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After all the feedback, I've taken another whack at this.

It seems that the term "WebRTC endpoint" is already used widely enough 
that it's worth continuing to use it. So I ended up with the following 
suggested text for -overview's definitions.

Comments?
If this seems OK, I'll emit another -overview next week with these 
definitions.

--------------------------

    o  A WebRTC User Agent (also called an UA or browser) is something 
that conforms to both the protocol specification and the Javascript API 
defined above.

    o  A WebRTC device is something that conforms to the protocol
       specification, but does not claim to implement the Javascript API.

    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.

    o  A WebRTC-compatible endpoint is an endpoint that is capable of 
successfully communicating with a WebRTC endpoint, but may fail to meet 
some requirement of the WebRTC endpoint. This may limit where in the 
network such an endpoint can be attached, or may limit the security 
guarantees that it offers to others.

    o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates 
media traffic to non-WebRTC entities.

-----------------------------

FOR TRANSPORT:

A WebRTC-compatible endpoint is capable of inititating or accepting a 
session with a WebRTC endpoint. The following requirements on a WebRTC 
endpoint are not required for such success:

- Support for full ICE. If the endpoint is only ever going to be 
attached to the public Internet, it does not need to be able to fix its 
own external address; ICE-Lite is enough.
- Support for the full suite of MTI codecs for a WebRTC endpoint. In 
particular, audio gateways that connect to native G.711 networks may 
choose to implement G.711 and not implement Opus.
- Offering BUNDLE or RTCP-MUX
- Using MSID in its offers or answers
<should congestion cutoff requirement be in or out?>
<there will be more>

Note that support for DTLS, ICE and TURN ARE required for a 
WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be 
used.



From nobody Fri Oct  3 04:23:17 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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>     o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates
> media traffic to non-WebRTC entities.

> Note that support for DTLS, ICE and TURN ARE required for a
> WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
> used.
[Raju] Though a typical WebRTC-compatible endpoint requires TURN, I don't t=
hink WebRTC gateway requires TURN handling. TURN should probably be tied to=
 the use of full ICE or private ip; if you have public ip and use ice-lite =
then TURN support is not required.

BR
Raju


From nobody Fri Oct  3 05:01:28 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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Hi,

First, I personally see no need for all these definitions.

I think it would be enough to have:

- WebRTC endpoint (e.g. a browser)
- WebRTC-compatible endpoint (e.g. a gateway)

If people really think we need more, I won't argue against. I just think it=
 becomes very messy, and people WILL end up using the wrong terminology :)


Second, you say:

	"Note that support for DTLS, ICE and TURN ARE required for a WebRTC-compat=
ible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used."

You already in the bullet list said support of ICE lite, so the text is con=
flicting.=20

I am not sure what you mean by "support for TURN". An ICE lite endpoint wil=
l not create TURN candidates etc. Of course, it may receive media via a TUR=
N server.

What do you mean by "support for DTLS"? I think you need to be a little mor=
e specific (later you do mention DTLS-SRTP in case of RTP).

Regards,

Christer




-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestran=
d
Sent: 3. lokakuuta 2014 10:44
To: rtcweb@ietf.org
Subject: [rtcweb] Definitions of WebRTC entities

After all the feedback, I've taken another whack at this.

It seems that the term "WebRTC endpoint" is already used widely enough that=
 it's worth continuing to use it. So I ended up with the following suggeste=
d text for -overview's definitions.

Comments?
If this seems OK, I'll emit another -overview next week with these definiti=
ons.

--------------------------

    o  A WebRTC User Agent (also called an UA or browser) is something that=
 conforms to both the protocol specification and the Javascript API defined=
 above.

    o  A WebRTC device is something that conforms to the protocol
       specification, but does not claim to implement the Javascript API.

    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.

    o  A WebRTC-compatible endpoint is an endpoint that is capable of succe=
ssfully communicating with a WebRTC endpoint, but may fail to meet some req=
uirement of the WebRTC endpoint. This may limit where in the network such a=
n endpoint can be attached, or may limit the security guarantees that it of=
fers to others.

    o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates media=
 traffic to non-WebRTC entities.

-----------------------------

FOR TRANSPORT:

A WebRTC-compatible endpoint is capable of inititating or accepting a sessi=
on with a WebRTC endpoint. The following requirements on a WebRTC endpoint =
are not required for such success:

- Support for full ICE. If the endpoint is only ever going to be attached t=
o the public Internet, it does not need to be able to fix its own external =
address; ICE-Lite is enough.
- Support for the full suite of MTI codecs for a WebRTC endpoint. In partic=
ular, audio gateways that connect to native G.711 networks may choose to im=
plement G.711 and not implement Opus.
- Offering BUNDLE or RTCP-MUX
- Using MSID in its offers or answers
<should congestion cutoff requirement be in or out?> <there will be more>

Note that support for DTLS, ICE and TURN ARE required for a WebRTC-compatib=
le endpoint, and if RTP is used at all, DTLS-SRTP MUST be used.


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


From nobody Fri Oct  3 05:47:42 2014
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From: Harald Alvestrand <harald@alvestrand.no>
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Yup, turn is wrong. And ice needs to be made more precise - point is, without any ice, things will not work at all.

DTLS: A webrtc endpoint either uses data channels, which require dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I figured it was safe to say that dtls was required.

Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg <christer.holmberg@ericsson.com>:
>Hi,
>
>First, I personally see no need for all these definitions.
>
>I think it would be enough to have:
>
>- WebRTC endpoint (e.g. a browser)
>- WebRTC-compatible endpoint (e.g. a gateway)
>
>If people really think we need more, I won't argue against. I just
>think it becomes very messy, and people WILL end up using the wrong
>terminology :)
>
>
>Second, you say:
>
>	"Note that support for DTLS, ICE and TURN ARE required for a
>WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST
>be used."
>
>You already in the bullet list said support of ICE lite, so the text is
>conflicting. 
>
>I am not sure what you mean by "support for TURN". An ICE lite endpoint
>will not create TURN candidates etc. Of course, it may receive media
>via a TURN server.
>
>What do you mean by "support for DTLS"? I think you need to be a little
>more specific (later you do mention DTLS-SRTP in case of RTP).
>
>Regards,
>
>Christer
>
>
>
>
>-----Original Message-----
>From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>Alvestrand
>Sent: 3. lokakuuta 2014 10:44
>To: rtcweb@ietf.org
>Subject: [rtcweb] Definitions of WebRTC entities
>
>After all the feedback, I've taken another whack at this.
>
>It seems that the term "WebRTC endpoint" is already used widely enough
>that it's worth continuing to use it. So I ended up with the following
>suggested text for -overview's definitions.
>
>Comments?
>If this seems OK, I'll emit another -overview next week with these
>definitions.
>
>--------------------------
>
>o  A WebRTC User Agent (also called an UA or browser) is something that
>conforms to both the protocol specification and the Javascript API
>defined above.
>
>    o  A WebRTC device is something that conforms to the protocol
>     specification, but does not claim to implement the Javascript API.
>
>    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
>
>o  A WebRTC-compatible endpoint is an endpoint that is capable of
>successfully communicating with a WebRTC endpoint, but may fail to meet
>some requirement of the WebRTC endpoint. This may limit where in the
>network such an endpoint can be attached, or may limit the security
>guarantees that it offers to others.
>
>o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates media
>traffic to non-WebRTC entities.
>
>-----------------------------
>
>FOR TRANSPORT:
>
>A WebRTC-compatible endpoint is capable of inititating or accepting a
>session with a WebRTC endpoint. The following requirements on a WebRTC
>endpoint are not required for such success:
>
>- Support for full ICE. If the endpoint is only ever going to be
>attached to the public Internet, it does not need to be able to fix its
>own external address; ICE-Lite is enough.
>- Support for the full suite of MTI codecs for a WebRTC endpoint. In
>particular, audio gateways that connect to native G.711 networks may
>choose to implement G.711 and not implement Opus.
>- Offering BUNDLE or RTCP-MUX
>- Using MSID in its offers or answers
><should congestion cutoff requirement be in or out?> <there will be
>more>
>
>Note that support for DTLS, ICE and TURN ARE required for a
>WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST
>be used.
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head></head><body>Yup, turn is wrong. And ice needs to be made more precise - point is, without any ice, things will not work at all.<br>
<br>
DTLS: A webrtc endpoint either uses data channels, which require dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I figured it was safe to say that dtls was required.<br><br><div class="gmail_quote">Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg &lt;christer.holmberg@ericsson.com&gt;:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre class="k9mail">Hi,<br /><br />First, I personally see no need for all these definitions.<br /><br />I think it would be enough to have:<br /><br />- WebRTC endpoint (e.g. a browser)<br />- WebRTC-compatible endpoint (e.g. a gateway)<br /><br />If people really think we need more, I won't argue against. I just think it becomes very messy, and people WILL end up using the wrong terminology :)<br /><br /><br />Second, you say:<br /><br /> "Note that support for DTLS, ICE and TURN ARE required for a WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used."<br /><br />You already in the bullet list said support of ICE lite, so the text is conflicting. <br /><br />I am not sure what you mean by "support for TURN". An ICE lite endpoint will not create TURN candidates etc. Of course, it may receive media via a TURN server.<br /><br />What do you mean by "support for DTLS"? I think you need to be a little more specific (later you do mention DTLS-SRTP in case of
RTP).<br /><br />Regards,<br /><br />Christer<br /><br /><br /><br /><br />-----Original Message-----<br />From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand<br />Sent: 3. lokakuuta 2014 10:44<br />To: rtcweb@ietf.org<br />Subject: [rtcweb] Definitions of WebRTC entities<br /><br />After all the feedback, I've taken another whack at this.<br /><br />It seems that the term "WebRTC endpoint" is already used widely enough that it's worth continuing to use it. So I ended up with the following suggested text for -overview's definitions.<br /><br />Comments?<br />If this seems OK, I'll emit another -overview next week with these definitions.<br /><br />--------------------------<br /><br />    o  A WebRTC User Agent (also called an UA or browser) is something that conforms to both the protocol specification and the Javascript API defined above.<br /><br />    o  A WebRTC device is something that conforms to the protocol<br />       specification, but does not
claim to implement the Javascript API.<br /><br />    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.<br /><br />    o  A WebRTC-compatible endpoint is an endpoint that is capable of successfully communicating with a WebRTC endpoint, but may fail to meet some requirement of the WebRTC endpoint. This may limit where in the network such an endpoint can be attached, or may limit the security guarantees that it offers to others.<br /><br />    o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates media traffic to non-WebRTC entities.<br /><br />-----------------------------<br /><br />FOR TRANSPORT:<br /><br />A WebRTC-compatible endpoint is capable of inititating or accepting a session with a WebRTC endpoint. The following requirements on a WebRTC endpoint are not required for such success:<br /><br />- Support for full ICE. If the endpoint is only ever going to be attached to the public Internet, it does not need to be able to fix its own external address;
ICE-Lite is enough.<br />- Support for the full suite of MTI codecs for a WebRTC endpoint. In particular, audio gateways that connect to native G.711 networks may choose to implement G.711 and not implement Opus.<br />- Offering BUNDLE or RTCP-MUX<br />- Using MSID in its offers or answers<br />&lt;should congestion cutoff requirement be in or out?&gt; &lt;there will be more&gt;<br /><br />Note that support for DTLS, ICE and TURN ARE required for a WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used.<br /><br /><br /><hr /><br />rtcweb mailing list<br />rtcweb@ietf.org<br /><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br /></pre></blockquote></div><br>
-- <br>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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From nobody Fri Oct  3 10:36:20 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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Date: Fri, 3 Oct 2014 17:36:14 +0000
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Date: Sat, 4 Oct 2014 06:41:23 +1000
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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These are good!

On 3 Oct 2014 17:44, "Harald Alvestrand" <harald@alvestrand.no> wrote:
>
> After all the feedback, I've taken another whack at this.
>
> It seems that the term "WebRTC endpoint" is already used widely enough
that it's worth continuing to use it. So I ended up with the following
suggested text for -overview's definitions.
>
> Comments?
> If this seems OK, I'll emit another -overview next week with these
definitions.
>
> --------------------------
>
>    o  A WebRTC User Agent (also called an UA or browser) is something
that conforms to both the protocol specification and the Javascript API
defined above.
>
>    o  A WebRTC device is something that conforms to the protocol
>       specification, but does not claim to implement the Javascript API.
>
>    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.

Since a webrtc UA is a superset of a webrtc device, webrtc endpoint and
webrtc device end up meaning the same, don't they?

>    o  A WebRTC-compatible endpoint is an endpoint that is capable of
successfully communicating with a WebRTC endpoint, but may fail to meet
some requirement of the WebRTC endpoint. This may limit where in the
network such an endpoint can be attached, or may limit the security
guarantees that it offers to others.
>
>    o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates
media traffic to non-WebRTC entities.
>
> -----------------------------
>
> FOR TRANSPORT:
>
> A WebRTC-compatible endpoint is capable of inititating or accepting a
session with a WebRTC endpoint. The following requirements on a WebRTC
endpoint are not required for such success:
>
> - Support for full ICE. If the endpoint is only ever going to be attached
to the public Internet, it does not need to be able to fix its own external
address; ICE-Lite is enough.
> - Support for the full suite of MTI codecs for a WebRTC endpoint. In
particular, audio gateways that connect to native G.711 networks may choose
to implement G.711 and not implement Opus.
> - Offering BUNDLE or RTCP-MUX
> - Using MSID in its offers or answers
> <should congestion cutoff requirement be in or out?>
> <there will be more>
>
> Note that support for DTLS, ICE and TURN ARE required for a
WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
used.
>
>

Is your intent to specify which parts of the protocol are not required to
be supported by webrtc-compatible endpoints in a rfc? I think it would be
useful...

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

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<p dir=3D"ltr">These are good!</p>
<p dir=3D"ltr">On 3 Oct 2014 17:44, &quot;Harald Alvestrand&quot; &lt;<a hr=
ef=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt;<br>
&gt; After all the feedback, I&#39;ve taken another whack at this.<br>
&gt;<br>
&gt; It seems that the term &quot;WebRTC endpoint&quot; is already used wid=
ely enough that it&#39;s worth continuing to use it. So I ended up with the=
 following suggested text for -overview&#39;s definitions.<br>
&gt;<br>
&gt; Comments?<br>
&gt; If this seems OK, I&#39;ll emit another -overview next week with these=
 definitions.<br>
&gt;<br>
&gt; --------------------------<br>
&gt;<br>
&gt; =C2=A0 =C2=A0o=C2=A0 A WebRTC User Agent (also called an UA or browser=
) is something that conforms to both the protocol specification and the Jav=
ascript API defined above.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0o=C2=A0 A WebRTC device is something that conforms to the=
 protocol<br>
&gt; =C2=A0 =C2=A0 =C2=A0 specification, but does not claim to implement th=
e Javascript API.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0o=C2=A0 A WebRTC endpoint is either a WebRTC UA or a WebR=
TC device.</p>
<p dir=3D"ltr">Since a webrtc UA is a superset of a webrtc device, webrtc e=
ndpoint and webrtc device end up meaning the same, don&#39;t they?<br></p>
<p dir=3D"ltr">&gt; =C2=A0 =C2=A0o=C2=A0 A WebRTC-compatible endpoint is an=
 endpoint that is capable of successfully communicating with a WebRTC endpo=
int, but may fail to meet some requirement of the WebRTC endpoint. This may=
 limit where in the network such an endpoint can be attached, or may limit =
the security guarantees that it offers to others.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0o=C2=A0 A WebRTC gateway is a WebRTC-compatible endpoint =
that mediates media traffic to non-WebRTC entities.<br>
&gt;<br>
&gt; -----------------------------<br>
&gt;<br>
&gt; FOR TRANSPORT:<br>
&gt;<br>
&gt; A WebRTC-compatible endpoint is capable of inititating or accepting a =
session with a WebRTC endpoint. The following requirements on a WebRTC endp=
oint are not required for such success:<br>
&gt;<br>
&gt; - Support for full ICE. If the endpoint is only ever going to be attac=
hed to the public Internet, it does not need to be able to fix its own exte=
rnal address; ICE-Lite is enough.<br>
&gt; - Support for the full suite of MTI codecs for a WebRTC endpoint. In p=
articular, audio gateways that connect to native G.711 networks may choose =
to implement G.711 and not implement Opus.<br>
&gt; - Offering BUNDLE or RTCP-MUX<br>
&gt; - Using MSID in its offers or answers<br>
&gt; &lt;should congestion cutoff requirement be in or out?&gt;<br>
&gt; &lt;there will be more&gt;<br>
&gt;<br>
&gt; Note that support for DTLS, ICE and TURN ARE required for a WebRTC-com=
patible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used.<br>
&gt;<br>
&gt;</p>
<p dir=3D"ltr">Is your intent to specify which parts of the protocol are no=
t required to be supported by webrtc-compatible endpoints in a rfc? I think=
 it would be useful...</p>
<p dir=3D"ltr">Silvia.<br>
 _______________________________________________<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>

--20cf303bfada9c779c05048ac1aa--


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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Den 03. okt. 2014 22:41, skrev Silvia Pfeiffer:
>
> These are good!
>
> On 3 Oct 2014 17:44, "Harald Alvestrand" <harald@alvestrand.no 
> <mailto:harald@alvestrand.no>> wrote:
> >
> > After all the feedback, I've taken another whack at this.
> >
> > It seems that the term "WebRTC endpoint" is already used widely 
> enough that it's worth continuing to use it. So I ended up with the 
> following suggested text for -overview's definitions.
> >
> > Comments?
> > If this seems OK, I'll emit another -overview next week with these 
> definitions.
> >
> > --------------------------
> >
> >    o  A WebRTC User Agent (also called an UA or browser) is 
> something that conforms to both the protocol specification and the 
> Javascript API defined above.
> >
> >    o  A WebRTC device is something that conforms to the protocol
> >       specification, but does not claim to implement the Javascript API.
> >
> >    o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
>
> Since a webrtc UA is a superset of a webrtc device, webrtc endpoint 
> and webrtc device end up meaning the same, don't they?
>
> >    o  A WebRTC-compatible endpoint is an endpoint that is capable of 
> successfully communicating with a WebRTC endpoint, but may fail to 
> meet some requirement of the WebRTC endpoint. This may limit where in 
> the network such an endpoint can be attached, or may limit the 
> security guarantees that it offers to others.
> >
> >    o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates 
> media traffic to non-WebRTC entities.
> >
> > -----------------------------
> >
> > FOR TRANSPORT:
> >
> > A WebRTC-compatible endpoint is capable of inititating or accepting 
> a session with a WebRTC endpoint. The following requirements on a 
> WebRTC endpoint are not required for such success:
> >
> > - Support for full ICE. If the endpoint is only ever going to be 
> attached to the public Internet, it does not need to be able to fix 
> its own external address; ICE-Lite is enough.
> > - Support for the full suite of MTI codecs for a WebRTC endpoint. In 
> particular, audio gateways that connect to native G.711 networks may 
> choose to implement G.711 and not implement Opus.
> > - Offering BUNDLE or RTCP-MUX
> > - Using MSID in its offers or answers
> > <should congestion cutoff requirement be in or out?>
> > <there will be more>
> >
> > Note that support for DTLS, ICE and TURN ARE required for a 
> WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST 
> be used.
> >
> >
>
> Is your intent to specify which parts of the protocol are not required 
> to be supported by webrtc-compatible endpoints in a rfc? I think it 
> would be useful...
>

Yes, -transport is intended to be published as an RFC. So is 
draft-alvestrand-rtcweb-gateways (if the group accepts it).


