&quot;Tah=
oma&quot;,&quot;sans-serif&quot;;color:black"> Mandyam, Giridhar [mandyam@q=
uicinc.com]<br>
<b>Sent:</b> Friday, August 31, 2012 8:16 AM<br>
<b>To:</b> Koen Vos; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>=
<br>
<b>Subject:</b> RE: [rtcweb] Opus over the Internet?</span><span style=3D"c=
olor:black"><o:p></o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thank you for pointing ou=
t these results.</span><span style=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><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">I think these results rep=
resent a very good start, but the results as provided may not be sufficient=
 to characterize Opus.&nbsp; I&#8217;ve already cited the Dynastat
 study in 3GPP2 for EVRC-B characterization in a separate email.&nbsp; In a=
ddition to packet loss (in the 3GPP2 study, the analog would be frame error=
 rate), background noise conditions were varied (based on MNRU, street nois=
e, car noise).&nbsp; This is the kind of testing
 that is typical for codecs standardized in 3GPP and 3GPP2.</span><span sty=
le=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><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">There are several details=
 missing.&nbsp; For instance,</span><span style=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><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">Was only office noise tes=
ted? If so, why? &nbsp;&nbsp;</span><span style=3D"color:black"><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">How many listeners were u=
sed?&nbsp;
</span><span style=3D"color:black"><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">What was the mix of talke=
rs (e.g. no. of female, no. of male)?
</span><span style=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><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">If you can point to more =
details of the testing, that would be very helpful for members of the group=
.</span><span style=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><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">-Giri Mandyam</span><span=
 style=3D"color:black"><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">&nbsp;</span><span style=
=3D"color:black"><o:p></o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:black">From:</span></b><span sty=
le=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot=
;;color:black">
<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> <a h=
ref=3D"mailto:[mailto:rtcweb-bounces@ietf.org]">
[mailto:rtcweb-bounces@ietf.org]</a> <b>On Behalf Of </b>Koen Vos<br>
<b>Sent:</b> Friday, August 31, 2012 7:38 AM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] Opus over the Internet?</span><span style=3D"c=
olor:black"><o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:black">&nbsp;<o:p></o:p></span>=
</p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">Lars Eggert wrote:</span><span style=3D"color:=
black"><o:p></o:p></span></p>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Courier New&quot;;color:black">&gt; That's all=
 great, but is there any qualitative comparison data of the different codec=
s available?
<br>
<br>
Please have a look at the test Dynastat did with SILK: <a href=3D"http://de=
veloper.skype.com/resources/SILKDataSheet.pdf" target=3D"_blank">
http://developer.skype.com/resources/SILKDataSheet.pdf</a><br>
<br>
Note that SILK in Opus is quite similar to the old SILK tested by Dynastat.=
&nbsp; All changes were thoroughly tested to not degrade performance, and i=
n most cases improved it.&nbsp; So the SILK Dynastat results give a lower b=
ound on the performance of a subset of the
 Opus modes.<br>
<br>
While these plots don't show G.722, it was in fact included in the test for=
 clean input signals without packet loss.&nbsp; According to the test resul=
ts:
<br>
- SILK at 8.85 kbps is outperformed by G.722 at 64 kbps with statistical si=
gnificance<br>
- SILK at 12.65 kbps is a statistical tie with G.722 at 64 kbps <br>
- SILK at 18.85 kbps and above outperforms G.722 at 64 kbps with statistica=
l significance.<br>
<br>
The packet loss in that test was maximally random (ie not bursty).&nbsp; Wh=
ile that may not perfectly mimic real networks, I can think of no reason wh=
y the results would be reversed or very different in practice.
<br>
<br>
To give one more measure of suitability for the Internet: Skype has used SI=
LK to deliver 100s of billions of Skype-to-Skype voice minutes over the Int=
ernet, and our users are very satisfied with it.
<br>
<br>
Hope this helps, <br>
koen.</span><span style=3D"color:black"><o:p></o:p></span></p>
<pre><span style=3D"color:black">Lars Eggert wrote:<br>That's all great, bu=
t is there any qualitative comparison data of the different codecs availabl=
e?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&nbsp;<o:p></o:p></span></pre>
<pre><span style=3D"color:black">-- <o:p></o:p></span></pre>
<pre><span style=3D"color:black">Sent from a mobile device; please excuse t=
ypos.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&nbsp;<o:p></o:p></span></pre>
<pre><span style=3D"color:black">On Aug 30, 2012, at 21:07, &quot;Jean-Marc=
 Valin&quot; &lt;jmvalin at mozilla.com&gt; wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&nbsp;<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; On 12-08-30 09:14 AM, Markus.Isomaki =
at nokia.com wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are there any tests that actually=
 compare different codecs, say Opus<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; vs. G.722 vs. AMR-WB, in typical =
Internet use, meaning some loss and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; jitter? I suppose the performance=
 is only partially related to the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; codec, and partially to other imp=
lementation decisions, so it might<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; be difficult to compare apples-wi=
th-apples. But since people are<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; arguing that Opus is an &quot;Int=
ernet codec&quot;, it would be actually nice<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; to see some results in this sense=
. Or does &quot;Internet codec&quot; mean<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; something else?<o:p></o:p></span>=
</pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; The term &quot;Internet codec&quot; m=
eans different things to different people,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; but to me this means more than &quot;=
having good packet loss concealment&quot;.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Sure we have PLC (though it was never=
 compared to G.722 or AMR-WB) and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; we also have (optional) low bit-rate =
embedded redundancy (also called<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; FEC) and (also optional) independent =
frames, but that's just one aspect.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; One of the first things we've been as=
ked when designing Opus was to make<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; the rate *really* adaptable because w=
e never know what kind of rates<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; will be available. This not only mean=
t having a wide range of bitrates,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; but also being able to vary in small =
increments. This is why Opus scales<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; from about 6 kb/s to 512 kb/s, in inc=
rements of 0.4 kb/s (one byte with<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; 20 ms frames). Compare that to AMR-WB=
, which scales from 6.6 to 23.85 in<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; (on average) ~2.5 kb/s increments. Th=
e reason Opus can have more than<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; 1200 possible bitrates is because on =
the Internet, other layers in the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; protocol stack provide octet granular=
ity framing/sizing. We don't need<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; to spend 11 bits signalling the bitra=
te because UDP already encodes the<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; packet size.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; There's also practical aspects. If yo=
u look at the rates supported by<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; AMR-WB, you see that *none* of them r=
epresents an integer number of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; bytes per frame. For example, 23.85 k=
b/s is 59 bytes plus 5 bits per<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; frame. It's definitely not the only c=
ause, but the bottom line is that<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; the payload format for AMR-WB (rfc486=
7) is quite complex. Now compare<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; this with the Opus RTP payload<o:p></=
o:p></span></pre>
<pre><span style=3D"color:black">&gt; (<a href=3D"http://tools.ietf.org/htm=
l/draft-spittka-payload-rtp-opus" target=3D"_blank">http://tools.ietf.org/h=
tml/draft-spittka-payload-rtp-opus</a>). Although<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; it's not complete, it's not only much=
 simpler, but it also makes it<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; possible to decode RTP packets withou=
t having even seen the SDP or any<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; out-of-band signalling.<o:p></o:p></s=
pan></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; People may have other definitions, bu=
t that's what *I* mean by Opus<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; being an &quot;Internet codec&quot;.<=