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


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    <div class="moz-cite-prefix">Den 03. okt. 2014 22:41, skrev Silvia
      Pfeiffer:<br>
    </div>
    <blockquote
cite="mid:CAHp8n2kV_LmKkBtzxdC1GZcBow1yq6=sZd2H5n2Zen7j5qVvFQ@mail.gmail.com"
      type="cite">
      <p dir="ltr">These are good!</p>
      <p dir="ltr">On 3 Oct 2014 17:44, "Harald Alvestrand" &lt;<a
          moz-do-not-send="true" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;
        wrote:<br>
        &gt;<br>
        &gt; After all the feedback, I've taken another whack at this.<br>
        &gt;<br>
        &gt; It seems that the term "WebRTC endpoint" is already used
        widely enough that it's worth continuing to use it. So I ended
        up with the following suggested text for -overview's
        definitions.<br>
        &gt;<br>
        &gt; Comments?<br>
        &gt; If this seems OK, I'll emit another -overview next week
        with these definitions.<br>
        &gt;<br>
        &gt; --------------------------<br>
        &gt;<br>
        &gt; Â  Â oÂ  A WebRTC User Agent (also called an UA or browser) is
        something that conforms to both the protocol specification and
        the Javascript API defined above.<br>
        &gt;<br>
        &gt; Â  Â oÂ  A WebRTC device is something that conforms to the
        protocol<br>
        &gt; Â  Â  Â  specification, but does not claim to implement the
        Javascript API.<br>
        &gt;<br>
        &gt; Â  Â oÂ  A WebRTC endpoint is either a WebRTC UA or a WebRTC
        device.</p>
      <p dir="ltr">Since a webrtc UA is a superset of a webrtc device,
        webrtc endpoint and webrtc device end up meaning the same, don't
        they?<br>
      </p>
      <p dir="ltr">&gt; Â  Â oÂ  A WebRTC-compatible endpoint is an
        endpoint that is capable of successfully communicating with a
        WebRTC endpoint, but may fail to meet some requirement of the
        WebRTC endpoint. This may limit where in the network such an
        endpoint can be attached, or may limit the security guarantees
        that it offers to others.<br>
        &gt;<br>
        &gt; Â  Â oÂ  A WebRTC gateway is a WebRTC-compatible endpoint that
        mediates media traffic to non-WebRTC entities.<br>
        &gt;<br>
        &gt; -----------------------------<br>
        &gt;<br>
        &gt; FOR TRANSPORT:<br>
        &gt;<br>
        &gt; A WebRTC-compatible endpoint is capable of inititating or
        accepting a session with a WebRTC endpoint. The following
        requirements on a WebRTC endpoint are not required for such
        success:<br>
        &gt;<br>
        &gt; - Support for full ICE. If the endpoint is only ever going
        to be attached to the public Internet, it does not need to be
        able to fix its own external address; ICE-Lite is enough.<br>
        &gt; - Support for the full suite of MTI codecs for a WebRTC
        endpoint. In particular, audio gateways that connect to native
        G.711 networks may choose to implement G.711 and not implement
        Opus.<br>
        &gt; - Offering BUNDLE or RTCP-MUX<br>
        &gt; - Using MSID in its offers or answers<br>
        &gt; &lt;should congestion cutoff requirement be in or out?&gt;<br>
        &gt; &lt;there will be more&gt;<br>
        &gt;<br>
        &gt; Note that support for DTLS, ICE and TURN ARE required for a
        WebRTC-compatible endpoint, and if RTP is used at all, DTLS-SRTP
        MUST be used.<br>
        &gt;<br>
        &gt;</p>
      <p dir="ltr">Is your intent to specify which parts of the protocol
        are not required to be supported by webrtc-compatible endpoints
        in a rfc? I think it would be useful...</p>
    </blockquote>
    <br>
    Yes, -transport is intended to be published as an RFC. So is
    draft-alvestrand-rtcweb-gateways (if the group accepts it).<br>
    <br>
    <br>
    <blockquote
cite="mid:CAHp8n2kV_LmKkBtzxdC1GZcBow1yq6=sZd2H5n2Zen7j5qVvFQ@mail.gmail.com"
      type="cite">
      <p dir="ltr">Silvia.<br>
        _______________________________________________<br>
        &gt; rtcweb mailing list<br>
        &gt; <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
        &gt; <a moz-do-not-send="true"
          href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
      </p>
    </blockquote>
    <br>
  </body>
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
References: <542E53D2.5040500@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465376@ESESSMB209.ericsson.se> <C45C84E3-FC63-4DF6-ABDE-701FC7584E3C@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465985@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D465A34@ESESSMB209.ericsson.se>
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Date: Tue, 7 Oct 2014 21:51:07 +0530
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/hbIcIOjkrzelSxytgNDcoPlDwAY
Subject: Re: [rtcweb] Definitions of WebRTC entities
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Hi Christer,

I have no issue with WebRTC User Agent, WebRTC device, WebRTC endpoint.

I have bit trouble with WebRTC compatible endpoint as a entity name as=20

1) It may pass SRTP/data channel=20
2) It is not required to be endpoint but it shall be middle box.

WebRTC gateway looks more appropriate entity name in those scenarios.

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: Friday, October 03, 2014 11:06 PM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Definitions of WebRTC entities
>=20
> Hi,
>=20
> >>DTLS: A webrtc endpoint either uses data channels, which require
> dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I
> figured it
> >>was safe to say that dtls was required.
> >
> >I think it would be better to explicitly indicate the usages for =
which
> DTLS needs to be supported, ie DTLS-SRTP for RTP and as defined for
> data channels.
> >Because, DTLS can be used for many different purposes, in different
> ways, so just saying =E2=80=9Csupport DTLS=E2=80=9D is unclear.
>=20
> In addition, it is probably useful to indicate that an compatible
> endpoint may not necessarily terminate all DTLS usages. For example, a
> gateway might simply pass through the data channel, and/or the SRTP
> traffic.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
> Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg
> <christer.holmberg@ericsson.com>:
> Hi,
>=20
> First, I personally see no need for all these definitions.
>=20
> I think it would be enough to have:
>=20
> - WebRTC endpoint (e.g. a browser)
> - WebRTC-compatible endpoint (e.g. a gateway)
>=20
> If people really think we need more, I won't argue against. I just
> think it becomes very messy, and people WILL end up using the wrong
> terminology :)
>=20
>=20
> Second, you say:
>=20
>  "Note that support for DTLS, ICE and TURN ARE required for a WebRTC-
> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
> used."
>=20
> You already in the bullet list said support of ICE lite, so the text =
is
> conflicting.
>=20
> I am not sure what you mean by "support for TURN". An ICE lite =
endpoint
> will not create TURN candidates etc. Of course, it may receive media
> via a TURN server.
>=20
> What do you mean by "support for DTLS"? I think you need to be a =
little
> more specific (later you do mention DTLS-SRTP in case of
> RTP).
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
> Sent: 3. lokakuuta 2014 10:44
> To: rtcweb@ietf.org
> Subject: [rtcweb] Definitions of WebRTC entities
>=20
> After all the feedback, I've taken another whack at this.
>=20
> It seems that the term "WebRTC endpoint" is already used widely enough
> that it's worth continuing to use it. So I ended up with the following
> suggested text for -overview's definitions.
>=20
> Comments?
> If this seems OK, I'll emit another -overview next week with these
> definitions.
>=20
> --------------------------
>=20
>     o  A WebRTC User Agent (also called an UA or browser) is something
> that conforms to both the protocol specification and the Javascript =
API
> defined above.
>=20
>     o  A WebRTC device is something that conforms to the protocol
>        specification, but does not
> claim to implement the Javascript API.
>=20
>     o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
>=20
>     o  A WebRTC-compatible endpoint is an endpoint that is capable of
> successfully communicating with a WebRTC endpoint, but may fail to =
meet
> some requirement of the WebRTC endpoint. This may limit where in the
> network such an endpoint can be attached, or may limit the security
> guarantees that it offers to others.
>=20
>     o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates
> media traffic to non-WebRTC entities.
>=20
> -----------------------------
>=20
> FOR TRANSPORT:
>=20
> A WebRTC-compatible endpoint is capable of inititating or accepting a
> session with a WebRTC endpoint. The following requirements on a WebRTC
> endpoint are not required for such success:
>=20
> - Support for full ICE. If the endpoint is only ever going to be
> attached to the public Internet, it does not need to be able to fix =
its
> own external address;
> ICE-Lite is enough.
> - Support for the full suite of MTI codecs for a WebRTC endpoint. In
> particular, audio gateways that connect to native G.711 networks may
> choose to implement G.711 and not implement Opus.
> - Offering BUNDLE or RTCP-MUX
> - Using MSID in its offers or answers
> <should congestion cutoff requirement be in or out?> <there will be
> more>
>=20
> Note that support for DTLS, ICE and TURN ARE required for a WebRTC-
> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be =
used.
> ________________________________________
>=20
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> --
> Sent from my Android device with K-9 Mail. Please excuse my brevity.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Oct  7 09:46:24 2014
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From: Harald Alvestrand <harald@alvestrand.no>
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To: Parthasarathi R <partha@parthasarathi.co.in>,  'Christer Holmberg' <christer.holmberg@ericsson.com>, rtcweb@ietf.org
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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On 10/07/2014 06:21 PM, Parthasarathi R wrote:
> Hi Christer,
>
> I have no issue with WebRTC User Agent, WebRTC device, WebRTC endpoint.
>
> I have bit trouble with WebRTC compatible endpoint as a entity name as 
>
> 1) It may pass SRTP/data channel 
What do you mean by "pass"? That's a slippery term.
> 2) It is not required to be endpoint but it shall be middle box.
What do you mean by "middle box"? Again, that term is slippery.
>
> WebRTC gateway looks more appropriate entity name in those scenarios.

As written in my proposal, a WebRTC gateway is a WebRTC compatible endpoint.

I think you will have to describe more fully what the device you are
thinking about looks like; if it's just the WebRTC endpoint functions
scattered over a set of boxes, it's possible I may call it a
"distributed WebRTC-compatible endpoint".

>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
>> Holmberg
>> Sent: Friday, October 03, 2014 11:06 PM
>> To: Harald Alvestrand; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> Hi,
>>
>>>> DTLS: A webrtc endpoint either uses data channels, which require
>> dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I
>> figured it
>>>> was safe to say that dtls was required.
>>> I think it would be better to explicitly indicate the usages for which
>> DTLS needs to be supported, ie DTLS-SRTP for RTP and as defined for
>> data channels.
>>> Because, DTLS can be used for many different purposes, in different
>> ways, so just saying â€œsupport DTLSâ€ is unclear.
>>
>> In addition, it is probably useful to indicate that an compatible
>> endpoint may not necessarily terminate all DTLS usages. For example, a
>> gateway might simply pass through the data channel, and/or the SRTP
>> traffic.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>> Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg
>> <christer.holmberg@ericsson.com>:
>> Hi,
>>
>> First, I personally see no need for all these definitions.
>>
>> I think it would be enough to have:
>>
>> - WebRTC endpoint (e.g. a browser)
>> - WebRTC-compatible endpoint (e.g. a gateway)
>>
>> If people really think we need more, I won't argue against. I just
>> think it becomes very messy, and people WILL end up using the wrong
>> terminology :)
>>
>>
>> Second, you say:
>>
>>  "Note that support for DTLS, ICE and TURN ARE required for a WebRTC-
>> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
>> used."
>>
>> You already in the bullet list said support of ICE lite, so the text is
>> conflicting.
>>
>> I am not sure what you mean by "support for TURN". An ICE lite endpoint
>> will not create TURN candidates etc. Of course, it may receive media
>> via a TURN server.
>>
>> What do you mean by "support for DTLS"? I think you need to be a little
>> more specific (later you do mention DTLS-SRTP in case of
>> RTP).
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>> Alvestrand
>> Sent: 3. lokakuuta 2014 10:44
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Definitions of WebRTC entities
>>
>> After all the feedback, I've taken another whack at this.
>>
>> It seems that the term "WebRTC endpoint" is already used widely enough
>> that it's worth continuing to use it. So I ended up with the following
>> suggested text for -overview's definitions.
>>
>> Comments?
>> If this seems OK, I'll emit another -overview next week with these
>> definitions.
>>
>> --------------------------
>>
>>     o  A WebRTC User Agent (also called an UA or browser) is something
>> that conforms to both the protocol specification and the Javascript API
>> defined above.
>>
>>     o  A WebRTC device is something that conforms to the protocol
>>        specification, but does not
>> claim to implement the Javascript API.
>>
>>     o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
>>
>>     o  A WebRTC-compatible endpoint is an endpoint that is capable of
>> successfully communicating with a WebRTC endpoint, but may fail to meet
>> some requirement of the WebRTC endpoint. This may limit where in the
>> network such an endpoint can be attached, or may limit the security
>> guarantees that it offers to others.
>>
>>     o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates
>> media traffic to non-WebRTC entities.
>>
>> -----------------------------
>>
>> FOR TRANSPORT:
>>
>> A WebRTC-compatible endpoint is capable of inititating or accepting a
>> session with a WebRTC endpoint. The following requirements on a WebRTC
>> endpoint are not required for such success:
>>
>> - Support for full ICE. If the endpoint is only ever going to be
>> attached to the public Internet, it does not need to be able to fix its
>> own external address;
>> ICE-Lite is enough.
>> - Support for the full suite of MTI codecs for a WebRTC endpoint. In
>> particular, audio gateways that connect to native G.711 networks may
>> choose to implement G.711 and not implement Opus.
>> - Offering BUNDLE or RTCP-MUX
>> - Using MSID in its offers or answers
>> <should congestion cutoff requirement be in or out?> <there will be
>> more>
>>
>> Note that support for DTLS, ICE and TURN ARE required for a WebRTC-
>> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used.
>> ________________________________________
>>
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> --
>> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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From nobody Tue Oct  7 10:34:52 2014
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/ewyzq3umq2MUoYm3Sahvg6Gfob8
Subject: Re: [rtcweb] Definitions of WebRTC entities
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On 10/07/2014 07:14 PM, Christer Holmberg wrote:
> Hi,
>
> In my opinion we have TWO types of ENDPOINTS: those that support the full set of RTCWEB toolset (e.g. browsers) and those who support a subset (e.g. gateways).
>
> I don't really care what we call them, but we shouldn't come up with lots of different names that people are going to use wrong.

People have also wanted to claim that something is fully
WebRTC-compliant without having a Javascript API on it. That's where the
"UA/browser" vs "Device" thing comes in.

I'm trying to make everyone happy :-)

>
> Regards,
>
> Christer
>
>
>
>
> -----Original Message-----
> From: Parthasarathi R [mailto:partha@parthasarathi.co.in] 
> Sent: 07 October 2014 19:21
> To: Christer Holmberg; 'Harald Alvestrand'; rtcweb@ietf.org
> Subject: RE: [rtcweb] Definitions of WebRTC entities
>
> Hi Christer,
>
> I have no issue with WebRTC User Agent, WebRTC device, WebRTC endpoint.
>
> I have bit trouble with WebRTC compatible endpoint as a entity name as 
>
> 1) It may pass SRTP/data channel
> 2) It is not required to be endpoint but it shall be middle box.
>
> WebRTC gateway looks more appropriate entity name in those scenarios.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer 
>> Holmberg
>> Sent: Friday, October 03, 2014 11:06 PM
>> To: Harald Alvestrand; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> Hi,
>>
>>>> DTLS: A webrtc endpoint either uses data channels, which require
>> dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I 
>> figured it
>>>> was safe to say that dtls was required.
>>> I think it would be better to explicitly indicate the usages for 
>>> which
>> DTLS needs to be supported, ie DTLS-SRTP for RTP and as defined for 
>> data channels.
>>> Because, DTLS can be used for many different purposes, in different
>> ways, so just saying â€œsupport DTLSâ€ is unclear.
>>
>> In addition, it is probably useful to indicate that an compatible 
>> endpoint may not necessarily terminate all DTLS usages. For example, a 
>> gateway might simply pass through the data channel, and/or the SRTP 
>> traffic.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>> Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg
>> <christer.holmberg@ericsson.com>:
>> Hi,
>>
>> First, I personally see no need for all these definitions.
>>
>> I think it would be enough to have:
>>
>> - WebRTC endpoint (e.g. a browser)
>> - WebRTC-compatible endpoint (e.g. a gateway)
>>
>> If people really think we need more, I won't argue against. I just 
>> think it becomes very messy, and people WILL end up using the wrong 
>> terminology :)
>>
>>
>> Second, you say:
>>
>>  "Note that support for DTLS, ICE and TURN ARE required for a WebRTC- 
>> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be 
>> used."
>>
>> You already in the bullet list said support of ICE lite, so the text 
>> is conflicting.
>>
>> I am not sure what you mean by "support for TURN". An ICE lite 
>> endpoint will not create TURN candidates etc. Of course, it may 
>> receive media via a TURN server.
>>
>> What do you mean by "support for DTLS"? I think you need to be a 
>> little more specific (later you do mention DTLS-SRTP in case of RTP).
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald 
>> Alvestrand
>> Sent: 3. lokakuuta 2014 10:44
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Definitions of WebRTC entities
>>
>> After all the feedback, I've taken another whack at this.
>>
>> It seems that the term "WebRTC endpoint" is already used widely enough 
>> that it's worth continuing to use it. So I ended up with the following 
>> suggested text for -overview's definitions.
>>
>> Comments?
>> If this seems OK, I'll emit another -overview next week with these 
>> definitions.
>>
>> --------------------------
>>
>>     o  A WebRTC User Agent (also called an UA or browser) is something 
>> that conforms to both the protocol specification and the Javascript 
>> API defined above.
>>
>>     o  A WebRTC device is something that conforms to the protocol
>>        specification, but does not
>> claim to implement the Javascript API.
>>
>>     o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
>>
>>     o  A WebRTC-compatible endpoint is an endpoint that is capable of 
>> successfully communicating with a WebRTC endpoint, but may fail to 
>> meet some requirement of the WebRTC endpoint. This may limit where in 
>> the network such an endpoint can be attached, or may limit the 
>> security guarantees that it offers to others.
>>
>>     o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates 
>> media traffic to non-WebRTC entities.
>>
>> -----------------------------
>>
>> FOR TRANSPORT:
>>
>> A WebRTC-compatible endpoint is capable of inititating or accepting a 
>> session with a WebRTC endpoint. The following requirements on a WebRTC 
>> endpoint are not required for such success:
>>
>> - Support for full ICE. If the endpoint is only ever going to be 
>> attached to the public Internet, it does not need to be able to fix 
>> its own external address; ICE-Lite is enough.
>> - Support for the full suite of MTI codecs for a WebRTC endpoint. In 
>> particular, audio gateways that connect to native G.711 networks may 
>> choose to implement G.711 and not implement Opus.
>> - Offering BUNDLE or RTCP-MUX
>> - Using MSID in its offers or answers
>> <should congestion cutoff requirement be in or out?> <there will be
>> more>
>>
>> Note that support for DTLS, ICE and TURN ARE required for a WebRTC- 
>> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be used.
>> ________________________________________
>>
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> --
>> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, Parthasarathi R <partha@parthasarathi.co.in>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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From nobody Tue Oct  7 20:52:02 2014
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/q7WL4WFJ5tpvtxYcL0dt1h0oIjo
Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Hello,

I have a couple of comments on draft-ietf-rtcweb-stun-consent-freshness-07:

1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both "ICE 
binding requests" and "STUN binding requests". Is there a distinction 
between them? e.g. "ICE binding requests" refer to 
7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request indicates 
a non-extended binding request as per RFC5389.

2) Just to confirm when Consent is first given. Is the 30sec consent 
period assumed after the initial ICE exchange? I found the second 
paragraph in cl.4.1 not so clear as it talks about subsequent consent 
before the initial consent in paragraph 5. I'd suggest modifying it 
along the lines of:
"An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP, 
DTLS), over any transport protocol (e.g., UDP, TCP) on an ICE-initiated 
connection unless the receiving endpoint consents to receive the data.  
Initial consent is granted as a result of a successful ICE connectivity 
check on a particular transport address. Consent expires after a fixed 
amount of time. As a result, subsequent consent MUST be obtained 
following the procedure described in this document.

During ICE restart consent checks MUST continue to be sent on previously 
validated pair, and MUST be responded to on the previously validated 
pair, until ICE restart completes."

Regards, Christian


From nobody Tue Oct  7 23:41:17 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christian Groves <Christian.Groves@nteczone.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Hi Christian,

Thanks for your comments. Please see inline

-----Original Message-----
From: Christian Groves <Christian.Groves@nteczone.com>
Date: Wednesday, 8 October 2014 9:21 am
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness

>Hello,
>
>I have a couple of comments on
>draft-ietf-rtcweb-stun-consent-freshness-07:
>
>1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both "ICE
>binding requests" and "STUN binding requests". Is there a distinction
>between them? e.g. "ICE binding requests" refer to
>7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request indicates
>a non-extended binding request as per RFC5389.

We will change it to STUN binding requests to keep it consistent with the
rest of the document.

>
>2) Just to confirm when Consent is first given. Is the 30sec consent
>period assumed after the initial ICE exchange?

Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.

"The initial Consent to send traffic is obtained by ICE.  Consent
   expires after 30 seconds."


> I found the second
>paragraph in cl.4.1 not so clear as it talks about subsequent consent
>before the initial consent in paragraph 5. I'd suggest modifying it
>along the lines of:
>"An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
>DTLS), over any transport protocol (e.g., UDP, TCP) on an ICE-initiated
>connection unless the receiving endpoint consents to receive the data.
>Initial consent is granted as a result of a successful ICE connectivity
>check on a particular transport address. Consent expires after a fixed
>amount of time. As a result, subsequent consent MUST be obtained
>following the procedure described in this document.
>
>During ICE restart consent checks MUST continue to be sent on previously
>validated pair, and MUST be responded to on the previously validated
>pair, until ICE restart completes."

Sure will make this change.