o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Cheers,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&nbsp;&nbsp;&nbsp; Jean-Marc<o:p></o:p=
></span></pre>
<pre><span style=3D"color:black">&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; -----Original Message----- Fr=
om: rtcweb-bounces at ietf.org<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; [<a href=3D"mailto:rtcweb-bou=
nces" target=3D"_blank">mailto:rtcweb-bounces</a> at ietf.org] On Behalf Of=
 ext Stefan Hakansson<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; LK Sent: 30 August, 2012 14:5=
8 To: Jean-Marc Valin Cc:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; rtcweb at ietf.org Subject: R=
e: [rtcweb] Confirmation of consensus on<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; audio codecs<o:p></o:p></span=
></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; On 08/29/2012 03:10 PM, Jean-=
Marc Valin wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; On 08/29/2012 05:51 AM, Stefan Ha=
kansson LK wrote:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; ...<o:p></o:p></span></pr=
e>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; That is great, bu=
t sort of underlines that there would be no<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; harm in delaying =
the decision until there are experiences<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; made from real wo=
rld use - 'cause it would not be that long<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; till that experie=
nce has been made (Markus also brought up<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; the IPR status as=
 a reason for waiting - I have no idea how<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; long it would tak=
e to know more about that).<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; As Harald is pointing out, rtcweb=
 implementations are going to ship <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; pretty soon.<o:p></o:p></span></p=
re>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; If that is the case (and =
I think and hope so), why would we need<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; to make it MTI before see=
ing it in action?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; [...]<o:p></o:p></span></=
pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Are you expecting *another* singl=
e, standardized, royalty-free codec <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; that operates over vast ranges of=
 bitrates and operating conditions, <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; from narrowband speech to stereo =
music, all with low delay, coming<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; out in the next year? If not, wha=
t are you really waiting for?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; No, I'm not expecting tha=
t. I would just prefer us to see that<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Opus does indeed deliver =
as promised before making it MTI. If it<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; does, fine. If not, we'd =
have to discuss what to do then.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; To me it is like if you'r=
e going to place a bet, for an upcoming<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; big race, on a horse that=
 has never been in an actual race, but<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; shows great promise. If y=
ou know that the horse is going to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; participate in a few prac=
tice races soon, would you not prefer to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; wait and see how it fares=
 in those before placing the bet (given<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; that the odds would not c=
hange)?<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Anyway, that's my view. L=
et's see what the chairs say.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; As Paul suggested=
, I was referring to the lack of formal,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; controlled, chara=
cterization tests. That is how other SDOs do<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; it. I don't think=
 that is the only way to do it, but I think<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; we should at leas=
t have either such tests conducted or<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; experience from d=
eployment and use (in a wide range of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt;&gt;&gt; conditions and de=
vice types) before making it MTI.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus has had &quot;ITU-style&quot=
; testing on English ( <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <a href=3D"http://www.opus-codec.=
org/comparison/GoogleTest1.pdf" target=3D"_blank">http://www.opus-codec.org=
/comparison/GoogleTest1.pdf</a> ) and Mandarin<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; ( <a href=3D"http://www.opus-code=
c.org/comparison/GoogleTest2.pdf" target=3D"_blank">http://www.opus-codec.o=
rg/comparison/GoogleTest2.pdf</a> ). And if you <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; don't trust Google on the tests I=
 linked to, it's also been tested<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; by Nokia:<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://researc=
h.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_" target=3D"_blank"=
>http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_</a>=
<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; Quality_Characterization_of_IETF_Opus=
_Codec.pdf<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Come on, those tests are =
very limited compared to a formal<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization test. (E=
xample: <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <a href=3D"http://www.itu=
.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx" target=3D"_blank">www=
.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx</a>).<o:p></o:p></=
span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; There is very little info=
 on the material used, the environment,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; processing and scripts an=
d so on. And there seems to be no tests<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; whatsoever (at least not =
involving humans) with actual channels<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; introducing jitter and lo=
sses.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; Note: I am in no way prop=
osing that a formal characterization<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; test is needed, or even t=
he right thing to do. It is a costly and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; time consuming process, a=
nd alternative approaches could prove to<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; be more efficient. What I=
 am saying is that I think we should not<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; mandate a codec that has =
neither gone through that kind of formal<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; characterization testing =
nor has any field experience from actual<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; use on a reasonable scale=
 (covering different conditions and<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; device types). It just se=
ems wrong, especially given that we<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; will soon have field expe=
rience.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Now, unlike other SDOs, the testi=
ng did not stop there. ITU-T codecs <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; generally end up being testing wi=
th something in the order of tens<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; of minutes worth of audio. In *ad=
dition* to that kind of testing,<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Opus also had automated testing w=
ith hundreds of years worth of<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; audio. If anything, I think other=
 SDOs should learn something here.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; This may very well be tru=
e. I guess this comes down to how much<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; you trust that the qualit=
y assessment models (e.g. PEAQ, POLQA)<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; give the same result as h=
uman test subjects would. But I think<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; this sounds like a really=
 good thing.<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; Jean-Marc<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; <o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; _____________________________=
__________________ rtcweb mailing<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt;&gt;&gt; list rtcweb at ietf.org <a hr=
ef=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https=
://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; _____________________________________=
__________<o:p></o:p></span></pre>
<pre><span style=3D"color:black">&gt; rtcweb mailing list<o:p></o:p></span>=
</pre>
<pre><span style=3D"color:black">&gt; rtcweb at ietf.org<o:p></o:p></span><=
/pre>
<pre><span style=3D"color:black">&gt; <a href=3D"https://www.ietf.org/mailm=
an/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo=
/rtcweb</a><o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;;color:black">&nbsp;</span><span style=3D"=
color:black"><o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
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On 08/31/2012 01:25 PM, Mandyam, Giridhar wrote:
> I'm not sure I understand your logic either.  You could make the
> same argument in favor of having AMR-NB or EVRC-B as MTI.  Why not
> do so and cover 100% of the use cases?

Problem is that adding AMR-NB or EVRC-B would only prevent failure
below 64 kb/s, but still leave us with embarrassingly bad quality. On
the other hand, Opus will work (i.e. not fail) for all use cases, and
actually be optimal or near-optimal for the vast majority of these use
cases. Want slightly lower quality below 10 kb/s, go ahead and
implement AMR/EVRC, but at least you're guarantee the call will not
fail if it's not implemented.

	Jean-Marc
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On 8/31/2012 4:50 PM, Jean-Marc Valin wrote:
> OK, here's something concrete for comparing Opus to AMR-WB in packet
> loss conditions. I've simulated 30% bursty packet loss in the
> following conditions:
> 1) AMR-WB mode 8 (23.85 kb/s)
> 2) Opus at 24 kb/s (no FEC)
> 3) Opus at 24 kb/s with FEC (24 kb/s is the total including FEC)
>
> I've uploaded the results at: http://jmvalin.ca/plc/
>
> I think everyone will agree that even without FEC, Opus sounds much
> better than AMR-WB. And when you turn FEC on, then you barely even
> notice the losses.

That's really, really good.  :-)  Especially with FEC.

-- 
Randell Jesup
randell-ietf@jesup.org

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At 11:12 AM -0400 8/31/12, John Leslie wrote:

>  Our issue here is Mandatory-to-Implement. It is very important to
>  have at least one MTI audio codec. I could live with that being G.711,
>  because I trust the market to _actually_ implement others.

Exactly.  The discussion has been going in my view off-track into 
debates about which codec is best for which environments.  The real 
issue is mandatory versus recommended.

We can pick G.711 as MTI and rely on implementers to support others.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
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has been sitting in them before one.
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Subject: Re: [rtcweb] Opus vs AMR-WB for packet loss
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We have made similar experiences when testing PLC behavior for Mumble. 
Opus handles package loss very gracefully even without FEC (which is 
what we are using). There are audible artifacts for sure, however they 
don't stand out like with many other codecs.