Regards,
Ram

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


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, <rtcweb@ietf.org>
References: <542E53D2.5040500@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465376@ESESSMB209.ericsson.se> <C45C84E3-FC63-4DF6-ABDE-701FC7584E3C@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465985@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D465A34@ESESSMB209.ericsson.se> <00f501cfe24a$b8515930$28f40b90$@co.in> <543418D5.8010509@alvestrand.no>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Hi Harald,

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Tuesday, October 07, 2014 10:16 PM
> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> Subject: Re: [rtcweb] Definitions of WebRTC entities
>=20
> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
> > Hi Christer,
> >
> > I have no issue with WebRTC User Agent, WebRTC device, WebRTC
> endpoint.
> >
> > I have bit trouble with WebRTC compatible endpoint as a entity name
> as
> >
> > 1) It may pass SRTP/data channel
> What do you mean by "pass"? That's a slippery term.
<Partha> "relay" will be more appropriate term as mentioned in Sec 5 =
Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>

> > 2) It is not required to be endpoint but it shall be middle box.
> What do you mean by "middle box"? Again, that term is slippery.

<Partha> I intent to say that the entity which is between two endpoints =
and it does not end any media session itself. Here, The confusion is =
that WebRTC compatible endpoint which is not an endpoint but it is a =
middle box. </Partha>

> >
> > WebRTC gateway looks more appropriate entity name in those =
scenarios.
>=20
> As written in my proposal, a WebRTC gateway is a WebRTC compatible
> endpoint.
>=20
<Partha> As per your proposal, we need to define WebRTC compatible =
endpoint first which is super set of WebRTC gateway. Then, we need to =
clarify which kind of WebRTC compatible endpoint qualify as WebRTC =
gateway. But Christer wishes to have only two definition (Full/Subset).  =
</Partha>

> I think you will have to describe more fully what the device you are
> thinking about looks like; if it's just the WebRTC endpoint functions
> scattered over a set of boxes, it's possible I may call it a
> "distributed WebRTC-compatible endpoint".
>=20
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
> >> Holmberg
> >> Sent: Friday, October 03, 2014 11:06 PM
> >> To: Harald Alvestrand; rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Definitions of WebRTC entities
> >>
> >> Hi,
> >>
> >>>> DTLS: A webrtc endpoint either uses data channels, which require
> >> dtls, or rtp, whuch requires DTLS-srtp, which requires dtls, so I
> >> figured it
> >>>> was safe to say that dtls was required.
> >>> I think it would be better to explicitly indicate the usages for
> which
> >> DTLS needs to be supported, ie DTLS-SRTP for RTP and as defined for
> >> data channels.
> >>> Because, DTLS can be used for many different purposes, in =
different
> >> ways, so just saying =E2=80=9Csupport DTLS=E2=80=9D is unclear.
> >>
> >> In addition, it is probably useful to indicate that an compatible
> >> endpoint may not necessarily terminate all DTLS usages. For =
example,
> a
> >> gateway might simply pass through the data channel, and/or the SRTP
> >> traffic.
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >>
> >>
> >>
> >> Den 3. oktober 2014 14:01:20 CEST, skrev Christer Holmberg
> >> <christer.holmberg@ericsson.com>:
> >> Hi,
> >>
> >> First, I personally see no need for all these definitions.
> >>
> >> I think it would be enough to have:
> >>
> >> - WebRTC endpoint (e.g. a browser)
> >> - WebRTC-compatible endpoint (e.g. a gateway)
> >>
> >> If people really think we need more, I won't argue against. I just
> >> think it becomes very messy, and people WILL end up using the wrong
> >> terminology :)
> >>
> >>
> >> Second, you say:
> >>
> >>  "Note that support for DTLS, ICE and TURN ARE required for a
> WebRTC-
> >> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
> >> used."
> >>
> >> You already in the bullet list said support of ICE lite, so the =
text
> is
> >> conflicting.
> >>
> >> I am not sure what you mean by "support for TURN". An ICE lite
> endpoint
> >> will not create TURN candidates etc. Of course, it may receive =
media
> >> via a TURN server.
> >>
> >> What do you mean by "support for DTLS"? I think you need to be a
> little
> >> more specific (later you do mention DTLS-SRTP in case of
> >> RTP).
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >>
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> >> Alvestrand
> >> Sent: 3. lokakuuta 2014 10:44
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] Definitions of WebRTC entities
> >>
> >> After all the feedback, I've taken another whack at this.
> >>
> >> It seems that the term "WebRTC endpoint" is already used widely
> enough
> >> that it's worth continuing to use it. So I ended up with the
> following
> >> suggested text for -overview's definitions.
> >>
> >> Comments?
> >> If this seems OK, I'll emit another -overview next week with these
> >> definitions.
> >>
> >> --------------------------
> >>
> >>     o  A WebRTC User Agent (also called an UA or browser) is
> something
> >> that conforms to both the protocol specification and the Javascript
> API
> >> defined above.
> >>
> >>     o  A WebRTC device is something that conforms to the protocol
> >>        specification, but does not
> >> claim to implement the Javascript API.
> >>
> >>     o  A WebRTC endpoint is either a WebRTC UA or a WebRTC device.
> >>
> >>     o  A WebRTC-compatible endpoint is an endpoint that is capable
> of
> >> successfully communicating with a WebRTC endpoint, but may fail to
> meet
> >> some requirement of the WebRTC endpoint. This may limit where in =
the
> >> network such an endpoint can be attached, or may limit the security
> >> guarantees that it offers to others.
> >>
> >>     o  A WebRTC gateway is a WebRTC-compatible endpoint that
> mediates
> >> media traffic to non-WebRTC entities.
> >>
> >> -----------------------------
> >>
> >> FOR TRANSPORT:
> >>
> >> A WebRTC-compatible endpoint is capable of inititating or accepting
> a
> >> session with a WebRTC endpoint. The following requirements on a
> WebRTC
> >> endpoint are not required for such success:
> >>
> >> - Support for full ICE. If the endpoint is only ever going to be
> >> attached to the public Internet, it does not need to be able to fix
> its
> >> own external address;
> >> ICE-Lite is enough.
> >> - Support for the full suite of MTI codecs for a WebRTC endpoint. =
In
> >> particular, audio gateways that connect to native G.711 networks =
may
> >> choose to implement G.711 and not implement Opus.
> >> - Offering BUNDLE or RTCP-MUX
> >> - Using MSID in its offers or answers
> >> <should congestion cutoff requirement be in or out?> <there will be
> >> more>
> >>
> >> Note that support for DTLS, ICE and TURN ARE required for a WebRTC-
> >> compatible endpoint, and if RTP is used at all, DTLS-SRTP MUST be
> used.
> >> ________________________________________
> >>
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >> --
> >> Sent from my Android device with K-9 Mail. Please excuse my =
brevity.
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --
> Surveillance is pervasive. Go Dark.


From nobody Wed Oct  8 19:00:00 2014
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Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Hello Ram,

Thanks for the response. A further question. The draft focuses on sender 
behaviour with regards to consent, i.e. it indicates that an endpoint 
must not send data if consent hasn't been granted. What about receiver 
behaviour? E.g. what happens if the sending end point is not well 
behaved and sends application data beyond the consent period, what 
should the receiver do?

Regards, Christian

On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
> Hi Christian,
>
> Thanks for your comments. Please see inline
>
> -----Original Message-----
> From: Christian Groves <Christian.Groves@nteczone.com>
> Date: Wednesday, 8 October 2014 9:21 am
> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>
>> Hello,
>>
>> I have a couple of comments on
>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>
>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both "ICE
>> binding requests" and "STUN binding requests". Is there a distinction
>> between them? e.g. "ICE binding requests" refer to
>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request indicates
>> a non-extended binding request as per RFC5389.
> We will change it to STUN binding requests to keep it consistent with the
> rest of the document.
>
>> 2) Just to confirm when Consent is first given. Is the 30sec consent
>> period assumed after the initial ICE exchange?
> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>
> "The initial Consent to send traffic is obtained by ICE.  Consent
>     expires after 30 seconds."
>
>
>> I found the second
>> paragraph in cl.4.1 not so clear as it talks about subsequent consent
>> before the initial consent in paragraph 5. I'd suggest modifying it
>> along the lines of:
>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
>> DTLS), over any transport protocol (e.g., UDP, TCP) on an ICE-initiated
>> connection unless the receiving endpoint consents to receive the data.
>> Initial consent is granted as a result of a successful ICE connectivity
>> check on a particular transport address. Consent expires after a fixed
>> amount of time. As a result, subsequent consent MUST be obtained
>> following the procedure described in this document.
>>
>> During ICE restart consent checks MUST continue to be sent on previously
>> validated pair, and MUST be responded to on the previously validated
>> pair, until ICE restart completes."
> Sure will make this change.
>
> Regards,
> Ram
>
>> Regards, Christian
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


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To: Christian Groves <Christian.Groves@nteczone.com>, "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
> Groves
> Sent: Thursday, October 09, 2014 7:30 AM
> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
> Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshnes=
s
>=20
> Hello Ram,
>=20
> Thanks for the response. A further question. The draft focuses on sender
> behaviour with regards to consent, i.e. it indicates that an endpoint mus=
t not
> send data if consent hasn't been granted. What about receiver behaviour?
> E.g. what happens if the sending end point is not well behaved and sends
> application data beyond the consent period, what should the receiver do ?

The receiver can use DTLS fatal error to immediately revoke the consent. Bu=
t if sender does not care about consent revocation then the receiver has to=
 rely on other techniques (like closing the FW mapping etc.).  The javascri=
pt may be malicious, hence consent checks are done by the browser (which se=
rves as user's TCB)) and browser has to be trusted to do the right thing.

-Tiru

>=20
> Regards, Christian
>=20
> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
> > Hi Christian,
> >
> > Thanks for your comments. Please see inline
> >
> > -----Original Message-----
> > From: Christian Groves <Christian.Groves@nteczone.com>
> > Date: Wednesday, 8 October 2014 9:21 am
> > To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> > Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
> >
> >> Hello,
> >>
> >> I have a couple of comments on
> >> draft-ietf-rtcweb-stun-consent-freshness-07:
> >>
> >> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
> >> "ICE binding requests" and "STUN binding requests". Is there a
> >> distinction between them? e.g. "ICE binding requests" refer to
> >> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
> >> indicates a non-extended binding request as per RFC5389.
> > We will change it to STUN binding requests to keep it consistent with
> > the rest of the document.
> >
> >> 2) Just to confirm when Consent is first given. Is the 30sec consent
> >> period assumed after the initial ICE exchange?
> > Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
> >
> > "The initial Consent to send traffic is obtained by ICE.  Consent
> >     expires after 30 seconds."
> >
> >
> >> I found the second
> >> paragraph in cl.4.1 not so clear as it talks about subsequent consent
> >> before the initial consent in paragraph 5. I'd suggest modifying it
> >> along the lines of:
> >> "An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
> >> DTLS), over any transport protocol (e.g., UDP, TCP) on an
> >> ICE-initiated connection unless the receiving endpoint consents to rec=
eive
> the data.
> >> Initial consent is granted as a result of a successful ICE
> >> connectivity check on a particular transport address. Consent expires
> >> after a fixed amount of time. As a result, subsequent consent MUST be
> >> obtained following the procedure described in this document.
> >>
> >> During ICE restart consent checks MUST continue to be sent on
> >> previously validated pair, and MUST be responded to on the previously
> >> validated pair, until ICE restart completes."
> > Sure will make this change.
> >
> > Regards,
> > Ram
> >
> >> Regards, Christian
> >>
> >> _______________________________________________
> >> 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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Hello Tiru,

So does this mean that the receiver also has to keep track of 30 second 
consent period to be able to identify the situation?

Regards, Christian

On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
>> Groves
>> Sent: Thursday, October 09, 2014 7:30 AM
>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
>> Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>>
>> Hello Ram,
>>
>> Thanks for the response. A further question. The draft focuses on sender
>> behaviour with regards to consent, i.e. it indicates that an endpoint must not
>> send data if consent hasn't been granted. What about receiver behaviour?
>> E.g. what happens if the sending end point is not well behaved and sends
>> application data beyond the consent period, what should the receiver do ?
> The receiver can use DTLS fatal error to immediately revoke the consent. But if sender does not care about consent revocation then the receiver has to rely on other techniques (like closing the FW mapping etc.).  The javascript may be malicious, hence consent checks are done by the browser (which serves as user's TCB)) and browser has to be trusted to do the right thing.
>
> -Tiru
>
>> Regards, Christian
>>
>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
>>> Hi Christian,
>>>
>>> Thanks for your comments. Please see inline
>>>
>>> -----Original Message-----
>>> From: Christian Groves <Christian.Groves@nteczone.com>
>>> Date: Wednesday, 8 October 2014 9:21 am
>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>> Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>>>
>>>> Hello,
>>>>
>>>> I have a couple of comments on
>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>>>
>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
>>>> "ICE binding requests" and "STUN binding requests". Is there a
>>>> distinction between them? e.g. "ICE binding requests" refer to
>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
>>>> indicates a non-extended binding request as per RFC5389.
>>> We will change it to STUN binding requests to keep it consistent with
>>> the rest of the document.
>>>
>>>> 2) Just to confirm when Consent is first given. Is the 30sec consent
>>>> period assumed after the initial ICE exchange?
>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>>>
>>> "The initial Consent to send traffic is obtained by ICE.  Consent
>>>      expires after 30 seconds."
>>>
>>>
>>>> I found the second
>>>> paragraph in cl.4.1 not so clear as it talks about subsequent consent
>>>> before the initial consent in paragraph 5. I'd suggest modifying it
>>>> along the lines of:
>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
>>>> DTLS), over any transport protocol (e.g., UDP, TCP) on an
>>>> ICE-initiated connection unless the receiving endpoint consents to receive
>> the data.
>>>> Initial consent is granted as a result of a successful ICE
>>>> connectivity check on a particular transport address. Consent expires
>>>> after a fixed amount of time. As a result, subsequent consent MUST be
>>>> obtained following the procedure described in this document.
>>>>
>>>> During ICE restart consent checks MUST continue to be sent on
>>>> previously validated pair, and MUST be responded to on the previously
>>>> validated pair, until ICE restart completes."
>>> Sure will make this change.
>>>
>>> Regards,
>>> Ram
>>>
>>>> Regards, Christian
>>>>
>>>> _______________________________________________
>>>> 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] Definitions of WebRTC entities
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On 10/08/2014 06:39 PM, Parthasarathi R wrote:
> Hi Harald,
>
> Please read inline.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Sent: Tuesday, October 07, 2014 10:16 PM
>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
>>> Hi Christer,
>>>
>>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
>> endpoint.
>>> I have bit trouble with WebRTC compatible endpoint as a entity name
>> as
>>> 1) It may pass SRTP/data channel
>> What do you mean by "pass"? That's a slippery term.
> <Partha> "relay" will be more appropriate term as mentioned in Sec 5 Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
>
>>> 2) It is not required to be endpoint but it shall be middle box.
>> What do you mean by "middle box"? Again, that term is slippery.
> <Partha> I intent to say that the entity which is between two endpoints and it does not end any media session itself. Here, The confusion is that WebRTC compatible endpoint which is not an endpoint but it is a middle box. </Partha>

Seems that this entity (whatever it's called) isn't an endpoint at all,
so defining terms for endpoints shouldn't be relevant to whatever this
device is.

There's always more boxes in the middle..... although as long as they
don't have the DTLS keys, it's limited what they can do to the packets.



>
>>> WebRTC gateway looks more appropriate entity name in those scenarios.
>> As written in my proposal, a WebRTC gateway is a WebRTC compatible
>> endpoint.
>>
> <Partha> As per your proposal, we need to define WebRTC compatible endpoint first which is super set of WebRTC gateway. Then, we need to clarify which kind of WebRTC compatible endpoint qualify as WebRTC gateway. But Christer wishes to have only two definition (Full/Subset).  </Partha>

And I don't agree with Christer, so then we're two :-)


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On 10/09/2014 08:50 AM, Christian Groves wrote:
> Hello Tiru,
>
> So does this mean that the receiver also has to keep track of 30
> second consent period to be able to identify the situation?

If the receiver cares about being flooded without its consent, the
receiver needs to detect that situation and take appropriate action.
Either by watching the 30-second timer or by some other means.

If the receiver doesn't care, it doesn't have to do anything.

I don't think the spec needs to mandate any particular reaction from the
receiver in this situation.




>
> Regards, Christian
>
> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
>>> -----Original Message-----
>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
>>> Groves
>>> Sent: Thursday, October 09, 2014 7:30 AM
>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Question on
>>> draft-ietf-rtcweb-stun-consent-freshness
>>>
>>> Hello Ram,
>>>
>>> Thanks for the response. A further question. The draft focuses on
>>> sender
>>> behaviour with regards to consent, i.e. it indicates that an
>>> endpoint must not
>>> send data if consent hasn't been granted. What about receiver
>>> behaviour?
>>> E.g. what happens if the sending end point is not well behaved and
>>> sends
>>> application data beyond the consent period, what should the receiver
>>> do ?
>> The receiver can use DTLS fatal error to immediately revoke the
>> consent. But if sender does not care about consent revocation then
>> the receiver has to rely on other techniques (like closing the FW
>> mapping etc.).  The javascript may be malicious, hence consent checks
>> are done by the browser (which serves as user's TCB)) and browser has
>> to be trusted to do the right thing.
>>
>> -Tiru
>>
>>> Regards, Christian
>>>
>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
>>>> Hi Christian,
>>>>
>>>> Thanks for your comments. Please see inline
>>>>
>>>> -----Original Message-----
>>>> From: Christian Groves <Christian.Groves@nteczone.com>
>>>> Date: Wednesday, 8 October 2014 9:21 am
>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>>> Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>>>>
>>>>> Hello,
>>>>>
>>>>> I have a couple of comments on
>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>>>>
>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
>>>>> "ICE binding requests" and "STUN binding requests". Is there a
>>>>> distinction between them? e.g. "ICE binding requests" refer to
>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
>>>>> indicates a non-extended binding request as per RFC5389.
>>>> We will change it to STUN binding requests to keep it consistent with
>>>> the rest of the document.
>>>>
>>>>> 2) Just to confirm when Consent is first given. Is the 30sec consent
>>>>> period assumed after the initial ICE exchange?
>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>>>>
>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
>>>>      expires after 30 seconds."
>>>>
>>>>
>>>>> I found the second
>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent consent
>>>>> before the initial consent in paragraph 5. I'd suggest modifying it
>>>>> along the lines of:
>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
>>>>> DTLS), over any transport protocol (e.g., UDP, TCP) on an
>>>>> ICE-initiated connection unless the receiving endpoint consents to
>>>>> receive
>>> the data.
>>>>> Initial consent is granted as a result of a successful ICE
>>>>> connectivity check on a particular transport address. Consent expires
>>>>> after a fixed amount of time. As a result, subsequent consent MUST be
>>>>> obtained following the procedure described in this document.
>>>>>
>>>>> During ICE restart consent checks MUST continue to be sent on
>>>>> previously validated pair, and MUST be responded to on the previously
>>>>> validated pair, until ICE restart completes."
>>>> Sure will make this change.
>>>>
>>>> Regards,
>>>> Ram
>>>>
>>>>> Regards, Christian
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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I don't think it should mandate a particular action but I think it would 
be good to mention some of the possibilities. As it stands nothing is 
mentioned in the draft about the role of the receiver.

If the receiver watches the 30 second timer, how does it distinguish 
between receiving a STUN binding request for the purposes of Consent 
(i.e. linked to application data signalling) and a STUN binding request 
sent for the purposes of a keep alive? I guess that it uses the fact 
that consent related STUN binding request are authenticated and 
keepalive STUN binding requests are not (as per cl.10/RFC5245. Is that 
correct?