Regards,
Stefan

On 31/08/2012 22:50, Jean-Marc Valin wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> OK, here's something concrete for comparing Opus to AMR-WB in packet
> loss conditions. I've simulated 30% bursty packet loss in the
> following conditions:
> 1) AMR-WB mode 8 (23.85 kb/s)
> 2) Opus at 24 kb/s (no FEC)
> 3) Opus at 24 kb/s with FEC (24 kb/s is the total including FEC)
>
> I've uploaded the results at: http://jmvalin.ca/plc/
>
> I think everyone will agree that even without FEC, Opus sounds much
> better than AMR-WB. And when you turn FEC on, then you barely even
> notice the losses.
>
> 	Jean-Marc
>
> On 08/31/2012 11:16 AM, Mandyam, Giridhar wrote:
>> Thank you for pointing out these results.
>>
>>
>>
>> I think these results represent a very good start, but the results
>> as provided may not be sufficient to characterize Opus.  I’ve
>> already cited the Dynastat study in 3GPP2 for EVRC-B
>> characterization in a separate email.  In addition to packet loss
>> (in the 3GPP2 study, the analog would be frame error rate),
>> background noise conditions were varied (based on MNRU, street
>> noise, car noise).  This is the kind of testing that is typical for
>> codecs standardized in 3GPP and 3GPP2.
>>
>>
>>
>> There are several details missing.  For instance,
>>
>>
>>
>> Was only office noise tested? If so, why?
>>
>> How many listeners were used?
>>
>> What was the mix of talkers (e.g. no. of female, no. of male)?
>>
>>
>>
>> If you can point to more details of the testing, that would be
>> very helpful for members of the group.
>>
>>
>>
>> -Giri Mandyam
>>
>>
>>
>> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
>> *On Behalf Of *Koen Vos *Sent:* Friday, August 31, 2012 7:38 AM
>> *To:* rtcweb@ietf.org *Subject:* Re: [rtcweb] Opus over the
>> Internet?
>>
>>
>>
>> Lars Eggert wrote:
>>
>>> That's all great, but is there any qualitative comparison data of
>>> the different codecs available?
>> Please have a look at the test Dynastat did with SILK:
>> http://developer.skype.com/resources/SILKDataSheet.pdf
>>
>> Note that SILK in Opus is quite similar to the old SILK tested by
>> Dynastat.  All changes were thoroughly tested to not degrade
>> performance, and in most cases improved it.  So the SILK Dynastat
>> results give a lower bound on the performance of a subset of the
>> Opus modes.
>>
>> While these plots don't show G.722, it was in fact included in the
>> test for clean input signals without packet loss.  According to the
>> test results: - SILK at 8.85 kbps is outperformed by G.722 at 64
>> kbps with statistical significance - SILK at 12.65 kbps is a
>> statistical tie with G.722 at 64 kbps - SILK at 18.85 kbps and
>> above outperforms G.722 at 64 kbps with statistical significance.
>>
>> The packet loss in that test was maximally random (ie not bursty).
>>   While that may not perfectly mimic real networks, I can think of
>> no reason why the results would be reversed or very different in
>> practice.
>>
>> To give one more measure of suitability for the Internet: Skype has
>> used SILK to deliver 100s of billions of Skype-to-Skype voice
>> minutes over the Internet, and our users are very satisfied with
>> it.
>>
>> Hope this helps, koen.
>>
>> Lars Eggert wrote: That's all great, but is there any qualitative
>> comparison data of the different codecs available?
>>
>>
>>
>> --
>>
>> Sent from a mobile device; please excuse typos.
>>
>>
>>
>> On Aug 30, 2012, at 21:07, "Jean-Marc Valin"<jmvalin at
>> mozilla.com>  wrote:
>>
>>
>>
>>> On 12-08-30 09:14 AM, Markus.Isomaki at nokia.com wrote:
>>>> Are there any tests that actually compare different codecs, say
>>>> Opus
>>>> vs. G.722 vs. AMR-WB, in typical Internet use, meaning some
>>>> loss and
>>>> jitter? I suppose the performance is only partially related to
>>>> the
>>>> codec, and partially to other implementation decisions, so it
>>>> might
>>>> be difficult to compare apples-with-apples. But since people
>>>> are
>>>> arguing that Opus is an "Internet codec", it would be actually
>>>> nice
>>>> to see some results in this sense. Or does "Internet codec"
>>>> mean
>>>> something else?
>>> The term "Internet codec" means different things to different
>>> people,
>>> but to me this means more than "having good packet loss
>>> concealment".
>>> Sure we have PLC (though it was never compared to G.722 or
>>> AMR-WB) and
>>> we also have (optional) low bit-rate embedded redundancy (also
>>> called
>>> FEC) and (also optional) independent frames, but that's just one
>>> aspect.
>>> One of the first things we've been asked when designing Opus was
>>> to make
>>> the rate *really* adaptable because we never know what kind of
>>> rates
>>> will be available. This not only meant having a wide range of
>>> bitrates,
>>> but also being able to vary in small increments. This is why Opus
>>> scales
>>> from about 6 kb/s to 512 kb/s, in increments of 0.4 kb/s (one
>>> byte with
>>> 20 ms frames). Compare that to AMR-WB, which scales from 6.6 to
>>> 23.85 in
>>> (on average) ~2.5 kb/s increments. The reason Opus can have more
>>> than
>>> 1200 possible bitrates is because on the Internet, other layers
>>> in the
>>> protocol stack provide octet granularity framing/sizing. We don't
>>> need
>>> to spend 11 bits signalling the bitrate because UDP already
>>> encodes the
>>> packet size.
>>> There's also practical aspects. If you look at the rates
>>> supported by
>>> AMR-WB, you see that *none* of them represents an integer number
>>> of
>>> bytes per frame. For example, 23.85 kb/s is 59 bytes plus 5 bits
>>> per
>>> frame. It's definitely not the only cause, but the bottom line is