Regards, Christian

On 9/10/2014 6:00 PM, Harald Alvestrand wrote:
> On 10/09/2014 08:50 AM, Christian Groves wrote:
>> Hello Tiru,
>>
>> So does this mean that the receiver also has to keep track of 30
>> second consent period to be able to identify the situation?
> If the receiver cares about being flooded without its consent, the
> receiver needs to detect that situation and take appropriate action.
> Either by watching the 30-second timer or by some other means.
>
> If the receiver doesn't care, it doesn't have to do anything.
>
> I don't think the spec needs to mandate any particular reaction from the
> receiver in this situation.
>
>
>
>
>> Regards, Christian
>>
>> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
>>>> -----Original Message-----
>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
>>>> Groves
>>>> Sent: Thursday, October 09, 2014 7:30 AM
>>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] Question on
>>>> draft-ietf-rtcweb-stun-consent-freshness
>>>>
>>>> Hello Ram,
>>>>
>>>> Thanks for the response. A further question. The draft focuses on
>>>> sender
>>>> behaviour with regards to consent, i.e. it indicates that an
>>>> endpoint must not
>>>> send data if consent hasn't been granted. What about receiver
>>>> behaviour?
>>>> E.g. what happens if the sending end point is not well behaved and
>>>> sends
>>>> application data beyond the consent period, what should the receiver
>>>> do ?
>>> The receiver can use DTLS fatal error to immediately revoke the
>>> consent. But if sender does not care about consent revocation then
>>> the receiver has to rely on other techniques (like closing the FW
>>> mapping etc.).  The javascript may be malicious, hence consent checks
>>> are done by the browser (which serves as user's TCB)) and browser has
>>> to be trusted to do the right thing.
>>>
>>> -Tiru
>>>
>>>> Regards, Christian
>>>>
>>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
>>>>> Hi Christian,
>>>>>
>>>>> Thanks for your comments. Please see inline
>>>>>
>>>>> -----Original Message-----
>>>>> From: Christian Groves <Christian.Groves@nteczone.com>
>>>>> Date: Wednesday, 8 October 2014 9:21 am
>>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>>>> Subject: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I have a couple of comments on
>>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>>>>>
>>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
>>>>>> "ICE binding requests" and "STUN binding requests". Is there a
>>>>>> distinction between them? e.g. "ICE binding requests" refer to
>>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
>>>>>> indicates a non-extended binding request as per RFC5389.
>>>>> We will change it to STUN binding requests to keep it consistent with
>>>>> the rest of the document.
>>>>>
>>>>>> 2) Just to confirm when Consent is first given. Is the 30sec consent
>>>>>> period assumed after the initial ICE exchange?
>>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>>>>>
>>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
>>>>>       expires after 30 seconds."
>>>>>
>>>>>
>>>>>> I found the second
>>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent consent
>>>>>> before the initial consent in paragraph 5. I'd suggest modifying it
>>>>>> along the lines of:
>>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP, SCTP,
>>>>>> DTLS), over any transport protocol (e.g., UDP, TCP) on an
>>>>>> ICE-initiated connection unless the receiving endpoint consents to
>>>>>> receive
>>>> the data.
>>>>>> Initial consent is granted as a result of a successful ICE
>>>>>> connectivity check on a particular transport address. Consent expires
>>>>>> after a fixed amount of time. As a result, subsequent consent MUST be
>>>>>> obtained following the procedure described in this document.
>>>>>>
>>>>>> During ICE restart consent checks MUST continue to be sent on
>>>>>> previously validated pair, and MUST be responded to on the previously
>>>>>> validated pair, until ICE restart completes."
>>>>> Sure will make this change.
>>>>>
>>>>> Regards,
>>>>> Ram
>>>>>
>>>>>> Regards, Christian
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [xrblock] I-D Action: draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00.txt
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Subject: [rtcweb] FW: [xrblock] I-D Action: draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00.txt
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FYI - this Internet-Draft was submitted in the XRBLOCK WG.=20

Comments are welcome.=20

Regards,

Dan


> -----Original Message-----
> From: xrblock [mailto:xrblock-bounces@ietf.org] On Behalf Of internet-
> drafts@ietf.org
> Sent: Thursday, October 09, 2014 11:27 AM
> To: i-d-announce@ietf.org
> Cc: xrblock@ietf.org
> Subject: [xrblock] I-D Action: draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-=
00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Metric Blocks for use with RTCP's Exten=
ded
> Report Framework Working Group of the IETF.
>=20
>         Title           : Considerations for Selecting RTCP Extended Repo=
rt (XR)
> Metrics for the RTCWEB Statistics API
>         Authors         : Rachel Huang
>                           Roni Even
>                           Varun Singh
>                           Dan Romascanu
>                           Lingli Deng
> 	Filename        : draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00.txt
> 	Pages           : 15
> 	Date            : 2014-10-08
>=20
> Abstract:
>    This document describes monitoring features related to RTCWEB.  It
>    provides a list of RTCP XR metrics, which are useful and may need to
>    be supported in some RTCWEB implementations.  It also describes a
>    list of additional identifiers for WebRTC's statistics API.  These
>    identifiers are a set of RTCP XR metrics related to the transport of
>    multimedia flows.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-xrblock-rtcweb-rtcp-xr-metric=
s/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> xrblock mailing list
> xrblock@ietf.org
> https://www.ietf.org/mailman/listinfo/xrblock


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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Christian Groves <Christian.Groves@nteczone.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
> Groves
> Sent: Thursday, October 09, 2014 12:51 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshnes=
s
>=20
> I don't think it should mandate a particular action but I think it would =
be
> good to mention some of the possibilities. As it stands nothing is mentio=
ned
> in the draft about the role of the receiver.

The role of receiver to immediately revoke consent either using STUN error =
code or DTLS fatal alert is discussed in =20
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-07#sect=
ion-4.2

-Tiru

>=20
> If the receiver watches the 30 second timer, how does it distinguish betw=
een
> receiving a STUN binding request for the purposes of Consent (i.e. linked=
 to
> application data signalling) and a STUN binding request sent for the purp=
oses
> of a keep alive? I guess that it uses the fact that consent related STUN =
binding
> request are authenticated and keepalive STUN binding requests are not (as
> per cl.10/RFC5245. Is that correct?
>=20
> Regards, Christian
>=20
> On 9/10/2014 6:00 PM, Harald Alvestrand wrote:
> > On 10/09/2014 08:50 AM, Christian Groves wrote:
> >> Hello Tiru,
> >>
> >> So does this mean that the receiver also has to keep track of 30
> >> second consent period to be able to identify the situation?
> > If the receiver cares about being flooded without its consent, the
> > receiver needs to detect that situation and take appropriate action.
> > Either by watching the 30-second timer or by some other means.
> >
> > If the receiver doesn't care, it doesn't have to do anything.
> >
> > I don't think the spec needs to mandate any particular reaction from
> > the receiver in this situation.
> >
> >
> >
> >
> >> Regards, Christian
> >>
> >> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
> >>>> -----Original Message-----
> >>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> >>>> Christian Groves
> >>>> Sent: Thursday, October 09, 2014 7:30 AM
> >>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
> >>>> Subject: Re: [rtcweb] Question on
> >>>> draft-ietf-rtcweb-stun-consent-freshness
> >>>>
> >>>> Hello Ram,
> >>>>
> >>>> Thanks for the response. A further question. The draft focuses on
> >>>> sender behaviour with regards to consent, i.e. it indicates that an
> >>>> endpoint must not send data if consent hasn't been granted. What
> >>>> about receiver behaviour?
> >>>> E.g. what happens if the sending end point is not well behaved and
> >>>> sends application data beyond the consent period, what should the
> >>>> receiver do ?
> >>> The receiver can use DTLS fatal error to immediately revoke the
> >>> consent. But if sender does not care about consent revocation then
> >>> the receiver has to rely on other techniques (like closing the FW
> >>> mapping etc.).  The javascript may be malicious, hence consent
> >>> checks are done by the browser (which serves as user's TCB)) and
> >>> browser has to be trusted to do the right thing.
> >>>
> >>> -Tiru
> >>>
> >>>> Regards, Christian
> >>>>
> >>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
> >>>>> Hi Christian,
> >>>>>
> >>>>> Thanks for your comments. Please see inline
> >>>>>
> >>>>> -----Original Message-----
> >>>>> From: Christian Groves <Christian.Groves@nteczone.com>
> >>>>> Date: Wednesday, 8 October 2014 9:21 am
> >>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> >>>>> Subject: [rtcweb] Question on
> >>>>> draft-ietf-rtcweb-stun-consent-freshness
> >>>>>
> >>>>>> Hello,
> >>>>>>
> >>>>>> I have a couple of comments on
> >>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
> >>>>>>
> >>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
> >>>>>> "ICE binding requests" and "STUN binding requests". Is there a
> >>>>>> distinction between them? e.g. "ICE binding requests" refer to
> >>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
> >>>>>> indicates a non-extended binding request as per RFC5389.
> >>>>> We will change it to STUN binding requests to keep it consistent
> >>>>> with the rest of the document.
> >>>>>
> >>>>>> 2) Just to confirm when Consent is first given. Is the 30sec
> >>>>>> consent period assumed after the initial ICE exchange?
> >>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
> >>>>>
> >>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
> >>>>>       expires after 30 seconds."
> >>>>>
> >>>>>
> >>>>>> I found the second
> >>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent
> >>>>>> consent before the initial consent in paragraph 5. I'd suggest
> >>>>>> modifying it along the lines of:
> >>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP,
> >>>>>> SCTP, DTLS), over any transport protocol (e.g., UDP, TCP) on an
> >>>>>> ICE-initiated connection unless the receiving endpoint consents
> >>>>>> to receive
> >>>> the data.
> >>>>>> Initial consent is granted as a result of a successful ICE
> >>>>>> connectivity check on a particular transport address. Consent
> >>>>>> expires after a fixed amount of time. As a result, subsequent
> >>>>>> consent MUST be obtained following the procedure described in this
> document.
> >>>>>>
> >>>>>> During ICE restart consent checks MUST continue to be sent on
> >>>>>> previously validated pair, and MUST be responded to on the
> >>>>>> previously validated pair, until ICE restart completes."
> >>>>> Sure will make this change.
> >>>>>
> >>>>> Regards,
> >>>>> Ram
> >>>>>
> >>>>>> Regards, Christian
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> rtcweb mailing list
> >>>>>> rtcweb@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>>> _______________________________________________
> >>>> rtcweb mailing list
> >>>> rtcweb@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/rtcweb
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/6tSNzOCt3DoMp0lUrIk_VgZMwrQ
Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Hello Tiru,

I've seen that text but it talks about the receiving endpoint of a STUN 
binding response. I was referring to the endpoint that receives the STUN 
binding request and sends the response. The draft concentrates on the 
STUN binding request initiator and its notion of consent. It doesn't 
explain what notion of consent that the receiver of the STUN binding 
request. E.g. a STUN binding request receiver may close the connection 
which leads to an immediate revokation but it doesn't mean it has the 
notion of consent. As Harald indicates below the receiver may not care 
at all about it.

Regards, Christian

On 9/10/2014 7:29 PM, Tirumaleswar Reddy (tireddy) wrote:
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
>> Groves
>> Sent: Thursday, October 09, 2014 12:51 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
>>
>> I don't think it should mandate a particular action but I think it would be
>> good to mention some of the possibilities. As it stands nothing is mentioned
>> in the draft about the role of the receiver.
> The role of receiver to immediately revoke consent either using STUN error code or DTLS fatal alert is discussed in
> http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-07#section-4.2
>
> -Tiru
>
>> If the receiver watches the 30 second timer, how does it distinguish between
>> receiving a STUN binding request for the purposes of Consent (i.e. linked to
>> application data signalling) and a STUN binding request sent for the purposes
>> of a keep alive? I guess that it uses the fact that consent related STUN binding
>> request are authenticated and keepalive STUN binding requests are not (as
>> per cl.10/RFC5245. Is that correct?
>>
>> Regards, Christian
>>
>> On 9/10/2014 6:00 PM, Harald Alvestrand wrote:
>>> On 10/09/2014 08:50 AM, Christian Groves wrote:
>>>> Hello Tiru,
>>>>
>>>> So does this mean that the receiver also has to keep track of 30
>>>> second consent period to be able to identify the situation?
>>> If the receiver cares about being flooded without its consent, the
>>> receiver needs to detect that situation and take appropriate action.
>>> Either by watching the 30-second timer or by some other means.
>>>
>>> If the receiver doesn't care, it doesn't have to do anything.
>>>
>>> I don't think the spec needs to mandate any particular reaction from
>>> the receiver in this situation.
>>>
>>>
>>>
>>>
>>>> Regards, Christian
>>>>
>>>> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
>>>>>> -----Original Message-----
>>>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of
>>>>>> Christian Groves
>>>>>> Sent: Thursday, October 09, 2014 7:30 AM
>>>>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
>>>>>> Subject: Re: [rtcweb] Question on
>>>>>> draft-ietf-rtcweb-stun-consent-freshness
>>>>>>
>>>>>> Hello Ram,
>>>>>>
>>>>>> Thanks for the response. A further question. The draft focuses on
>>>>>> sender behaviour with regards to consent, i.e. it indicates that an
>>>>>> endpoint must not send data if consent hasn't been granted. What
>>>>>> about receiver behaviour?
>>>>>> E.g. what happens if the sending end point is not well behaved and
>>>>>> sends application data beyond the consent period, what should the
>>>>>> receiver do ?
>>>>> The receiver can use DTLS fatal error to immediately revoke the
>>>>> consent. But if sender does not care about consent revocation then
>>>>> the receiver has to rely on other techniques (like closing the FW
>>>>> mapping etc.).  The javascript may be malicious, hence consent
>>>>> checks are done by the browser (which serves as user's TCB)) and
>>>>> browser has to be trusted to do the right thing.
>>>>>
>>>>> -Tiru
>>>>>
>>>>>> Regards, Christian
>>>>>>
>>>>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
>>>>>>> Hi Christian,
>>>>>>>
>>>>>>> Thanks for your comments. Please see inline
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Christian Groves <Christian.Groves@nteczone.com>
>>>>>>> Date: Wednesday, 8 October 2014 9:21 am
>>>>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>>>>>> Subject: [rtcweb] Question on
>>>>>>> draft-ietf-rtcweb-stun-consent-freshness
>>>>>>>
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> I have a couple of comments on
>>>>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>>>>>>>
>>>>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to both
>>>>>>>> "ICE binding requests" and "STUN binding requests". Is there a
>>>>>>>> distinction between them? e.g. "ICE binding requests" refer to
>>>>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
>>>>>>>> indicates a non-extended binding request as per RFC5389.
>>>>>>> We will change it to STUN binding requests to keep it consistent
>>>>>>> with the rest of the document.
>>>>>>>
>>>>>>>> 2) Just to confirm when Consent is first given. Is the 30sec
>>>>>>>> consent period assumed after the initial ICE exchange?
>>>>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>>>>>>>
>>>>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
>>>>>>>        expires after 30 seconds."
>>>>>>>
>>>>>>>
>>>>>>>> I found the second
>>>>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent
>>>>>>>> consent before the initial consent in paragraph 5. I'd suggest
>>>>>>>> modifying it along the lines of:
>>>>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP,
>>>>>>>> SCTP, DTLS), over any transport protocol (e.g., UDP, TCP) on an
>>>>>>>> ICE-initiated connection unless the receiving endpoint consents
>>>>>>>> to receive
>>>>>> the data.
>>>>>>>> Initial consent is granted as a result of a successful ICE
>>>>>>>> connectivity check on a particular transport address. Consent
>>>>>>>> expires after a fixed amount of time. As a result, subsequent
>>>>>>>> consent MUST be obtained following the procedure described in this
>> document.
>>>>>>>> During ICE restart consent checks MUST continue to be sent on
>>>>>>>> previously validated pair, and MUST be responded to on the
>>>>>>>> previously validated pair, until ICE restart completes."
>>>>>>> Sure will make this change.
>>>>>>>
>>>>>>> Regards,
>>>>>>> Ram
>>>>>>>
>>>>>>>> Regards, Christian
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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 nobody Thu Oct  9 05:18:48 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christian Groves <Christian.Groves@nteczone.com>, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Date: Thu, 9 Oct 2014 12:18:39 +0000
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References: <5434B4D9.8000302@nteczone.com> <D05AD834.1500B%rmohanr@cisco.com> <5435EC15.3040908@nteczone.com> <913383AAA69FF945B8F946018B75898A2833830C@xmb-rcd-x10.cisco.com> <54363020.2040807@nteczone.com> <543632A7.6050401@alvestrand.no> <54363775.40108@nteczone.com> <913383AAA69FF945B8F946018B75898A28338614@xmb-rcd-x10.cisco.com> <54365A2B.9080007@nteczone.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/z11nvAMIBP2o_qEhkjyOv_uLGC0
Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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Hi,

Why would the receiver have to do anything? Each endpoint uses consent chec=
ks to verify that it can SEND media.

Also, keep in mind that if the remote peer is ICE lite, it won't send any c=
onsent checks.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian Groves
Sent: 9. lokakuuta 2014 12:50
To: Tirumaleswar Reddy (tireddy); rtcweb@ietf.org
Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness

Hello Tiru,

I've seen that text but it talks about the receiving endpoint of a STUN bin=
ding response. I was referring to the endpoint that receives the STUN bindi=
ng request and sends the response. The draft concentrates on the STUN bindi=
ng request initiator and its notion of consent. It doesn't explain what not=
ion of consent that the receiver of the STUN binding request. E.g. a STUN b=
inding request receiver may close the connection which leads to an immediat=
e revokation but it doesn't mean it has the notion of consent. As Harald in=
dicates below the receiver may not care at all about it.

Regards, Christian

On 9/10/2014 7:29 PM, Tirumaleswar Reddy (tireddy) wrote:
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian=20
>> Groves
>> Sent: Thursday, October 09, 2014 12:51 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Question on=20
>> draft-ietf-rtcweb-stun-consent-freshness
>>
>> I don't think it should mandate a particular action but I think it=20
>> would be good to mention some of the possibilities. As it stands=20
>> nothing is mentioned in the draft about the role of the receiver.
> The role of receiver to immediately revoke consent either using STUN=20
> error code or DTLS fatal alert is discussed in
> http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-07
> #section-4.2
>
> -Tiru
>
>> If the receiver watches the 30 second timer, how does it distinguish=20
>> between receiving a STUN binding request for the purposes of Consent=20
>> (i.e. linked to application data signalling) and a STUN binding=20
>> request sent for the purposes of a keep alive? I guess that it uses=20
>> the fact that consent related STUN binding request are authenticated=20
>> and keepalive STUN binding requests are not (as per cl.10/RFC5245. Is th=
at correct?
>>
>> Regards, Christian
>>
>> On 9/10/2014 6:00 PM, Harald Alvestrand wrote:
>>> On 10/09/2014 08:50 AM, Christian Groves wrote:
>>>> Hello Tiru,
>>>>
>>>> So does this mean that the receiver also has to keep track of 30=20
>>>> second consent period to be able to identify the situation?
>>> If the receiver cares about being flooded without its consent, the=20
>>> receiver needs to detect that situation and take appropriate action.
>>> Either by watching the 30-second timer or by some other means.
>>>
>>> If the receiver doesn't care, it doesn't have to do anything.
>>>
>>> I don't think the spec needs to mandate any particular reaction from=20
>>> the receiver in this situation.
>>>
>>>
>>>
>>>
>>>> Regards, Christian
>>>>
>>>> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
>>>>>> -----Original Message-----
>>>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
>>>>>> Christian Groves
>>>>>> Sent: Thursday, October 09, 2014 7:30 AM
>>>>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
>>>>>> Subject: Re: [rtcweb] Question on=20
>>>>>> draft-ietf-rtcweb-stun-consent-freshness
>>>>>>
>>>>>> Hello Ram,
>>>>>>
>>>>>> Thanks for the response. A further question. The draft focuses on=20
>>>>>> sender behaviour with regards to consent, i.e. it indicates that=20
>>>>>> an endpoint must not send data if consent hasn't been granted.=20
>>>>>> What about receiver behaviour?
>>>>>> E.g. what happens if the sending end point is not well behaved=20
>>>>>> and sends application data beyond the consent period, what should=20
>>>>>> the receiver do ?
>>>>> The receiver can use DTLS fatal error to immediately revoke the=20
>>>>> consent. But if sender does not care about consent revocation then=20
>>>>> the receiver has to rely on other techniques (like closing the FW=20
>>>>> mapping etc.).  The javascript may be malicious, hence consent=20
>>>>> checks are done by the browser (which serves as user's TCB)) and=20
>>>>> browser has to be trusted to do the right thing.
>>>>>
>>>>> -Tiru
>>>>>
>>>>>> Regards, Christian
>>>>>>
>>>>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
>>>>>>> Hi Christian,
>>>>>>>
>>>>>>> Thanks for your comments. Please see inline
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Christian Groves <Christian.Groves@nteczone.com>
>>>>>>> Date: Wednesday, 8 October 2014 9:21 am
>>>>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>>>>>> Subject: [rtcweb] Question on
>>>>>>> draft-ietf-rtcweb-stun-consent-freshness
>>>>>>>
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> I have a couple of comments on
>>>>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
>>>>>>>>
>>>>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to=20
>>>>>>>> both "ICE binding requests" and "STUN binding requests". Is=20
>>>>>>>> there a distinction between them? e.g. "ICE binding requests"=20
>>>>>>>> refer to
>>>>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request=20
>>>>>>>> indicates a non-extended binding request as per RFC5389.
>>>>>>> We will change it to STUN binding requests to keep it consistent=20
>>>>>>> with the rest of the document.
>>>>>>>
>>>>>>>> 2) Just to confirm when Consent is first given. Is the 30sec=20
>>>>>>>> consent period assumed after the initial ICE exchange?
>>>>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient.
>>>>>>>
>>>>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
>>>>>>>        expires after 30 seconds."
>>>>>>>
>>>>>>>
>>>>>>>> I found the second
>>>>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent=20
>>>>>>>> consent before the initial consent in paragraph 5. I'd suggest=20
>>>>>>>> modifying it along the lines of:
>>>>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP,=20
>>>>>>>> SCTP, DTLS), over any transport protocol (e.g., UDP, TCP) on an=20
>>>>>>>> ICE-initiated connection unless the receiving endpoint consents=20
>>>>>>>> to receive
>>>>>> the data.
>>>>>>>> Initial consent is granted as a result of a successful ICE=20
>>>>>>>> connectivity check on a particular transport address. Consent=20
>>>>>>>> expires after a fixed amount of time. As a result, subsequent=20
>>>>>>>> consent MUST be obtained following the procedure described in=20
>>>>>>>> this
>> document.
>>>>>>>> During ICE restart consent checks MUST continue to be sent on=20
>>>>>>>> previously validated pair, and MUST be responded to on the=20
>>>>>>>> previously validated pair, until ICE restart completes."
>>>>>>> Sure will make this change.
>>>>>>>
>>>>>>> Regards,
>>>>>>> Ram
>>>>>>>
>>>>>>>> Regards, Christian
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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

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


From nobody Thu Oct  9 06:36:18 2014
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Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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On 10/09/2014 11:49 AM, Christian Groves wrote:
> Hello Tiru,
>
> I've seen that text but it talks about the receiving endpoint of a 
> STUN binding response. I was referring to the endpoint that receives 
> the STUN binding request and sends the response.