>>> that
>>> the payload format for AMR-WB (rfc4867) is quite complex. Now
>>> compare
>>> this with the Opus RTP payload
>>> (http://tools.ietf.org/html/draft-spittka-payload-rtp-opus).
>>> Although
>>> it's not complete, it's not only much simpler, but it also makes
>>> it
>>> possible to decode RTP packets without having even seen the SDP
>>> or any
>>> out-of-band signalling.
>>> People may have other definitions, but that's what *I* mean by
>>> Opus
>>> being an "Internet codec".
>>> Cheers,
>>> Jean-Marc
>>>>> -----Original Message----- From: rtcweb-bounces at ietf.org
>>>>> [mailto:rtcweb-bounces at ietf.org] On Behalf Of ext Stefan
>>>>> Hakansson
>>>>> LK Sent: 30 August, 2012 14:58 To: Jean-Marc Valin Cc:
>>>>> rtcweb at ietf.org Subject: Re: [rtcweb] Confirmation of
>>>>> consensus on
>>>>> audio codecs
>>>>> On 08/29/2012 03:10 PM, Jean-Marc Valin wrote:
>>>> On 08/29/2012 05:51 AM, Stefan Hakansson LK wrote:
>>>>>> ...
>>>>>>>> That is great, but sort of underlines that there would
>>>>>>>> be no
>>>>>>>> harm in delaying the decision until there are
>>>>>>>> experiences
>>>>>>>> made from real world use - 'cause it would not be that
>>>>>>>> long
>>>>>>>> till that experience has been made (Markus also brought
>>>>>>>> up
>>>>>>>> the IPR status as a reason for waiting - I have no idea
>>>>>>>> how
>>>>>>>> long it would take to know more about that).
>>>> As Harald is pointing out, rtcweb implementations are going to
>>>> ship
>>>> pretty soon.
>>>>>> If that is the case (and I think and hope so), why would we
>>>>>> need
>>>>>> to make it MTI before seeing it in action?
>>>>>> [...]
>>>> Are you expecting *another* single, standardized, royalty-free
>>>> codec
>>>> that operates over vast ranges of bitrates and operating
>>>> conditions,
>>>> from narrowband speech to stereo music, all with low delay,
>>>> coming
>>>> out in the next year? If not, what are you really waiting for?
>>>>>> No, I'm not expecting that. I would just prefer us to see
>>>>>> that
>>>>>> Opus does indeed deliver as promised before making it MTI.
>>>>>> If it
>>>>>> does, fine. If not, we'd have to discuss what to do then.
>>>>>> To me it is like if you're going to place a bet, for an
>>>>>> upcoming
>>>>>> big race, on a horse that has never been in an actual race,
>>>>>> but
>>>>>> shows great promise. If you know that the horse is going
>>>>>> to
>>>>>> participate in a few practice races soon, would you not
>>>>>> prefer to
>>>>>> wait and see how it fares in those before placing the bet
>>>>>> (given
>>>>>> that the odds would not change)?
>>>>>> Anyway, that's my view. Let's see what the chairs say.
>>>>>>>> As Paul suggested, I was referring to the lack of
>>>>>>>> formal,
>>>>>>>> controlled, characterization tests. That is how other
>>>>>>>> SDOs do
>>>>>>>> it. I don't think that is the only way to do it, but I
>>>>>>>> think
>>>>>>>> we should at least have either such tests conducted or
>>>>>>>> experience from deployment and use (in a wide range of
>>>>>>>> conditions and device types) before making it MTI.
>>>> Opus has had "ITU-style" testing on English (
>>>> http://www.opus-codec.org/comparison/GoogleTest1.pdf ) and
>>>> Mandarin
>>>> ( http://www.opus-codec.org/comparison/GoogleTest2.pdf ). And
>>>> if you
>>>> don't trust Google on the tests I linked to, it's also been
>>>> tested
>>>> by Nokia:
>>>>>> http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_
>>> Quality_Characterization_of_IETF_Opus_Codec.pdf
>>>>>> Come on, those tests are very limited compared to a formal
>>>>>> characterization test. (Example:
>>>>>> www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx
>>>>>> <http://www.itu.int/dms_pub/itu-t/opb/tut/T-TUT-ASC-2010-MSW-E.docx>).
>>>>>>   There is very little info on the material used, the
>>>>>> environment,
>>>>>> processing and scripts and so on. And there seems to be no
>>>>>> tests
>>>>>> whatsoever (at least not involving humans) with actual
>>>>>> channels
>>>>>> introducing jitter and losses.
>>>>>> Note: I am in no way proposing that a formal
>>>>>> characterization
>>>>>> test is needed, or even the right thing to do. It is a
>>>>>> costly and
>>>>>> time consuming process, and alternative approaches could
>>>>>> prove to
>>>>>> be more efficient. What I am saying is that I think we
>>>>>> should not
>>>>>> mandate a codec that has neither gone through that kind of
>>>>>> formal
>>>>>> characterization testing nor has any field experience from
>>>>>> actual
>>>>>> use on a reasonable scale (covering different conditions
>>>>>> and
>>>>>> device types). It just seems wrong, especially given that
>>>>>> we
>>>>>> will soon have field experience.
>>>> Now, unlike other SDOs, the testing did not stop there. ITU-T
>>>> codecs
>>>> generally end up being testing with something in the order of
>>>> tens
>>>> of minutes worth of audio. In *addition* to that kind of
>>>> testing,
>>>> Opus also had automated testing with hundreds of years worth
>>>> of
>>>> audio. If anything, I think other SDOs should learn something
>>>> here.
>>>>>> This may very well be true. I guess this comes down to how
>>>>>> much
>>>>>> you trust that the quality assessment models (e.g. PEAQ,
>>>>>> POLQA)
>>>>>> give the same result as human test subjects would. But I
>>>>>> think
>>>>>> this sounds like a really good thing.
>>>> Jean-Marc
>>>>> _______________________________________________ rtcweb
>>>>> mailing
>>>>> list rtcweb at ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb at 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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> https://www.ietf.org/mailman/listinfo/rtcweb

From gmaxwell@juniper.net  Fri Aug 31 15:37:12 2012
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From: Gregory Maxwell <gmaxwell@juniper.net>
To: Randall Gellens <randy@qualcomm.com>, John Leslie <john@jlc.net>, Randell Jesup <randell-ietf@jesup.org>
Date: Fri, 31 Aug 2012 15:35:02 -0700
Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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Randall Gellens [randy@qualcomm.com] wrote:
> We can pick G.711 as MTI and rely on implementers to support others.

And then end up with a specification which is not actually interoperable fo=
r but a fraction of the well known intended applications.

From hacker.stefan@gmail.com  Fri Aug 31 16:15:40 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 01/09/2012 00:09, Randall Gellens wrote:
> Exactly.  The discussion has been going in my view off-track into 
> debates about which codec is best for which environments.  The real 
> issue is mandatory versus recommended.
>
> We can pick G.711 as MTI and rely on implementers to support others.
>
Imho this would be a big mistake. Even if implementers were to pick up 
Opus as quasi MTI after the fact, which I'm not convinced would reliably 
happen, this would critically damage the abilities of RTCWEB at launch.

Remember that the first implementations are not likely to venture far 
into "recommended" territory and for many Opus is still an unknown no 
matter its credentials and advantages.

Opus as an MTI offers the chance to make RTCWEB worth, even desirable, 
using Out-Of-The-Box for the majority of the intended use-cases. It 
ensures implementers have a codec at their hands that not only 
guarantees interoperability but also lets them pick the best possible 
quality, be it for low/high bandwidth or CPU or latency.

Regards,
Stefan

From ron@debian.org  Fri Aug 31 20:32:06 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On Fri, Aug 31, 2012 at 08:40:43AM -0700, Cameron Byrne wrote:
> Agree with the above. G.711 for mti only.
> 
> I want to push Opus and promote its use, but MTI is not the right method
> for that.

I haven't got the impression from this discussion that anybody thinks it
is about "pushing Opus" at all.  The question is what is best for rtcweb.

Since the charter of this group is to produce a standard that facilitates
"direct interactive rich communication", using things that already exist
as much as possible - I thought it was clear that the only important point
is that all this engineering work to produce rtcweb should not be a complete
waste of time for the people actually implementing it, and produce something
that gives such a poor experience in its baseline configuration that nobody
would ever actually want to use it for any of its intended purposes.

That Opus is a very obvious choice of existing technology, which no other
single codec comes anywhere close to matching for this purpose, stands out
like an elephant in the room.  Blind men might argue about whether it is a
pillar or a brush, or whether enough double blinded men have yet touched it
to be sure of this, but no amount of "pushing" is going to realistically
move it from that position one way or the other.

Pretending that other much-poorer codecs could fill this role is abandoning
the "rich communication" part of the charter, and condemning this group to
an embarrassing failure to achieve its goals in any meaningful way.

Let's not do that shall we.  It seems quite clear that many people have an
intense interest in its success.  Delaying this indefinitely, or selecting
poorer options than are already available today do not seem like the choices
that rational people who actually do want this to succeed would make.

Interoperability requires an adequate MTI codec.  Nobody has proposed an
alternative codec that is not clearly inadequate by comparison to Opus.
In a room full of clever engineers who want rtcweb to excel as a best of
breed specification, this decision should be a total no-brainer.

This time would be much better spent on the Actually Hard questions that
still remain for rtcweb.