Aha - I was thinking that the "receiver" was the media receiver (the one 
who cares whether he's being DOSed or not), you were talking about the 
receiver of the STUN binding request.

> The draft concentrates on the STUN binding request initiator and its 
> notion of consent. It doesn't explain what notion of consent that the 
> receiver of the STUN binding request.
I think that's simple.

The receiver of the STUN binding request has no business trying to 
figure out why it got a binding request. Its business is to answer it.

It's the initiator who knows why the request was sent.


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From: Cullen Jennings <fluffy@iii.ca>
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Subject: [rtcweb] Dates for rtcweb drafts
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Once again, the chairs are updating dates for documents. I'm forming a =
list of when we hope to have an RFC for various drafts. In many ways the =
relative ordering of when these drafts is probably more accurate than =
the actually dates. The straw man for dates is

I-D.ietf-rtcweb-alpn  2015 Feb=20

I-D.ietf-rtcweb-audio 2015 May

I-D.ietf-rtcweb-constraints-registry 2015 Jan=20

I-D.ietf-rtcweb-data-channel  2014 Dec

I-D.ietf-rtcweb-data-protocol 2014 Dec

I-D.ietf-rtcweb-jsep  2015 May=20

I-D.ietf-rtcweb-overview 2015 May=20

I-D.ietf-rtcweb-rtp-usage 2015 Jan

I-D.ietf-rtcweb-security-arch 2015 Dec=20

I-D.ietf-rtcweb-security 2015 Dec=20

I-D.ietf-rtcweb-stun-consent-freshness 2015 Feb

I-D.ietf-rtcweb-transports 2015 Jan

I-D.ietf-rtcweb-video 2015 May

If you are an author or involved with these drafts or just have a strong =
opinion on what the date to be an RFC will be and want to use a =
different date than the ones above, let me know.

Thanks, Cullen

If you are interested, I have a document where I track the dates of most =
the normative dependencies for WebRTC at=20

https://tools.ietf.org/html/draft-jennings-rtcweb-deps



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On 9 October 2014 12:08, Cullen Jennings <fluffy@iii.ca> wrote:
> I-D.ietf-rtcweb-jsep  2015 May

https://www.youtube.com/watch?v=dik_wnOE4dk&index=2&t=33s&list=WL

This is a really large document and a good proportion of it hasn't
been reviewed or discussed.


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--047d7b86e1ec080a03050503269e
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On Thu, Oct 9, 2014 at 1:00 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 9 October 2014 12:08, Cullen Jennings <fluffy@iii.ca> wrote:
> > I-D.ietf-rtcweb-jsep  2015 May
>
> https://www.youtube.com/watch?v=dik_wnOE4dk&index=2&t=33s&list=WL
>
> This is a really large document and a good proportion of it hasn't
> been reviewed or discussed


... or written.

-Ekr


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

--047d7b86e1ec080a03050503269e
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Oct 9, 2014 at 1:00 PM, Martin Thomson <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@=
gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On 9 Oct=
ober 2014 12:08, Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca">fluff=
y@iii.ca</a>&gt; wrote:<br>
&gt; I-D.ietf-rtcweb-jsep=C2=A0 2015 May<br>
<br>
<a href=3D"https://www.youtube.com/watch?v=3Ddik_wnOE4dk&amp;index=3D2&amp;=
t=3D33s&amp;list=3DWL" target=3D"_blank">https://www.youtube.com/watch?v=3D=
dik_wnOE4dk&amp;index=3D2&amp;t=3D33s&amp;list=3DWL</a><br>
<br>
This is a really large document and a good proportion of it hasn&#39;t<br>
been reviewed or discussed</blockquote><div><br></div><div>... or written.<=
/div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><div 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></div>

--047d7b86e1ec080a03050503269e--


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On 9 October 2014 13:18, Eric Rescorla <ekr@rtfm.com> wrote:
>> > I-D.ietf-rtcweb-jsep  2015 May
>>
>> https://www.youtube.com/watch?v=dik_wnOE4dk&index=2&t=33s&list=WL
>>
>> This is a really large document and a good proportion of it hasn't
>> been reviewed or discussed
>
>
> ... or written.

I don't know, it looks like you've been making good progress.  You
might be half-way.

The "Parsing an Offer", "Parsing an Answer", "Applying a Local
Description" and "Applying a Remote Description" sections might be
empty now, but they aren't that hard.


From nobody Thu Oct  9 17:48:38 2014
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Subject: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'WebRTC Data Channels'
  <draft-ietf-rtcweb-data-channel-12.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-10-23. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebRTC framework specifies protocol support for direct
   interactive rich communication using audio, video, and data between
   two peers' web-browsers.  This document specifies the non-media data
   transport aspects of the WebRTC framework.  It provides an
   architectural overview of how the Stream Control Transmission
   Protocol (SCTP) is used in the WebRTC context as a generic transport
   service allowing WEB-browsers to exchange generic data from peer to
   peer.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/ballot/


No IPR declarations have been submitted directly on this I-D.



From nobody Thu Oct  9 21:03:26 2014
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References: <5434B4D9.8000302@nteczone.com> <D05AD834.1500B%rmohanr@cisco.com> <5435EC15.3040908@nteczone.com> <913383AAA69FF945B8F946018B75898A2833830C@xmb-rcd-x10.cisco.com> <54363020.2040807@nteczone.com> <543632A7.6050401@alvestrand.no> <54363775.40108@nteczone.com> <913383AAA69FF945B8F946018B75898A28338614@xmb-rcd-x10.cisco.com> <54365A2B.9080007@nteczone.com> <7594FB04B1934943A5C02806D1A2204B1D47093C@ESESSMB209.ericsson.se>
Date: Fri, 10 Oct 2014 09:33:09 +0530
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness
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On Thu, Oct 9, 2014 at 5:48 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> Why would the receiver have to do anything? Each endpoint uses consent
> checks to verify that it can SEND media.
>
> Also, keep in mind that if the remote peer is ICE lite, it won't send any
> consent checks.
>

=E2=80=8BGiven the security issue with ICE lite, I am wondering whether we =
should
deprecate it going forward. It was originally conceived as an initial step
towards a full ICE implementation, but I think we didn't have WebRTC then :=
)

Muthu


>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
> Groves
> Sent: 9. lokakuuta 2014 12:50
> To: Tirumaleswar Reddy (tireddy); rtcweb@ietf.org
> Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshnes=
s
>
> Hello Tiru,
>
> I've seen that text but it talks about the receiving endpoint of a STUN
> binding response. I was referring to the endpoint that receives the STUN
> binding request and sends the response. The draft concentrates on the STU=
N
> binding request initiator and its notion of consent. It doesn't explain
> what notion of consent that the receiver of the STUN binding request. E.g=
.
> a STUN binding request receiver may close the connection which leads to a=
n
> immediate revokation but it doesn't mean it has the notion of consent. As
> Harald indicates below the receiver may not care at all about it.
>
> Regards, Christian
>
> On 9/10/2014 7:29 PM, Tirumaleswar Reddy (tireddy) wrote:
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian
> >> Groves
> >> Sent: Thursday, October 09, 2014 12:51 PM
> >> To: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Question on
> >> draft-ietf-rtcweb-stun-consent-freshness
> >>
> >> I don't think it should mandate a particular action but I think it
> >> would be good to mention some of the possibilities. As it stands
> >> nothing is mentioned in the draft about the role of the receiver.
> > The role of receiver to immediately revoke consent either using STUN
> > error code or DTLS fatal alert is discussed in
> > http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-07
> > #section-4.2
> >
> > -Tiru
> >
> >> If the receiver watches the 30 second timer, how does it distinguish
> >> between receiving a STUN binding request for the purposes of Consent
> >> (i.e. linked to application data signalling) and a STUN binding
> >> request sent for the purposes of a keep alive? I guess that it uses
> >> the fact that consent related STUN binding request are authenticated
> >> and keepalive STUN binding requests are not (as per cl.10/RFC5245. Is
> that correct?
> >>
> >> Regards, Christian
> >>
> >> On 9/10/2014 6:00 PM, Harald Alvestrand wrote:
> >>> On 10/09/2014 08:50 AM, Christian Groves wrote:
> >>>> Hello Tiru,
> >>>>
> >>>> So does this mean that the receiver also has to keep track of 30
> >>>> second consent period to be able to identify the situation?
> >>> If the receiver cares about being flooded without its consent, the
> >>> receiver needs to detect that situation and take appropriate action.
> >>> Either by watching the 30-second timer or by some other means.
> >>>
> >>> If the receiver doesn't care, it doesn't have to do anything.
> >>>
> >>> I don't think the spec needs to mandate any particular reaction from
> >>> the receiver in this situation.
> >>>
> >>>
> >>>
> >>>
> >>>> Regards, Christian
> >>>>
> >>>> On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:
> >>>>>> -----Original Message-----
> >>>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> >>>>>> Christian Groves
> >>>>>> Sent: Thursday, October 09, 2014 7:30 AM
> >>>>>> To: Ram Mohan R (rmohanr); rtcweb@ietf.org
> >>>>>> Subject: Re: [rtcweb] Question on
> >>>>>> draft-ietf-rtcweb-stun-consent-freshness
> >>>>>>
> >>>>>> Hello Ram,
> >>>>>>
> >>>>>> Thanks for the response. A further question. The draft focuses on
> >>>>>> sender behaviour with regards to consent, i.e. it indicates that
> >>>>>> an endpoint must not send data if consent hasn't been granted.
> >>>>>> What about receiver behaviour?
> >>>>>> E.g. what happens if the sending end point is not well behaved
> >>>>>> and sends application data beyond the consent period, what should
> >>>>>> the receiver do ?
> >>>>> The receiver can use DTLS fatal error to immediately revoke the
> >>>>> consent. But if sender does not care about consent revocation then
> >>>>> the receiver has to rely on other techniques (like closing the FW
> >>>>> mapping etc.).  The javascript may be malicious, hence consent
> >>>>> checks are done by the browser (which serves as user's TCB)) and
> >>>>> browser has to be trusted to do the right thing.
> >>>>>
> >>>>> -Tiru
> >>>>>
> >>>>>> Regards, Christian
> >>>>>>
> >>>>>> On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:
> >>>>>>> Hi Christian,
> >>>>>>>
> >>>>>>> Thanks for your comments. Please see inline
> >>>>>>>
> >>>>>>> -----Original Message-----
> >>>>>>> From: Christian Groves <Christian.Groves@nteczone.com>
> >>>>>>> Date: Wednesday, 8 October 2014 9:21 am
> >>>>>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> >>>>>>> Subject: [rtcweb] Question on
> >>>>>>> draft-ietf-rtcweb-stun-consent-freshness
> >>>>>>>
> >>>>>>>> Hello,
> >>>>>>>>
> >>>>>>>> I have a couple of comments on
> >>>>>>>> draft-ietf-rtcweb-stun-consent-freshness-07:
> >>>>>>>>
> >>>>>>>> 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-freshness refers to
> >>>>>>>> both "ICE binding requests" and "STUN binding requests". Is
> >>>>>>>> there a distinction between them? e.g. "ICE binding requests"
> >>>>>>>> refer to
> >>>>>>>> 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and STUN binding request
> >>>>>>>> indicates a non-extended binding request as per RFC5389.
> >>>>>>> We will change it to STUN binding requests to keep it consistent
> >>>>>>> with the rest of the document.
> >>>>>>>
> >>>>>>>> 2) Just to confirm when Consent is first given. Is the 30sec
> >>>>>>>> consent period assumed after the initial ICE exchange?
> >>>>>>> Paragraph 4 in Section 4.1 has this text. Hope this is sufficient=
.
> >>>>>>>
> >>>>>>> "The initial Consent to send traffic is obtained by ICE.  Consent
> >>>>>>>        expires after 30 seconds."
> >>>>>>>
> >>>>>>>
> >>>>>>>> I found the second
> >>>>>>>> paragraph in cl.4.1 not so clear as it talks about subsequent
> >>>>>>>> consent before the initial consent in paragraph 5. I'd suggest
> >>>>>>>> modifying it along the lines of:
> >>>>>>>> "An endpoint MUST NOT send application data (e.g., RTP, RTCP,
> >>>>>>>> SCTP, DTLS), over any transport protocol (e.g., UDP, TCP) on an
> >>>>>>>> ICE-initiated connection unless the receiving endpoint consents
> >>>>>>>> to receive
> >>>>>> the data.
> >>>>>>>> Initial consent is granted as a result of a successful ICE
> >>>>>>>> connectivity check on a particular transport address. Consent
> >>>>>>>> expires after a fixed amount of time. As a result, subsequent
> >>>>>>>> consent MUST be obtained following the procedure described in
> >>>>>>>> this
> >> document.
> >>>>>>>> During ICE restart consent checks MUST continue to be sent on
> >>>>>>>> previously validated pair, and MUST be responded to on the
> >>>>>>>> previously validated pair, until ICE restart completes."
> >>>>>>> Sure will make this change.
> >>>>>>>
> >>>>>>> Regards,
> >>>>>>> Ram
> >>>>>>>
> >>>>>>>> Regards, Christian
> >>>>>>>>
> >>>>>>>> _______________________________________________
> >>>>>>>> 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
>
> _______________________________________________
> 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
>

--f46d042184598d908d050509a02f
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small"><span style=3D"font-family:arial">On Th=
u, Oct 9, 2014 at 5:48 PM, Christer Holmberg </span><span dir=3D"ltr" style=
=3D"font-family:arial">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com=
" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span><span styl=
e=3D"font-family:arial"> wrote:</span><br></div><div class=3D"gmail_extra">=
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
Why would the receiver have to do anything? Each endpoint uses consent chec=
ks to verify that it can SEND media.<br>
<br>
Also, keep in mind that if the remote peer is ICE lite, it won&#39;t send a=
ny consent checks.<br></blockquote><div><br></div><div><div class=3D"gmail_=
default" style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=
=E2=80=8BGiven the security issue with ICE lite, I am wondering whether we =
should deprecate it going forward. It was originally conceived as an initia=
l step towards a full ICE implementation, but I think we didn&#39;t have We=
bRTC then :)</div><div class=3D"gmail_default" style=3D"font-family:arial,h=
elvetica,sans-serif;font-size:small"><br></div><div class=3D"gmail_default"=
 style=3D"font-family:arial,helvetica,sans-serif;font-size:small">Muthu</di=
v></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Regards,<br>
<br>
Christer<br>
<span class=3D"im HOEnZb"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-boun=
ces@ietf.org</a>] On Behalf Of Christian Groves<br>
</span><span class=3D"im HOEnZb">Sent: 9. lokakuuta 2014 12:50<br>
To: Tirumaleswar Reddy (tireddy); <a href=3D"mailto:rtcweb@ietf.org">rtcweb=
@ietf.org</a><br>
Subject: Re: [rtcweb] Question on draft-ietf-rtcweb-stun-consent-freshness<=
br>
<br>
</span><div class=3D"HOEnZb"><div class=3D"h5">Hello Tiru,<br>
<br>
I&#39;ve seen that text but it talks about the receiving endpoint of a STUN=
 binding response. I was referring to the endpoint that receives the STUN b=
inding request and sends the response. The draft concentrates on the STUN b=
inding request initiator and its notion of consent. It doesn&#39;t explain =
what notion of consent that the receiver of the STUN binding request. E.g. =
a STUN binding request receiver may close the connection which leads to an =
immediate revokation but it doesn&#39;t mean it has the notion of consent. =
As Harald indicates below the receiver may not care at all about it.<br>
<br>
Regards, Christian<br>
<br>
On 9/10/2014 7:29 PM, Tirumaleswar Reddy (tireddy) wrote:<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rt=
cweb-bounces@ietf.org</a>] On Behalf Of Christian<br>
&gt;&gt; Groves<br>
&gt;&gt; Sent: Thursday, October 09, 2014 12:51 PM<br>
&gt;&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; Subject: Re: [rtcweb] Question on<br>
&gt;&gt; draft-ietf-rtcweb-stun-consent-freshness<br>
&gt;&gt;<br>
&gt;&gt; I don&#39;t think it should mandate a particular action but I thin=
k it<br>
&gt;&gt; would be good to mention some of the possibilities. As it stands<b=
r>
&gt;&gt; nothing is mentioned in the draft about the role of the receiver.<=
br>
&gt; The role of receiver to immediately revoke consent either using STUN<b=
r>
&gt; error code or DTLS fatal alert is discussed in<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-f=
reshness-07" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb=
-stun-consent-freshness-07</a><br>
&gt; #section-4.2<br>
&gt;<br>
&gt; -Tiru<br>
&gt;<br>
&gt;&gt; If the receiver watches the 30 second timer, how does it distingui=
sh<br>
&gt;&gt; between receiving a STUN binding request for the purposes of Conse=
nt<br>
&gt;&gt; (i.e. linked to application data signalling) and a STUN binding<br=
>
&gt;&gt; request sent for the purposes of a keep alive? I guess that it use=
s<br>
&gt;&gt; the fact that consent related STUN binding request are authenticat=
ed<br>
&gt;&gt; and keepalive STUN binding requests are not (as per cl.10/RFC5245.=
 Is that correct?<br>
&gt;&gt;<br>
&gt;&gt; Regards, Christian<br>
&gt;&gt;<br>
&gt;&gt; On 9/10/2014 6:00 PM, Harald Alvestrand wrote:<br>
&gt;&gt;&gt; On 10/09/2014 08:50 AM, Christian Groves wrote:<br>
&gt;&gt;&gt;&gt; Hello Tiru,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; So does this mean that the receiver also has to keep track=
 of 30<br>
&gt;&gt;&gt;&gt; second consent period to be able to identify the situation=
?<br>
&gt;&gt;&gt; If the receiver cares about being flooded without its consent,=
 the<br>
&gt;&gt;&gt; receiver needs to detect that situation and take appropriate a=
ction.<br>
&gt;&gt;&gt; Either by watching the 30-second timer or by some other means.=
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If the receiver doesn&#39;t care, it doesn&#39;t have to do an=
ything.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I don&#39;t think the spec needs to mandate any particular rea=
ction from<br>
&gt;&gt;&gt; the receiver in this situation.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Regards, Christian<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On 9/10/2014 5:42 PM, Tirumaleswar Reddy (tireddy) wrote:<=
br>
&gt;&gt;&gt;&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt;&gt;&gt;&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-boun=
ces@ietf.org">rtcweb-bounces@ietf.org</a>] On Behalf Of<br>
&gt;&gt;&gt;&gt;&gt;&gt; Christian Groves<br>
&gt;&gt;&gt;&gt;&gt;&gt; Sent: Thursday, October 09, 2014 7:30 AM<br>
&gt;&gt;&gt;&gt;&gt;&gt; To: Ram Mohan R (rmohanr); <a href=3D"mailto:rtcwe=
b@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt;&gt;&gt;&gt; Subject: Re: [rtcweb] Question on<br>
&gt;&gt;&gt;&gt;&gt;&gt; draft-ietf-rtcweb-stun-consent-freshness<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Hello Ram,<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Thanks for the response. A further question. The d=
raft focuses on<br>
&gt;&gt;&gt;&gt;&gt;&gt; sender behaviour with regards to consent, i.e. it =
indicates that<br>
&gt;&gt;&gt;&gt;&gt;&gt; an endpoint must not send data if consent hasn&#39=
;t been granted.<br>
&gt;&gt;&gt;&gt;&gt;&gt; What about receiver behaviour?<br>
&gt;&gt;&gt;&gt;&gt;&gt; E.g. what happens if the sending end point is not =
well behaved<br>
&gt;&gt;&gt;&gt;&gt;&gt; and sends application data beyond the consent peri=
od, what should<br>
&gt;&gt;&gt;&gt;&gt;&gt; the receiver do ?<br>
&gt;&gt;&gt;&gt;&gt; The receiver can use DTLS fatal error to immediately r=
evoke the<br>
&gt;&gt;&gt;&gt;&gt; consent. But if sender does not care about consent rev=
ocation then<br>
&gt;&gt;&gt;&gt;&gt; the receiver has to rely on other techniques (like clo=
sing the FW<br>
&gt;&gt;&gt;&gt;&gt; mapping etc.).=C2=A0 The javascript may be malicious, =
hence consent<br>
&gt;&gt;&gt;&gt;&gt; checks are done by the browser (which serves as user&#=
39;s TCB)) and<br>
&gt;&gt;&gt;&gt;&gt; browser has to be trusted to do the right thing.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; -Tiru<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Regards, Christian<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; On 8/10/2014 5:41 PM, Ram Mohan R (rmohanr) wrote:=
<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Hi Christian,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Thanks for your comments. Please see inline<br=
>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; From: Christian Groves &lt;<a href=3D"mailto:C=
hristian.Groves@nteczone.com">Christian.Groves@nteczone.com</a>&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Date: Wednesday, 8 October 2014 9:21 am<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; To: &quot;<a href=3D"mailto:rtcweb@ietf.org">r=
tcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a>&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Subject: [rtcweb] Question on<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; draft-ietf-rtcweb-stun-consent-freshness<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Hello,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I have a couple of comments on<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; draft-ietf-rtcweb-stun-consent-freshness-0=
7:<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; 1) Cl.4.1/draft-ietf-rtcweb-stun-consent-f=
reshness refers to<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; both &quot;ICE binding requests&quot; and =
&quot;STUN binding requests&quot;. Is<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; there a distinction between them? e.g. &qu=
ot;ICE binding requests&quot;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; refer to<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; 7.1.2/draft-ietf-mmusic-rfc5245bis-02 and =
STUN binding request<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; indicates a non-extended binding request a=
s per RFC5389.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; We will change it to STUN binding requests to =
keep it consistent<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; with the rest of the document.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; 2) Just to confirm when Consent is first g=
iven. Is the 30sec<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; consent period assumed after the initial I=
CE exchange?<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Paragraph 4 in Section 4.1 has this text. Hope=
 this is sufficient.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; &quot;The initial Consent to send traffic is o=
btained by ICE.=C2=A0 Consent<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 expires after 30 se=
conds.&quot;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I found the second<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; paragraph in cl.4.1 not so clear as it tal=
ks about subsequent<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; consent before the initial consent in para=
graph 5. I&#39;d suggest<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; modifying it along the lines of:<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &quot;An endpoint MUST NOT send applicatio=
n data (e.g., RTP, RTCP,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; SCTP, DTLS), over any transport protocol (=
e.g., UDP, TCP) on an<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; ICE-initiated connection unless the receiv=
ing endpoint consents<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; to receive<br>
&gt;&gt;&gt;&gt;&gt;&gt; the data.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Initial consent is granted as a result of =
a successful ICE<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; connectivity check on a particular transpo=
rt address. Consent<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; expires after a fixed amount of time. As a=
 result, subsequent<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; consent MUST be obtained following the pro=
cedure described in<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; this<br>
&gt;&gt; document.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; During ICE restart consent checks MUST con=
tinue to be sent on<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; previously validated pair, and MUST be res=
ponded to on the<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; previously validated pair, until ICE resta=
rt completes.&quot;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Sure will make this change.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Ram<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Regards, Christian<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; __________________________________________=
_____<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@=
ietf.org</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/li=
stinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcw=
eb</a><br>
&gt;&gt;&gt;&gt;&gt;&gt; _______________________________________________<br=
>
&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><b=
r>
&gt;&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--f46d042184598d908d050509a02f--