 Best,
 Ron



From basilgohar@librevideo.org  Fri Aug 31 20:42:26 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On 08/31/2012 11:19 PM, Ron wrote:
> On Fri, Aug 31, 2012 at 08:40:43AM -0700, Cameron Byrne wrote:
>> Agree with the above. G.711 for mti only.
>>
>> I want to push Opus and promote its use, but MTI is not the right method
>> for that.
> I haven't got the impression from this discussion that anybody thinks it
> is about "pushing Opus" at all.  The question is what is best for rtcweb.
>
> Since the charter of this group is to produce a standard that facilitates
> "direct interactive rich communication", using things that already exist
> as much as possible - I thought it was clear that the only important point
> is that all this engineering work to produce rtcweb should not be a complete
> waste of time for the people actually implementing it, and produce something
> that gives such a poor experience in its baseline configuration that nobody
> would ever actually want to use it for any of its intended purposes.
>
> That Opus is a very obvious choice of existing technology, which no other
> single codec comes anywhere close to matching for this purpose, stands out
> like an elephant in the room.  Blind men might argue about whether it is a
> pillar or a brush, or whether enough double blinded men have yet touched it
> to be sure of this, but no amount of "pushing" is going to realistically
> move it from that position one way or the other.
>
> Pretending that other much-poorer codecs could fill this role is abandoning
> the "rich communication" part of the charter, and condemning this group to
> an embarrassing failure to achieve its goals in any meaningful way.
>
> Let's not do that shall we.  It seems quite clear that many people have an
> intense interest in its success.  Delaying this indefinitely, or selecting
> poorer options than are already available today do not seem like the choices
> that rational people who actually do want this to succeed would make.
>
> Interoperability requires an adequate MTI codec.  Nobody has proposed an
> alternative codec that is not clearly inadequate by comparison to Opus.
> In a room full of clever engineers who want rtcweb to excel as a best of
> breed specification, this decision should be a total no-brainer.
>
> This time would be much better spent on the Actually Hard questions that
> still remain for rtcweb.
>
>   Best,
>   Ron
+1, I couldn't have expressed myself any better than this, personally.

-- 
Libre Video
http://librevideo.org


From ron@debian.org  Fri Aug 31 20:43:12 2012
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From: Ron <ron@debian.org>
To: rtcweb@ietf.org
Message-ID: <20120901034309.GG23434@audi.shelbyville.oz>
References: <9E2843EA-EBB9-40B3-898C-6B5216FAE7A5@cisco.com> <CABRok6kSU9RFQo5GEQqtSRi2dTPV3fCN2MERfSQKorH1Hnbh4w@mail.gmail.com>
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Subject: Re: [rtcweb] Confirmation of consensus on audio codecs
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On Thu, Aug 30, 2012 at 01:46:21PM +0100, Neil Stratford wrote:
> On Thu, Aug 16, 2012 at 6:15 PM, Cullen Jennings (fluffy)
> <fluffy@cisco.com> wrote:
> > At the last meeting we took a hum on selecting Opus and G.711 as the mediatory to implement audio codecs. If there is any new opinions please send them to the list by August 30th, after which the chairs will make a determination of consensus.
> 
> I support both G.711 and Opus as mandatory to implement.
> 
> G.711 alone does not meet the rtcweb goal of producing a viable
> alternative to browser plugins as it will be impossible for an
> application developer to guarantee an acceptable audio quality
> (defined as at least as good as when using plugins) for browser to
> browser calls.

Indeed.  +1 for Opus and G.711 as MTI.

 Ron



From cb.list6@gmail.com  Fri Aug 31 21:38:14 2012
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From: Cameron Byrne <cb.list6@gmail.com>
To: Ron <ron@debian.org>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On Fri, Aug 31, 2012 at 8:19 PM, Ron <ron@debian.org> wrote:
> On Fri, Aug 31, 2012 at 08:40:43AM -0700, Cameron Byrne wrote:
>> Agree with the above. G.711 for mti only.
>>
>> I want to push Opus and promote its use, but MTI is not the right method
>> for that.
>
> I haven't got the impression from this discussion that anybody thinks it
> is about "pushing Opus" at all.  The question is what is best for rtcweb.
>
> Since the charter of this group is to produce a standard that facilitates
> "direct interactive rich communication", using things that already exist
> as much as possible - I thought it was clear that the only important point
> is that all this engineering work to produce rtcweb should not be a complete
> waste of time for the people actually implementing it, and produce something
> that gives such a poor experience in its baseline configuration that nobody
> would ever actually want to use it for any of its intended purposes.
>
> That Opus is a very obvious choice of existing technology, which no other
> single codec comes anywhere close to matching for this purpose, stands out
> like an elephant in the room.  Blind men might argue about whether it is a
> pillar or a brush, or whether enough double blinded men have yet touched it
> to be sure of this, but no amount of "pushing" is going to realistically
> move it from that position one way or the other.
>
> Pretending that other much-poorer codecs could fill this role is abandoning
> the "rich communication" part of the charter, and condemning this group to
> an embarrassing failure to achieve its goals in any meaningful way.
>
> Let's not do that shall we.  It seems quite clear that many people have an
> intense interest in its success.  Delaying this indefinitely, or selecting
> poorer options than are already available today do not seem like the choices
> that rational people who actually do want this to succeed would make.
>
> Interoperability requires an adequate MTI codec.  Nobody has proposed an
> alternative codec that is not clearly inadequate by comparison to Opus.
> In a room full of clever engineers who want rtcweb to excel as a best of
> breed specification, this decision should be a total no-brainer.
>

MTI is not about a room full of clever engineers, this thread and the
position statements from XYZ companies should make that clear.  MTI is
about the least common denominator for a series of latent issues,
including IPR FUD and legacy baggage.