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On 9 Oct 2014, at 20:08, Cullen Jennings <fluffy@iii.ca> wrote:
> Once again, the chairs are updating dates for documents. I'm forming a =
list of when we hope to have an RFC for various drafts. In many ways the =
relative ordering of when these drafts is probably more accurate than =
the actually dates. The straw man for dates is
=85
> I-D.ietf-rtcweb-rtp-usage 2015 Jan

The only open issue with this draft is FEC support. Varun recently =
submitted draft-singh-payload-rtp-1d2d-parity-scheme-00 as a possible =
FEC scheme for WebRTC. Feedback from the working group on whether that =
draft is suitable to be referenced from the rtp-usage draft would be =
helpful.

--=20
Colin Perkins
https://csperkins.org/





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I'd like the chairs to indicate whether or not they are going to 
entertain -gateways for WG draft status too.



On 10/09/2014 09:08 PM, Cullen Jennings wrote:
> Once again, the chairs are updating dates for documents. I'm forming a list of when we hope to have an RFC for various drafts. In many ways the relative ordering of when these drafts is probably more accurate than the actually dates. The straw man for dates is
>
> I-D.ietf-rtcweb-alpn  2015 Feb
>
> I-D.ietf-rtcweb-audio 2015 May
>
> I-D.ietf-rtcweb-constraints-registry 2015 Jan
>
> I-D.ietf-rtcweb-data-channel  2014 Dec
>
> I-D.ietf-rtcweb-data-protocol 2014 Dec
>
> I-D.ietf-rtcweb-jsep  2015 May
>
> I-D.ietf-rtcweb-overview 2015 May
>
> I-D.ietf-rtcweb-rtp-usage 2015 Jan
>
> I-D.ietf-rtcweb-security-arch 2015 Dec
>
> I-D.ietf-rtcweb-security 2015 Dec
I assume "2014 Dec" is the intended target date here?
>
> I-D.ietf-rtcweb-stun-consent-freshness 2015 Feb
>
> I-D.ietf-rtcweb-transports 2015 Jan
>
> I-D.ietf-rtcweb-video 2015 May
>
> If you are an author or involved with these drafts or just have a strong opinion on what the date to be an RFC will be and want to use a different date than the ones above, let me know.
>
> Thanks, Cullen
>
> If you are interested, I have a document where I track the dates of most the normative dependencies for WebRTC at
>
> https://tools.ietf.org/html/draft-jennings-rtcweb-deps
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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Hi,

I have some questions on the connection liveness/heartbeat section of the d=
raft.


Q1:
=3D=3D=3D

The text says:

        "Similarly, if packets haven't been received within a certain perio=
d,
        an application can request a consent check (heartbeat) be generated=
."

BUT, if an endpoint is anyway sending consent requests every 5th second, wh=
y aren't the those enough as hearbeats? As long as the endpoint receives re=
sponses, it knows the remote endpoint is alive (the text even says that an =
endpoint is considered alive as long as some data is received from it).

And, if the endpoint does NOT receive responses to the consent requests, co=
nsent will expire anyway.

I DO understand that an ICE lite may want to send a heartbeat request if it=
 does not receive any data, as it doesn't send consent requests.


Q2:
=3D=3D=3D

The text says:


        "Sending consent checks (heartbeats) at a high rate could allow a

        malicious application to generate congestion, so applications MUST

        NOT be able to send heartbeats at an average rate of more than 1 pe=
r

        second."



I don't think applications should be alowed to generate heartbeats this oft=
en, and I see no need for it.


Q3:
=3D=3D=3D

The spec is unclear on what happens is a response to a heartbeat request is=
 not received. I guess that is an application decision, but I think the spe=
c should be clear that it does NOT have any impact on consent - the "real" =
consent requests are used to determine.


Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">I have some questions on the connection liveness/hea=
rtbeat section of the draft.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Q1:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The text says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;Similarl=
y, if packets haven't been received within a certain period,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; an application can r=
equest a consent check (heartbeat) be generated.&#8221;<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">BUT, if an endpoint is anyway sending consent requests eve=
ry 5<sup>th</sup> second, why aren&#8217;t the those enough as hearbeats? A=
s long as the endpoint receives responses, it knows
 the remote endpoint is alive (the text even says that an endpoint is consi=
dered alive as long as some data is received from it).
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">And, if the endpoint does NOT receive responses to the con=
sent requests, consent will expire anyway.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">I DO understand that an ICE lite may want to send a heartb=
eat request if it does not receive any data, as it doesn&#8217;t send conse=
nt requests.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Q2:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">The text says:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;Sending consent chec=
ks (heartbeats) at a high rate could allow a<o:p></o:p></pre>
<pre>&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; malicious application to generat=
e congestion, so applications MUST<o:p></o:p></pre>
<pre>&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; NOT be able to send heartbeats a=
t an average rate of more than 1 per<o:p></o:p></pre>
<pre>&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; second.&#8221;<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>I don&#8217;t think applications should be alowed to generate heartbea=
ts this often, and I see no need for it.<o:p></o:p></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Q3:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">The spec is unclear on what happens is a response to a hea=
rtbeat request is not received. I guess that is an application decision, bu=
t I think the spec should be clear that it does
 NOT have any impact on consent &#8211; the &#8220;real&#8221; consent requ=
ests are used to determine.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

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From: <stephane.proust@orange.com>
To: "'rtcweb@ietf.org'" <rtcweb@ietf.org>
Thread-Topic: Work progress on draft-ietf-rtcweb-audio-codecs-for-interop
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Date: Fri, 10 Oct 2014 14:25:21 +0000
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Subject: [rtcweb] Work progress on draft-ietf-rtcweb-audio-codecs-for-interop
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Dear all,

This document (http://tools.ietf.org/id/draft-proust-rtcweb-audio-codecs-fo=
r-interop-00.txt) being now a shared working document, I would be happy to =
get the views of the Group on how to jointly progress and improve it.

In order to trigger this discussion, I would propose the following modifica=
tions with 2 main purposes:
1) Remove any reference to "legacy" (and the corresponding definition secti=
on n=B03): Even with the definition in section 3, this word "legacy" remain=
s unclear and not really needed for the understanding of this draft: there =
are no reasons to limit the possible WebRTC interoperability use cases to "=
legacy" interop assuming we know clearly what "legacy" covers: any interope=
rability could be relevant if justified by real and significant use cases.
2) Emphasize and detail for AMR and AMR-WB some concrete points related to =
codec offer that are important for interoperability (which is indeed the co=
re motivation of this draft)

As Editor of the document, I would be happy to consider any other additiona=
l text you may propose and include it in the document if agreeable by the G=
roup

Best regards

St=E9phane

---------------------------------------------------------------------------=
----------------
Proposed changes:

Modif 1
Title
[Additional WebRTC audio codecs for interoperability with legacy networks. ]
->=20
[Additional WebRTC audio codecs for interoperability. ]

Modif 2
[3.  Definitions - Legacy networks... ] -> removed

Modif 3
[4.  Rationale for additional WebRTC codecs
  ...The WebRTC technology is however expected to have more extended usage
  to communicate with  other types of clients.  It can be used for instance=
 as an
  Access  technology to 3GPP IMS services or to interoperate with fixed or
   mobile VoIP legacy HD voice service...]
->
[4.  Rationale for additional WebRTC codecs
   ...The WebRTC technology is however expected to be used to communicate w=
ith
   other types of clients using other technologies. It can be used for inst=
ance
   as an access technology to 3GPP IMS services (e.g. VoLTE, ViLTE) or to i=
nteroperate with fixed or
   mobile Circuit Switched or VoIP services like mobile 3GPP 3G/2G Circuit
   Switched voice or DECT based VoIP telephony...]

Modif 4
Text added to section [5.1.3 Guidelines for AMR-WB usage and implementation=
 with WebRTC]
[ In order to maximize the possibility of successful call establishment for=
 WebRTC client offering AMR-WB it is important that the WebRTC client:=20
-   Offer AMR in addition to AMR-WB with AMR-WB, being a wideband codec,=20
     listed first as preferred payload type with respect to other narrow ba=
nd
    codecs (AMR, G.711 .) and with Bandwidth Efficient payload format prefe=
rred.=09
-  Be capable of operating AMR-WB with any subset of the nine codec modes a=
nd
    source controlled rate operation.
- Offer at least one AMR-WB configuration with parameter settings as defined
   in Table 6.1 and in Table 6.2 of [TS26.114].In order to maximize the
   interoperability and quality this offer do not restrict the codec modes
   offered. Restrictions in the use of codec modes may be included in the a=
nswer ]

Modif 5
Text added to section [5.2.3.  Guidelines for AMR usage and implementation =
with WebRTC]
[In order to maximize the possibility of successful call establishment for =
WebRTC
  client offering AMR, it is important that the WebRTC client:=20
-  Be capable of operating AMR with any subset of the eight codec modes and
   source controlled rate operation.
- Offer at least one configuration with parameter settings as defined in
   Table 6.1 and in Table 6.2 of [TS26.114].In order to maximize the
   interoperability and quality this offer shall not restrict AMR codec mod=
es
   offered. Restrictions in the use of codec modes may be included in the a=
nswer.

 For convenience, I attach a version in Word format with changes as revisio=
n marks



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On 10 October 2014 03:52, Harald Alvestrand <harald@alvestrand.no> wrote:
> I'd like the chairs to indicate whether or not they are going to entertain
> -gateways for WG draft status too.

Maybe you should instead ask the working group.


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On 10/10/2014 06:35 PM, Martin Thomson wrote:
> On 10 October 2014 03:52, Harald Alvestrand <harald@alvestrand.no> wrote:
>> I'd like the chairs to indicate whether or not they are going to entertain
>> -gateways for WG draft status too.
> Maybe you should instead ask the working group.

That's the question I think it's the WG chairs' job to ask.


-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Oct 10 10:13:41 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Dates for rtcweb drafts
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While you try to figure out who should ask the question, I'll give MY answe=
r: I think the draft should be adopted as a WG item :)

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestran=
d
Sent: 10 October 2014 19:59
To: Martin Thomson
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Dates for rtcweb drafts

On 10/10/2014 06:35 PM, Martin Thomson wrote:
> On 10 October 2014 03:52, Harald Alvestrand <harald@alvestrand.no> wrote:
>> I'd like the chairs to indicate whether or not they are going to=20
>> entertain -gateways for WG draft status too.
> Maybe you should instead ask the working group.

That's the question I think it's the WG chairs' job to ask.


--
Surveillance is pervasive. Go Dark.

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


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Fri, 10 Oct 2014 10:38:46 -0700
To: Christer Holmberg <christer.holmberg@ericsson.com>
Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/unME9ug4buA6_rOHBzrOsO7UKgA
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Dates for rtcweb drafts
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+1.



> On Oct 10, 2014, at 10:13, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:
>=20
> While you try to figure out who should ask the question, I'll give MY answ=
er: I think the draft should be adopted as a WG item :)
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestra=
nd
> Sent: 10 October 2014 19:59
> To: Martin Thomson
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Dates for rtcweb drafts
>=20
>> On 10/10/2014 06:35 PM, Martin Thomson wrote:
>>> On 10 October 2014 03:52, Harald Alvestrand <harald@alvestrand.no> wrote=
:
>>> I'd like the chairs to indicate whether or not they are going to=20
>>> entertain -gateways for WG draft status too.
>> Maybe you should instead ask the working group.
>=20
> That's the question I think it's the WG chairs' job to ask.
>=20
>=20
> --
> Surveillance is pervasive. Go Dark.
>=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: "Wyss, Felix" <Felix.Wyss@inin.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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I'm sorry for being a bit late to the party, but I have a concern about sec=
tion 6.5 allowing use of the DTLS role for picking the SCTP stream identifi=
ers (odd vs. even).   It's an abstraction leak and complicates the implemen=
tation of middleboxes.  The DTLS role is determined during the connection e=
stablishment, so a middlebox that manages connections with two endpoints to=
 record the media passing through it can't guarantee that one side is the D=
TLS server and the other the DTLS client because in its offers it is requir=
ed to specify a=3Dsetup:actpass (see RFC#5763).  That means it's at the mer=
cy of the endpoint as to which DTLS role they pick.  Worse, it is desirable=
 for the answerers to pick 'active' as that can speed up the DTLS negotiati=
on. =20

I feel it would be better to explicitly require that applications are respo=
nsible for identifier collision avoidance instead of allowing them to rely =
on the DTLS roles.=20

Thanks,
--Felix Wyss
Interactive Intelligence, Inc.

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of The IESG
> Sent: Thursday, October 09, 2014 20:49
> To: IETF-Announce
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (Web=
RTC
> Data Channels) to Proposed Standard
>=20
>=20
> The IESG has received a request from the Real-Time Communication in WEB-
> browsers WG (rtcweb) to consider the following document:
> - 'WebRTC Data Channels'
>   <draft-ietf-rtcweb-data-channel-12.txt> as Proposed Standard
>=20
> The IESG plans to make a decision in the next few weeks, and solicits fin=
al
> comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2014-10-23. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the beginnin=
g of
> the Subject line to allow automated sorting.
>=20
> Abstract
>=20
>=20
>    The WebRTC framework specifies protocol support for direct
>    interactive rich communication using audio, video, and data between
>    two peers' web-browsers.  This document specifies the non-media data
>    transport aspects of the WebRTC framework.  It provides an
>    architectural overview of how the Stream Control Transmission
>    Protocol (SCTP) is used in the WebRTC context as a generic transport
>    service allowing WEB-browsers to exchange generic data from peer to
>    peer.
>=20
>=20
>=20
>=20
> The file can be obtained via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/
>=20
> IESG discussion can be tracked via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/ballot/
>=20
>=20
> No IPR declarations have been submitted directly on this I-D.
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Fri Oct 10 13:33:53 2014
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On 10 October 2014 12:11, Wyss, Felix <Felix.Wyss@inin.com> wrote:
> I feel it would be better to explicitly require that applications are responsible for identifier collision avoidance instead of allowing them to rely on the DTLS roles.

Are you suggesting that we might want to consider the effect of a MitM
attack on the robustness of this?


From nobody Fri Oct 10 13:52:47 2014
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From: Suhas Nandakumar <suhasietf@gmail.com>
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Subject: [rtcweb] New Draft: draft-nandakumar-mmusic-proto-iana-registration
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--f46d043bd8eeef8735050517ba3c
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Hello All

Here is the link to MMUSIC SDP draft that registers new proto attributes
for RTP/TCP for TLS and SAVP(F) variants.

http://tools.ietf.org/html/draft-nandakumar-mmusic-proto-iana-registration-00


This plan is move all these variants into a MMUSIC draft instead of
defining it in the JSEP going further


Look forward for the feedback

Cheers
Suhas

--f46d043bd8eeef8735050517ba3c
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<div dir=3D"ltr">Hello All<div><br></div><div>Here is the link to MMUSIC SD=
P draft that registers new proto attributes for RTP/TCP for TLS and SAVP(F)=
 variants.</div><div><br></div><div><a href=3D"http://tools.ietf.org/html/d=
raft-nandakumar-mmusic-proto-iana-registration-00">http://tools.ietf.org/ht=
ml/draft-nandakumar-mmusic-proto-iana-registration-00</a><br></div><div><br=
></div><div><br></div><div>This plan is move all these variants into a MMUS=
IC draft instead of defining it in the JSEP going further</div><div><br></d=
iv><div><br></div><div>Look forward for the feedback</div><div><br></div><d=
iv>Cheers</div><div>Suhas</div></div>

--f46d043bd8eeef8735050517ba3c--


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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'WebRTC Data Channel Establishment Protocol'
  <draft-ietf-rtcweb-data-protocol-08.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-10-24. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebRTC framework specifies protocol support for direct
   interactive rich communication using audio, video, and data between
   two peers' web-browsers.  This document specifies a simple protocol
   for establishing symmetric Data Channels between the peers.  It uses
   a two way handshake and allows sending of user data without waiting
   for the handshake to complete.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/ballot/


No IPR declarations have been submitted directly on this I-D.



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From nobody Sat Oct 11 00:41:44 2014
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On 11 Oct 2014, at 09:21, Wyss, Felix <Felix.Wyss@inin.com> wrote:

> I'm not looking at this in terms of MitM attacks but for topologies =
where we want to have back-to-back WebRTC sessions going through a =
legitimate middlebox.  That middlebox passes the media and data between =
these sessions while recording and/or performing analytics on it.  =
Requiring the=20
So doesn't it terminate the DTLS connections of both peers?