Consenting adults are free to offer and answer whatever codecs they like.

CB

> This time would be much better spent on the Actually Hard questions that
> still remain for rtcweb.
>
>  Best,
>  Ron
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ron@debian.org  Fri Aug 31 23:12:42 2012
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From: Ron <ron@debian.org>
To: Cameron Byrne <cb.list6@gmail.com>
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References: <CAC8DBE4E9704C41BCB290C2F3CC921A162D278D@nasanexd01h.na.qualcomm.com> <503FC1BF.5020004@alvestrand.no> <CAC8DBE4E9704C41BCB290C2F3CC921A162D2B0F@nasanexd01h.na.qualcomm.com> <5040541C.5020008@alvestrand.no> <20120831133845.GW72831@verdi> <5040CE32.5050003@jesup.org> <20120831151247.GY72831@verdi> <CAD6AjGToznJtNdSzyFbxKhhXQTLTuOWnPutOYDCQCVH_8mRZ5w@mail.gmail.com> <20120901031955.GF23434@audi.shelbyville.oz> <CAD6AjGTN6JTQ50bj-6v2vvw56HOWNA7VmP-BZFwNWiX-c0AAnA@mail.gmail.com>
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On Fri, Aug 31, 2012 at 09:38:11PM -0700, Cameron Byrne wrote:
> On Fri, Aug 31, 2012 at 8:19 PM, Ron <ron@debian.org> wrote:
> > On Fri, Aug 31, 2012 at 08:40:43AM -0700, Cameron Byrne wrote:
> >> Agree with the above. G.711 for mti only.
> >>
> >> I want to push Opus and promote its use, but MTI is not the right method
> >> for that.
> >
> > I haven't got the impression from this discussion that anybody thinks it
> > is about "pushing Opus" at all.  The question is what is best for rtcweb.
> >
> > Since the charter of this group is to produce a standard that facilitates
> > "direct interactive rich communication", using things that already exist
> > as much as possible - I thought it was clear that the only important point
> > is that all this engineering work to produce rtcweb should not be a complete
> > waste of time for the people actually implementing it, and produce something
> > that gives such a poor experience in its baseline configuration that nobody
> > would ever actually want to use it for any of its intended purposes.
> >
> > That Opus is a very obvious choice of existing technology, which no other
> > single codec comes anywhere close to matching for this purpose, stands out
> > like an elephant in the room.  Blind men might argue about whether it is a
> > pillar or a brush, or whether enough double blinded men have yet touched it
> > to be sure of this, but no amount of "pushing" is going to realistically
> > move it from that position one way or the other.
> >
> > Pretending that other much-poorer codecs could fill this role is abandoning
> > the "rich communication" part of the charter, and condemning this group to
> > an embarrassing failure to achieve its goals in any meaningful way.
> >
> > Let's not do that shall we.  It seems quite clear that many people have an
> > intense interest in its success.  Delaying this indefinitely, or selecting
> > poorer options than are already available today do not seem like the choices
> > that rational people who actually do want this to succeed would make.
> >
> > Interoperability requires an adequate MTI codec.  Nobody has proposed an
> > alternative codec that is not clearly inadequate by comparison to Opus.
> > In a room full of clever engineers who want rtcweb to excel as a best of
> > breed specification, this decision should be a total no-brainer.
> >
> 
> MTI is not about a room full of clever engineers, this thread and the
> position statements from XYZ companies should make that clear.  MTI is
> about the least common denominator for a series of latent issues,

Yes, and least common denominator here doesn't mean "the worst possible
codec that we can think of which might be called a codec".  Nobody has
proposed any credible alternative to Opus that might reasonably be seen
as a plausible common denominator to a broad spectrum of the use cases.

Opus doesn't just simply cover that range.  It trounces pretty much every
other codec across the whole spectrum, except for a tiny little corner of
the extreme low-bandwidth case, where it's marginally worse than something
which already sounds terrible.

When your least common denominator also happens to be the best of breed
across that whole range, that's a Good Thing right?  Something to thank
your lucky engineering stars for, not something to prevaricate about.

> including IPR FUD and legacy baggage.

FUD has no place here whatsoever, however popular it might be.
Legacy baggage is a "will consider" item in the group charter, not a
"will be stupidly constrained by to the detriment of the project".

> Consenting adults are free to offer and answer whatever codecs they like.

Yes.  Yes, they are.  And nobody has said or is saying they can't do so.

It's also not the question here.  The question is what is the minimum
guarantee that any rtcweb user will have when talking to any other
rtcweb user, without first needing to sign consent forms and download
optional extras (possibly for a reasonably non-discriminatory fee,
and even possibly trojaned).

If the answer to that is "something terrible", then it's like suggesting
that "Consenting adults are free to put wheels on their car".

Of course they are, but it will be pretty useless for its intended purpose
without them - and nobody in their right mind would make something like
that deliberately.

 hth,
 Ron