Best regards
Michael
> middlebox to worry about how the two endpoints might pick SCTP stream =
identifiers and possibly having to map or rewrite them unnecessarily =
complicates their implementation. =20
>=20
> IMHO from the perspective of the data channels, the DTLS layer should =
be an opaque tunnel ("bump in the stack") that passes packets between =
endpoints.  The DTLS role of an endpoint that emerges during connection =
establishment should not be relevant to the operation of the data =
channels.  Hence my concerns about the abstraction leak. =20
>=20
> Thanks,
> --Felix
>=20
>> -----Original Message-----
>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>> Sent: Friday, October 10, 2014 16:34
>> To: Wyss, Felix
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Last Call: =
<draft-ietf-rtcweb-data-channel-12.txt>
>> (WebRTC Data Channels) to Proposed Standard
>>=20
>> On 10 October 2014 12:11, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>>> I feel it would be better to explicitly require that applications =
are
>> responsible for identifier collision avoidance instead of allowing =
them to rely
>> on the DTLS roles.
>>=20
>> Are you suggesting that we might want to consider the effect of a =
MitM
>> attack on the robustness of this?
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 2014-09-28 23:49, Michael Tuexen wrote:
> On 16 Aug 2014, at 08:44, Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> wrote:
>
>> We have discussed the lack of description of failure handling in draft-ietf-rtcweb-data-channel a couple of times.
>>
>> Explanations of mechanisms have been provided in mail answers that I have not seen documented elsewhere, but they have not been introduced in the draft.
>>
>> I think that for specification and implementation of the WebRTC API, there is a need for clear specification how the channels behave in failure situations.
>>
>> I suggest to insert a new chapter 6.7 in the draft with the following contents.
>>
>> -------------------------------------------------------------------Proposed new section 6.7-------------------------------------------------------------------------------------------
>> 6.7 Transmission failure and error handling
>>
>> Failures can occur in an open channel by transmission error detection and by SCTP watchdog error indications.
>> If transmission in a reliable channel fails, the association where the channel belongs will be aborted. If all retries for a watchdog expires, the corresponding association will be aborted.  All channels in an association that is aborted because of errors need to be closed by the party detecting the failure. Network error indications shall be provided to upper layers on all closed channels.
>>
>> If retransmissions or transmission time is exceeded in a partially reliable channel, the transmission SHALL be allowed to continue with other data items. A transmission error indication SHALL be provided to upper layers on that channel.
>>
>> If reception out-of order is detected from an SCTP channel for which ordered transmission is requested, the receiver SHALL close the data channel, and provide an transmission error indication to upper layers.
>>
>> If a message with an unsupported PPID is received or some logical error is detected by the receiver, the receiver SHALL close the corresponding data channel.
>> -----------------------------------------------------------------------end of proposal---------------------------------------------------------------------------------------------------------------
>>
>> I am not sure about what the two last kinds of errors should result in and hope that the experts can adjust the wording.
> Hi Gunnar,
>
> I think I have covered the contents your are suggesting, but I added some of the text in a section called SCTP Association Management.
>
> Please let me know if the new revision addresses your comments.
Yes, I think that in version -12, there is sufficient indication of the 
behavior in failure situations.

Thanks,
/Gunnar
>
> Best regards
> Michael
>> Regards
>>
>> Gunnar
>> Gunnar Hellström
>> Omnitor
>> gunnar.hellstrom@omnitor.se
>>
>> On 2014-08-11 20:13, IESG Secretary wrote:
>>> The IESG would like to retract the current Last Call on the following
>>> document:
>>>
>>> - 'WebRTC Data Channels'
>>>    <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>>
>>> The document's state has been set back to "AD Evaluation."
>>>
>>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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On 11 Oct 2014, at 08:41, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:

> On 11 Oct 2014, at 09:21, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>=20
>> I'm not looking at this in terms of MitM attacks but for topologies =
where we want to have back-to-back WebRTC sessions going through a =
legitimate middlebox.  That middlebox passes the media and data between =
these sessions while recording and/or performing analytics on it.  =
Requiring the=20
> So doesn't it terminate the DTLS connections of both peers?

Yes, it does. I think the worry is that the middle box may end up as =
DTLS server to both legs.
That leaves both endpoints as DTLS clients. This then breaks the =
odd-even split and risks the endpoints allocating
the same stream numbers.

For this to happen you have to have a middle box that offers actpass to =
both endpoints, agrees to passive=20
on both legs and then proceeds to terminate the DTLS but bridge the =
SCTP. Which only works if it also
re-writes the signalling fingerprints =91correctly' too.

Tim.


>=20
> Best regards
> Michael
>> middlebox to worry about how the two endpoints might pick SCTP stream =
identifiers and possibly having to map or rewrite them unnecessarily =
complicates their implementation. =20
>>=20
>> IMHO from the perspective of the data channels, the DTLS layer should =
be an opaque tunnel ("bump in the stack") that passes packets between =
endpoints.  The DTLS role of an endpoint that emerges during connection =
establishment should not be relevant to the operation of the data =
channels.  Hence my concerns about the abstraction leak. =20
>>=20
>> Thanks,
>> --Felix
>>=20
>>> -----Original Message-----
>>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>>> Sent: Friday, October 10, 2014 16:34
>>> To: Wyss, Felix
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Last Call: =
<draft-ietf-rtcweb-data-channel-12.txt>
>>> (WebRTC Data Channels) to Proposed Standard
>>>=20
>>> On 10 October 2014 12:11, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>>>> I feel it would be better to explicitly require that applications =
are
>>> responsible for identifier collision avoidance instead of allowing =
them to rely
>>> on the DTLS roles.
>>>=20
>>> Are you suggesting that we might want to consider the effect of a =
MitM
>>> attack on the robustness of this?
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On 11 Oct 2014, at 08:41, Michael =
Tuexen &lt;<a =
href=3D"mailto:Michael.Tuexen@lurchi.franken.de">Michael.Tuexen@lurchi.fra=
nken.de</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;">On 11 Oct 2014, at 09:21, Wyss, Felix =
&lt;<a href=3D"mailto:Felix.Wyss@inin.com">Felix.Wyss@inin.com</a>&gt; =
wrote:<br><br><blockquote type=3D"cite">I'm not looking at this in terms =
of MitM attacks but for topologies where we want to have back-to-back =
WebRTC sessions going through a legitimate middlebox. &nbsp;That =
middlebox passes the media and data between these sessions while =
recording and/or performing analytics on it. &nbsp;Requiring the<span =
class=3D"Apple-converted-space">&nbsp;</span><br></blockquote>So doesn't =
it terminate the DTLS connections of both =
peers?<br></div></blockquote><div><br></div><div>Yes, it does. I think =
the worry is that the middle box may end up as DTLS server to both =
legs.</div><div>That leaves both endpoints as DTLS clients. This then =
breaks the odd-even split and risks the endpoints =
allocating</div><div>the same stream =
numbers.</div><div><br></div><div>For this to happen you have to have a =
middle box that offers actpass to both endpoints, agrees to =
passive&nbsp;</div><div>on both legs and then proceeds to terminate the =
DTLS but bridge the SCTP. Which only works if it =
also</div><div>re-writes the signalling fingerprints =91correctly' =
too.</div><div><br></div><div>Tim.</div><div><br></div><br><blockquote =
type=3D"cite"><div style=3D"font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><br>Best =
regards<br>Michael<br><blockquote type=3D"cite">middlebox to worry about =
how the two endpoints might pick SCTP stream identifiers and possibly =
having to map or rewrite them unnecessarily complicates their =
implementation. &nbsp;<br><br>IMHO from the perspective of the data =
channels, the DTLS layer should be an opaque tunnel ("bump in the =
stack") that passes packets between endpoints. &nbsp;The DTLS role of an =
endpoint that emerges during connection establishment should not be =
relevant to the operation of the data channels. &nbsp;Hence my concerns =
about the abstraction leak. =
&nbsp;<br><br>Thanks,<br>--Felix<br><br><blockquote =
type=3D"cite">-----Original Message-----<br>From: Martin Thomson [<a =
href=3D"mailto:martin.thomson@gmail.com">mailto:martin.thomson@gmail.com</=
a>]<br>Sent: Friday, October 10, 2014 16:34<br>To: Wyss, Felix<br>Cc: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>Subject: Re: =
[rtcweb] Last Call: =
&lt;draft-ietf-rtcweb-data-channel-12.txt&gt;<br>(WebRTC Data Channels) =
to Proposed Standard<br><br>On 10 October 2014 12:11, Wyss, Felix &lt;<a =
href=3D"mailto:Felix.Wyss@inin.com">Felix.Wyss@inin.com</a>&gt; =
wrote:<br><blockquote type=3D"cite">I feel it would be better to =
explicitly require that applications are<br></blockquote>responsible for =
identifier collision avoidance instead of allowing them to rely<br>on =
the DTLS roles.<br><br>Are you suggesting that we might want to consider =
the effect of a MitM<br>attack on the robustness of =
this?<br></blockquote>_______________________________________________<br>r=
tcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br><br></blockquote><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">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></div></blockquote></div><br></body></html>=

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On 11 Oct 2014, at 11:59, tim panton <tim@phonefromhere.com> wrote:

>=20
> On 11 Oct 2014, at 08:41, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>=20
>> On 11 Oct 2014, at 09:21, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>>=20
>>> I'm not looking at this in terms of MitM attacks but for topologies =
where we want to have back-to-back WebRTC sessions going through a =
legitimate middlebox.  That middlebox passes the media and data between =
these sessions while recording and/or performing analytics on it.  =
Requiring the=20
>> So doesn't it terminate the DTLS connections of both peers?
>=20
> Yes, it does. I think the worry is that the middle box may end up as =
DTLS server to both legs.
And why can't the middle box avoid this?
> That leaves both endpoints as DTLS clients. This then breaks the =
odd-even split and risks the endpoints allocating
> the same stream numbers.
I think "risks" is the wrong term here. I guess they will run into this =
almost always.

Best regards
Michael
>=20
> For this to happen you have to have a middle box that offers actpass =
to both endpoints, agrees to passive=20
> on both legs and then proceeds to terminate the DTLS but bridge the =
SCTP. Which only works if it also
> re-writes the signalling fingerprints =91correctly' too.
>=20
> Tim.
>=20
>=20
>>=20
>> Best regards
>> Michael
>>> middlebox to worry about how the two endpoints might pick SCTP =
stream identifiers and possibly having to map or rewrite them =
unnecessarily complicates their implementation. =20
>>>=20
>>> IMHO from the perspective of the data channels, the DTLS layer =
should be an opaque tunnel ("bump in the stack") that passes packets =
between endpoints.  The DTLS role of an endpoint that emerges during =
connection establishment should not be relevant to the operation of the =
data channels.  Hence my concerns about the abstraction leak. =20
>>>=20
>>> Thanks,
>>> --Felix
>>>=20
>>>> -----Original Message-----
>>>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>>>> Sent: Friday, October 10, 2014 16:34
>>>> To: Wyss, Felix
>>>> Cc: rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] Last Call: =
<draft-ietf-rtcweb-data-channel-12.txt>
>>>> (WebRTC Data Channels) to Proposed Standard
>>>>=20
>>>> On 10 October 2014 12:11, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>>>>> I feel it would be better to explicitly require that applications =
are
>>>> responsible for identifier collision avoidance instead of allowing =
them to rely
>>>> on the DTLS roles.
>>>>=20
>>>> Are you suggesting that we might want to consider the effect of a =
MitM
>>>> attack on the robustness of this?
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, tim panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> > Yes, it does. I think the worry is that the middle box may end up as DT=
LS server to both legs.
> And why can't the middle box avoid this?

How can it if it is the offerer on both legs?  From RFC#5763, section 5 (pa=
ge 10): "The endpoint that is the offerer MUST use the setup attribute valu=
e of setup:actpass and be prepared to receive a client_hello before it rece=
ives the answer."

The answerer thus gets to pick whether it is the client or server.  As ment=
ioned in my original message, the answerer picking client is desirable as i=
t can speed up the DTLS negotiation.  So almost invariably the middlebox wi=
ll be DTLS server on both legs.


> > That leaves both endpoints as DTLS clients. This then breaks the
> > odd-even split and risks the endpoints allocating the same stream numbe=
rs.
> I think "risks" is the wrong term here. I guess they will run into this a=
lmost always.

Not necessarily if the endpoints draw their IDs at random.  That makes this=
 particularly problematic as everything may appear to work fine but then fa=
il intermittently in the field due to collisions. =20

--Felix


From nobody Sat Oct 11 10:25:18 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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On 11 Oct 2014, at 18:58, Wyss, Felix <Felix.Wyss@inin.com> wrote:

>>> Yes, it does. I think the worry is that the middle box may end up as =
DTLS server to both legs.
>> And why can't the middle box avoid this?
>=20
> How can it if it is the offerer on both legs?  =46rom RFC#5763, =
section 5 (page 10): "The endpoint that is the offerer MUST use the =
setup attribute value of setup:actpass and be prepared to receive a =
client_hello before it receives the answer."
OK, I see.
>=20
> The answerer thus gets to pick whether it is the client or server.  As =
mentioned in my original message, the answerer picking client is =
desirable as it can speed up the DTLS negotiation.  So almost invariably =
the middlebox will be DTLS server on both legs.
Makes sense.
>=20
>=20
>>> That leaves both endpoints as DTLS clients. This then breaks the
>>> odd-even split and risks the endpoints allocating the same stream =
numbers.
>> I think "risks" is the wrong term here. I guess they will run into =
this almost always.
>=20
> Not necessarily if the endpoints draw their IDs at random.  That makes =
this particularly problematic as everything may appear to work fine but =
then fail intermittently in the field due to collisions. =20
As far as I know the browsers currently don't use random values. They =
start from small IDs and increment.
Since your middlebox is terminating DTLS, I think you could also =
terminate both SCTP associations. But I guess
you want to avoid that. So can't you just rewrite the stream IDs?

Best regards
Michael
>=20
> --Felix
>=20
>=20


From nobody Sat Oct 11 10:30:24 2014
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On 11 Oct 2014, at 19:25, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:

> On 11 Oct 2014, at 18:58, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>=20
>>>> Yes, it does. I think the worry is that the middle box may end up =
as DTLS server to both legs.
>>> And why can't the middle box avoid this?
>>=20
>> How can it if it is the offerer on both legs?  =46rom RFC#5763, =
section 5 (page 10): "The endpoint that is the offerer MUST use the =
setup attribute value of setup:actpass and be prepared to receive a =
client_hello before it receives the answer."
> OK, I see.
>>=20
>> The answerer thus gets to pick whether it is the client or server.  =
As mentioned in my original message, the answerer picking client is =
desirable as it can speed up the DTLS negotiation.  So almost invariably =
the middlebox will be DTLS server on both legs.
> Makes sense.
>>=20
>>=20
>>>> That leaves both endpoints as DTLS clients. This then breaks the
>>>> odd-even split and risks the endpoints allocating the same stream =
numbers.
>>> I think "risks" is the wrong term here. I guess they will run into =
this almost always.
>>=20
>> Not necessarily if the endpoints draw their IDs at random.  That =
makes this particularly problematic as everything may appear to work =
fine but then fail intermittently in the field due to collisions. =20
> As far as I know the browsers currently don't use random values. They =
start from small IDs and increment.
> Since your middlebox is terminating DTLS, I think you could also =
terminate both SCTP associations. But I guess
> you want to avoid that. So can't you just rewrite the stream IDs?
My fault: You can't. Both sides might derive something from the ID =
locally seen and assume it
is the same on the other end...

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


From nobody Sun Oct 12 03:18:32 2014
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Given that the middle box has to:

- decrypt and encrypt all packets (because the keys are different)
- assert its own identity towards both endpoints (because otherwise DTLS
doesn't work)
- be trusted by both endpoints (otherwise it's a MITM attack by definition)
- manage SCTP congestion control independently on the two legs (because
that's how SCTP works)

I don't see the reshuffling of packets between channel numbers to be an
excessive *additional* burden.

The kind of "middle box" that "just shuffles bits" while doing something
to them or with them in addition doesn't exist in the WebRTC context.


On 10/11/2014 06:58 PM, Wyss, Felix wrote:
>>> Yes, it does. I think the worry is that the middle box may end up as DTLS server to both legs.
>> And why can't the middle box avoid this?
> How can it if it is the offerer on both legs?  From RFC#5763, section 5 (page 10): "The endpoint that is the offerer MUST use the setup attribute value of setup:actpass and be prepared to receive a client_hello before it receives the answer."
>
> The answerer thus gets to pick whether it is the client or server.  As mentioned in my original message, the answerer picking client is desirable as it can speed up the DTLS negotiation.  So almost invariably the middlebox will be DTLS server on both legs.
>
>
>>> That leaves both endpoints as DTLS clients. This then breaks the
>>> odd-even split and risks the endpoints allocating the same stream numbers.
>> I think "risks" is the wrong term here. I guess they will run into this almost always.
> Not necessarily if the endpoints draw their IDs at random.  That makes this particularly problematic as everything may appear to work fine but then fail intermittently in the field due to collisions.  
>
> --Felix
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


From nobody Sun Oct 12 06:23:09 2014
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> Given that the middle box has to:
>=20
> - decrypt and encrypt all packets (because the keys are different)
> - assert its own identity towards both endpoints (because otherwise DTLS
> doesn't work)
> - be trusted by both endpoints (otherwise it's a MITM attack by definitio=
n)
> - manage SCTP congestion control independently on the two legs (because
> that's how SCTP works)
>=20
> I don't see the reshuffling of packets between channel numbers to be an
> excessive *additional* burden.
>=20
> The kind of "middle box" that "just shuffles bits" while doing something
> to them or with them in addition doesn't exist in the WebRTC context.
[Raju]=20
+1.
Also, such a middle box can even go with Out-of-band negotiation (instead o=
f inband DCEP with odd/even rule)=20
of stream ids, in which case stream ids are selected by the application and=
 odd/even rule does not apply.


BR
Raju


From nobody Sun Oct 12 06:30:52 2014
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On 12 Oct 2014, at 12:18, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> Given that the middle box has to:
>=20
> - decrypt and encrypt all packets (because the keys are different)
> - assert its own identity towards both endpoints (because otherwise =
DTLS
> doesn't work)
> - be trusted by both endpoints (otherwise it's a MITM attack by =
definition)
> - manage SCTP congestion control independently on the two legs =
(because
> that's how SCTP works)
>=20
> I don't see the reshuffling of packets between channel numbers to be =
an
> excessive *additional* burden.
I agree CPU wise. But I have a question: Don't expect both endpoints of =
a
data channel to see the same "id" attribute? And isn't this attribute
the stream ID? So can conceptually a middlebox map the streams IDs?

Best regards
Michael=20
>=20
> The kind of "middle box" that "just shuffles bits" while doing =
something
> to them or with them in addition doesn't exist in the WebRTC context.
>=20
>=20
> On 10/11/2014 06:58 PM, Wyss, Felix wrote:
>>>> Yes, it does. I think the worry is that the middle box may end up =
as DTLS server to both legs.
>>> And why can't the middle box avoid this?
>> How can it if it is the offerer on both legs?  =46rom RFC#5763, =
section 5 (page 10): "The endpoint that is the offerer MUST use the =
setup attribute value of setup:actpass and be prepared to receive a =
client_hello before it receives the answer."
>>=20
>> The answerer thus gets to pick whether it is the client or server.  =
As mentioned in my original message, the answerer picking client is =
desirable as it can speed up the DTLS negotiation.  So almost invariably =
the middlebox will be DTLS server on both legs.
>>=20
>>=20
>>>> That leaves both endpoints as DTLS clients. This then breaks the
>>>> odd-even split and risks the endpoints allocating the same stream =
numbers.
>>> I think "risks" is the wrong term here. I guess they will run into =
this almost always.
>> Not necessarily if the endpoints draw their IDs at random.  That =
makes this particularly problematic as everything may appear to work =
fine but then fail intermittently in the field due to collisions. =20
>>=20
>> --Felix
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> Given that the middle box has to:
>=20
> - decrypt and encrypt all packets (because the keys are different)
> - assert its own identity towards both endpoints (because otherwise DTLS =
doesn't work)
> - be trusted by both endpoints (otherwise it's a MITM attack by definitio=
n)
All these are not relevant with respect to how the SCTP packets get from on=
e WebRTC endpoint to the other.  That's really all the data channels care a=
bout.  The rest is opaque to them, assumed a given and the concern of other=
 layers in the WebRTC stack.

> - manage SCTP congestion control independently on the two legs (because t=
hat's how SCTP works)
Can you please elaborate on why you believe that's required?  As I understa=
nd it, from SCTP's perspective in WebRTC DTLS represents the IP layer.  Doe=
sn't that mean congestion control is strictly above DTLS and SCTP should no=
t care about how many hops its packets go through and whether any of these =
hops decrypt/re-encrypt the packets or pass them on as-is?=20

> I don't see the reshuffling of packets between channel numbers to be an e=
xcessive *additional* burden.
I respectfully disagree as it adds unnecessary complexity.  Michael further=
more has made a good argument on why it's not that simple.  To add to his c=
oncerns: are we certain that none of the application protocols using the da=
ta channels will ever reference stream IDs of one SCTP stream in the data o=
f another SCTP stream?  If that ever were the case, the middlebox would hav=
e to understand these application protocols and rewrite the stream data, no=
t just the SCTP packets.=20

I furthermore have still not seen a convincing argument for requiring this =
abstraction leak from the lower level by using the DTLS role for something =
unrelated to DTLS' purpose.  Whether an endpoint started as server or clien=
t is completely irrelevant to the payload data transported over it.  While =
it may be a seemingly convenient hack to avoid explicit conflict resolution=
, abstraction leaks like that invariably have unintended consequences--as w=
e can see here. =20

> The kind of "middle box" that "just shuffles bits" while doing something =
to them or with them in addition doesn't exist in the WebRTC context.
Well, then it should.  I presented a use-case of a middlebox that records a=
nd/or performs analytics of the WebRTC session media.  That is a common app=
lication in VoIP.=20

--Felix


=09


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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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On 10/12/2014 03:30 PM, Michael Tuexen wrote:
> On 12 Oct 2014, at 12:18, Harald Alvestrand <harald@alvestrand.no> wrote:
>
>> Given that the middle box has to:
>>
>> - decrypt and encrypt all packets (because the keys are different)
>> - assert its own identity towards both endpoints (because otherwise DTLS
>> doesn't work)
>> - be trusted by both endpoints (otherwise it's a MITM attack by definition)
>> - manage SCTP congestion control independently on the two legs (because
>> that's how SCTP works)
>>
>> I don't see the reshuffling of packets between channel numbers to be an
>> excessive *additional* burden.
> I agree CPU wise. But I have a question: Don't expect both endpoints of a
> data channel to see the same "id" attribute? And isn't this attribute
> the stream ID? So can conceptually a middlebox map the streams IDs?

The two ends don't see the same identity either, so the middle box is 
not transparent.

The application has to know the middle box is there. So it has to be 
written to deal with differing numbers at the two ends too.



From nobody Sun Oct 12 09:50:57 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: Harald Alvestrand <harald@alvestrand.no>
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On 12 Oct 2014, at 17:28, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 10/12/2014 03:30 PM, Michael Tuexen wrote:
>> On 12 Oct 2014, at 12:18, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>=20
>>> Given that the middle box has to:
>>>=20
>>> - decrypt and encrypt all packets (because the keys are different)
>>> - assert its own identity towards both endpoints (because otherwise =
DTLS
>>> doesn't work)
>>> - be trusted by both endpoints (otherwise it's a MITM attack by =
definition)
>>> - manage SCTP congestion control independently on the two legs =
(because
>>> that's how SCTP works)
>>>=20
>>> I don't see the reshuffling of packets between channel numbers to be =
an
>>> excessive *additional* burden.
>> I agree CPU wise. But I have a question: Don't expect both endpoints =
of a
>> data channel to see the same "id" attribute? And isn't this attribute
>> the stream ID? So can conceptually a middlebox map the streams IDs?
>=20
> The two ends don't see the same identity either, so the middle box is =
not transparent.
OK. So this means that the ID defined in=20
http://w3c.github.io/webrtc-pc/#widl-RTCDataChannel-id
doesn't have to be the same on both ends, right? It would be useful
to make that clear in the document. I always assumed that they are the =
same
on both ends...

Best regards
Michael
>=20
> The application has to know the middle box is there. So it has to be =
written to deal with differing numbers at the two ends too.
>=20
>=20
>=20


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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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On 10/12/2014 06:50 PM, Michael Tuexen wrote:
> On 12 Oct 2014, at 17:28, Harald Alvestrand <harald@alvestrand.no> wrote:
>
>> On 10/12/2014 03:30 PM, Michael Tuexen wrote:
>>> On 12 Oct 2014, at 12:18, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>
>>>> Given that the middle box has to:
>>>>
>>>> - decrypt and encrypt all packets (because the keys are different)
>>>> - assert its own identity towards both endpoints (because otherwise DTLS
>>>> doesn't work)
>>>> - be trusted by both endpoints (otherwise it's a MITM attack by definition)
>>>> - manage SCTP congestion control independently on the two legs (because
>>>> that's how SCTP works)
>>>>
>>>> I don't see the reshuffling of packets between channel numbers to be an
>>>> excessive *additional* burden.
>>> I agree CPU wise. But I have a question: Don't expect both endpoints of a
>>> data channel to see the same "id" attribute? And isn't this attribute
>>> the stream ID? So can conceptually a middlebox map the streams IDs?
>> The two ends don't see the same identity either, so the middle box is not transparent.
> OK. So this means that the ID defined in 
> http://w3c.github.io/webrtc-pc/#widl-RTCDataChannel-id
> doesn't have to be the same on both ends, right?

It has to be the same on both ends of the SCTP connection.

It's just that (I think) you can't splice SCTP connections together;
either you terminate them, or you don't.

That's my understanding - YMMV.

>  It would be useful
> to make that clear in the document. I always assumed that they are the same
> on both ends...
>
> Best regards
> Michael
>> The application has to know the middle box is there. So it has to be written to deal with differing numbers at the two ends too.
>>
>>
>>


-- 
Surveillance is pervasive. Go Dark.


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On 08/21/2014 09:36 PM, Martin Thomson wrote:
> On 21 August 2014 11:25, Roman Shpount <roman@telurix.com> wrote:
>> All entities receive peer transport information from elsewhere, including
>> gateways running ICE-Lite. Does it mean all of them need to perform consent?
> That's the logical conclusion, yes.

Careful with the words here.

Every browser that sends data/media needs to verify that the reciever
desires to continue to receive it. That means that it needs to initiate
STUN binding requests for consent verification.

Every entity, everywhere, needs to respond to those binding requests.

I'm not sure which of those roles the words "perform consent" is meant
to cover.


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


-- 
Surveillance is pervasive. Go Dark.


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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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On 12 Oct 2014, at 19:46, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 10/12/2014 06:50 PM, Michael Tuexen wrote:
>> On 12 Oct 2014, at 17:28, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>=20
>>> On 10/12/2014 03:30 PM, Michael Tuexen wrote:
>>>> On 12 Oct 2014, at 12:18, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>>>=20
>>>>> Given that the middle box has to:
>>>>>=20
>>>>> - decrypt and encrypt all packets (because the keys are different)
>>>>> - assert its own identity towards both endpoints (because =
otherwise DTLS
>>>>> doesn't work)
>>>>> - be trusted by both endpoints (otherwise it's a MITM attack by =
definition)
>>>>> - manage SCTP congestion control independently on the two legs =
(because
>>>>> that's how SCTP works)
>>>>>=20
>>>>> I don't see the reshuffling of packets between channel numbers to =
be an
>>>>> excessive *additional* burden.
>>>> I agree CPU wise. But I have a question: Don't expect both =
endpoints of a
>>>> data channel to see the same "id" attribute? And isn't this =
attribute
>>>> the stream ID? So can conceptually a middlebox map the streams IDs?
>>> The two ends don't see the same identity either, so the middle box =
is not transparent.
>> OK. So this means that the ID defined in=20
>> http://w3c.github.io/webrtc-pc/#widl-RTCDataChannel-id
>> doesn't have to be the same on both ends, right?
>=20
> It has to be the same on both ends of the SCTP connection.
This means you can't remap the streams. Or am I missing something?

Best regards
Michael
>=20
> It's just that (I think) you can't splice SCTP connections together;
> either you terminate them, or you don't.
>=20
> That's my understanding - YMMV.
>=20
>> It would be useful
>> to make that clear in the document. I always assumed that they are =
the same
>> on both ends...
>>=20
>> Best regards
>> Michael
>>> The application has to know the middle box is there. So it has to be =
written to deal with differing numbers at the two ends too.
>>>=20
>>>=20
>>>=20
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20
>=20


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> >> OK. So this means that the ID defined in
> >> http://w3c.github.io/webrtc-pc/#widl-RTCDataChannel-id
> >> doesn't have to be the same on both ends, right?
> >
> > It has to be the same on both ends of the SCTP connection.
> This means you can't remap the streams. Or am I missing something?
[Raju] When middle box is in the bearer path each end will see middle box a=
s its peer.
If the DCEP is used then it gets terminated at middle box and re-originated=
 towards=20
the other end after necessary stream id mapping.
One way to avoid such mapping is to use external negotiation of stream ids =
as=20
an alternative to inband DCEP.

BR
raju


From nobody Mon Oct 13 02:51:20 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Overview: Real Time Protocols for Browser-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-12.txt
	Pages           : 22
	Date            : 2014-10-13

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications WebRTC
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview-12

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-12


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

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


From nobody Mon Oct 13 10:03:20 2014
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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Michael Tuexen" <Michael.Tuexen@lurchi.franken.de>, Harald Alvestrand <harald@alvestrand.no>
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> > > It has to be the same on both ends of the SCTP connection.
> > This means you can't remap the streams. Or am I missing something?
> [Raju] When middle box is in the bearer path each end will see middle box
> as its peer.
> If the DCEP is used then it gets terminated at middle box and re-
> originated towards the other end after necessary stream id mapping.
> One way to avoid such mapping is to use external negotiation of stream id=
s
> as an alternative to inband DCEP.

They would only see it as a peer if it were a full B2B user agent.  As an e=
xample, in VoIP the endpoints do not consider an SBC as a peer--yet it sits=
 in the middle and may inspect the media passing through it. =20

With respect to DCEP: If we already define a protocol for establishing the =
channels in-band, why not make ID selection and conflict resolution an expl=
icit part of it?  That certainly would beat the IMHO rather distasteful DTL=
S role hack.

--Felix


From nobody Mon Oct 13 10:58:30 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "Wyss, Felix" <Felix.Wyss@inin.com>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Date: Mon, 13 Oct 2014 17:58:11 +0000
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References: <20141010004836.12666.88765.idtracker@ietfa.amsl.com> <91953101b2634ec69d14e120ea62d929@CY1PR0501MB1579.namprd05.prod.outlook.com> <CABkgnnWE7czWqi4vQRb5d50N2k95joc_Zcvw6w6g7SOjU9b+9g@mail.gmail.com> <03a3bbbc282b4e6d88d587931b46b5f8@CY1PR0501MB1579.namprd05.prod.outlook.com> <57B40110-2F21-4893-B77C-54F46D18567F@lurchi.franken.de> <1DE35922-E890-40C2-87E9-C8C12235920E@phonefromhere.com> <E53B9B7E-37F0-495C-B1BA-DE0A48AC6D73@lurchi.franken.de> <abdfd3ef58dd40028e8d5e2248b5a995@CY1PR0501MB1579.namprd05.prod.outlook.com> <543A5570.7060208@alvestrand.no> <B53499C4-6B51-4E8E-87C2-C8E5C10DBC34@lurchi.franken.de> <543A9E11.2030605@alvestrand.no> <F5C54F5E-C301-4EC7-9536-E43EA3A16863@lurchi.franken.de> <543ABE69.8030104@alvestrand.no> <99078EA5-5FE7-431F-9C70-EEA882F4C3C6@lurchi.franken.de> <E1FE4C082A89A246A11D7F32A95A17828E5DA2AC@US70UWXCHMBA02.zam.alcatel-lucent.com> <71e2b21516cb496eb4d1ad2b26e53a29@CY1PR0501MB1579.namprd05.prod.outlook.com>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> They would only see it as a peer if it were a full B2B user agent.  As an
> example, in VoIP the endpoints do not consider an SBC as a peer--yet it s=
its
> in the middle and may inspect the media passing through it.

<Raju>
I agree that from signaling point of view SBCs can be as transparent as pos=
sible.=20
But from bearer point of view, in majority of the cases SBCs are not transp=
arent and acts as a peer.
Especially when services like NAP/T traversal, transcoding, encryption (DTL=
S) are involved.
</Raju>

>=20
> With respect to DCEP: If we already define a protocol for establishing th=
e
> channels in-band, why not make ID selection and conflict resolution an
> explicit part of it?  That certainly would beat the IMHO rather distastef=
ul
> DTLS role hack.

<Raju>
Id selection is already part of the API.
odd/even rule applies only when createDataChannel () API does not provide a=
n id.
An application can always override this by providing its own id allocation =
scheme.
e.g.:
- Random id (2^16 unique ids). There is still a chance for glare.
- In case of glare (upon SCTP stream reset) app can try to generate another=
 id.

I am still not convinced of a middle box which terminates DTLS but neither
maps stream ids nor uses external data channel negotiation. IMHO, it has to
do either of those or clients need to have their own explicit management of=
 ids.

So, I am not sure of complicating the DCEP to add conflict resolution is wo=
rth it.

</Raju>

BR
Raju


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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 13 Oct 2014 12:27:06 -0700
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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As pointed out by others mentioning "external negotiation", the
application can use any mechanism it wants to choose the stream
identifiers (SIDs).  The even/odd choice is merely the default
behavior there for the convenience of simple applications.  More
advanced applications, such as the kind you are describing with a
middle box, can ignore the default even/odd behavior and choose the
SIDs directly.

On Fri, Oct 10, 2014 at 12:11 PM, Wyss, Felix <Felix.Wyss@inin.com> wrote:
> I'm sorry for being a bit late to the party, but I have a concern about s=
ection 6.5 allowing use of the DTLS role for picking the SCTP stream identi=
fiers (odd vs. even).   It's an abstraction leak and complicates the implem=
entation of middleboxes.  The DTLS role is determined during the connection=
 establishment, so a middlebox that manages connections with two endpoints =
to record the media passing through it can't guarantee that one side is the=
 DTLS server and the other the DTLS client because in its offers it is requ=
ired to specify a=3Dsetup:actpass (see RFC#5763).  That means it's at the m=
ercy of the endpoint as to which DTLS role they pick.  Worse, it is desirab=
le for the answerers to pick 'active' as that can speed up the DTLS negotia=
tion.
>
> I feel it would be better to explicitly require that applications are res=
ponsible for identifier collision avoidance instead of allowing them to rel=
y on the DTLS roles.
>
> Thanks,
> --Felix Wyss
> Interactive Intelligence, Inc.
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of The IESG
>> Sent: Thursday, October 09, 2014 20:49
>> To: IETF-Announce
>> Cc: rtcweb@ietf.org
>> Subject: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (We=
bRTC
>> Data Channels) to Proposed Standard
>>
>>
>> The IESG has received a request from the Real-Time Communication in WEB-
>> browsers WG (rtcweb) to consider the following document:
>> - 'WebRTC Data Channels'
>>   <draft-ietf-rtcweb-data-channel-12.txt> as Proposed Standard
>>
>> The IESG plans to make a decision in the next few weeks, and solicits fi=
nal
>> comments on this action. Please send substantive comments to the
>> ietf@ietf.org mailing lists by 2014-10-23. Exceptionally, comments may b=
e
>> sent to iesg@ietf.org instead. In either case, please retain the beginni=
ng of
>> the Subject line to allow automated sorting.
>>
>> Abstract
>>
>>
>>    The WebRTC framework specifies protocol support for direct
>>    interactive rich communication using audio, video, and data between
>>    two peers' web-browsers.  This document specifies the non-media data
>>    transport aspects of the WebRTC framework.  It provides an
>>    architectural overview of how the Stream Control Transmission
>>    Protocol (SCTP) is used in the WebRTC context as a generic transport
>>    service allowing WEB-browsers to exchange generic data from peer to
>>    peer.
>>
>>
>>
>>
>> The file can be obtained via
>> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/
>>
>> IESG discussion can be tracked via
>> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/ballot/
>>
>>
>> No IPR declarations have been submitted directly on this I-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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Subject: [rtcweb] Comments on draft-ietf-rtcweb-video Section 4.2 (H.264)
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--089e013c6c2416bca3050555d553
Content-Type: text/plain; charset=UTF-8

This section needs more work.  Aside from requiring support for
packetization mode 1, we could use additional text on robustness.

Could aspects of existing H.264 robustness recommendations (such 3GPP
S4-140961, designed for Video over LTE) could be incorporated?

      S4-140961

CR 26.114-0287 rev 1 Video telephony robustness improvements (Release 12)

Qualcomm Incorporated, Ericsson

10, 11



agreed

15.4

--089e013c6c2416bca3050555d553
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<div dir=3D"ltr"><div>This section needs more work.=C2=A0 Aside from requir=
ing support for packetization mode 1, we could use additional text on robus=
tness.=C2=A0 </div><div><br></div><div>Could aspects of existing H.264=C2=
=A0robustness=C2=A0recommendations (such 3GPP S4-140961, designed for Video=
 over LTE) could be incorporated?=C2=A0 =C2=A0</div><div><br></div><div><fo=
nt color=3D"#000000" face=3D"Times New Roman" size=3D"3">

</font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
 </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
 </font><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
</font><table width=3D"652" style=3D"margin:auto auto auto 2pt;border-colla=
pse:collapse" border=3D"0" cellspacing=3D"0" cellpadding=3D"0"><tbody><tr s=
tyle=3D"height:24.4pt"><td width=3D"84" valign=3D"top" style=3D"background:=
white;padding:0in 2pt;border:1pt solid windowtext;width:63pt;height:24.4pt"=
><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p style=3D"margin:6pt 2.85pt"><span style=3D"color:blue;font-size=
:10pt">S4-140961</span></p><font color=3D"#000000" face=3D"Times New Roman"=
 size=3D"3">
  </font></td><td width=3D"133" valign=3D"top" style=3D"background:white;bo=
rder-width:1pt 1pt 1pt 0px;border-style:solid solid solid none;border-color=
:windowtext windowtext windowtext rgb(0,0,0);padding:0in 2pt;width:100.05pt=
;height:24.4pt"><font color=3D"#000000" face=3D"Times New Roman" size=3D"3"=
>
  </font><p style=3D"margin:6pt 2.85pt 6pt 0in;line-height:10pt"><span lang=
=3D"EN-GB" style=3D"color:black;font-family:&quot;Arial&quot;,&quot;sans-se=
rif&quot;;font-size:10pt">CR 26.114-0287 rev 1 Video
  telephony robustness improvements (</span><span lang=3D"EN-GB" style=3D"c=
olor:black;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;font-size:1=
0pt">Release</span><span lang=3D"EN-GB" style=3D"color:black;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;;font-size:10pt"> 12)</span></p><font=
 color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font></td><td width=3D"113" valign=3D"top" style=3D"background:white;bo=
rder-width:1pt 1pt 1pt 0px;border-style:solid solid solid none;border-color=
:windowtext windowtext windowtext rgb(0,0,0);padding:0in 2pt;width:85.05pt;=
height:24.4pt"><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p style=3D"margin:6pt 2.85pt"><span lang=3D"EN-GB" style=3D"font-=
size:10pt"><font color=3D"#000000">Qualcomm Incorporated, Ericsson</font></=
span></p><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font></td><td width=3D"66" style=3D"background:white;border-width:1pt 1=
pt 1pt 0px;border-style:solid solid solid none;border-color:windowtext wind=
owtext windowtext rgb(0,0,0);padding:0in 2pt;width:49.6pt;height:24.4pt"><f=
ont color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p align=3D"center" style=3D"margin:6pt 2.85pt;text-align:center">=
<span lang=3D"EN-GB" style=3D"font-size:10pt"><font color=3D"#000000">10,
  11</font></span></p><font color=3D"#000000" face=3D"Times New Roman" size=
=3D"3">
  </font></td><td width=3D"85" valign=3D"top" style=3D"background:white;bor=
der-width:1pt 1pt 1pt 0px;border-style:solid solid solid none;border-color:=
windowtext windowtext windowtext rgb(0,0,0);padding:0in 2pt;width:63.8pt;he=
ight:24.4pt"><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p style=3D"margin:6pt 2.85pt"><span lang=3D"EN-GB" style=3D"font-=
size:10pt"><font color=3D"#000000">=C2=A0</font></span></p><font color=3D"#=
000000" face=3D"Times New Roman" size=3D"3">
  </font></td><td width=3D"85" valign=3D"top" style=3D"background:white;bor=
der-width:1pt 1pt 1pt 0px;border-style:solid solid solid none;border-color:=
windowtext windowtext windowtext rgb(0,0,0);padding:0in 2pt;width:63.75pt;h=
eight:24.4pt"><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p style=3D"margin:6pt 2.85pt"><span style=3D"color:blue;font-size=
:10pt">agreed</span></p><font color=3D"#000000" face=3D"Times New Roman" si=
ze=3D"3">
  </font></td><td width=3D"85" valign=3D"top" style=3D"background:white;bor=
der-width:1pt 1pt 1pt 0px;border-style:solid solid solid none;border-color:=
windowtext windowtext windowtext rgb(0,0,0);padding:0in 2pt;width:63.8pt;he=
ight:24.4pt"><font color=3D"#000000" face=3D"Times New Roman" size=3D"3">
  </font><p style=3D"margin:6pt 2.85pt"><span lang=3D"EN-GB" style=3D"font-=
size:10pt"><font color=3D"#000000">15.4</font></span></p><font color=3D"#00=
0000" face=3D"Times New Roman" size=3D"3">
  </font></td></tr></tbody></table><font color=3D"#000000" face=3D"Times Ne=
w Roman" size=3D"3">

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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Consent Freshness: Connection liveness/heartbeats
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/L4ekZArCiOJFsDUHkEpLqxl9wMk
Subject: Re: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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Hi,

I suggest that we REMOVE the liveness/heartbeats feature from the consent r=
efresh draft. As I said earlier, I see no need for sending additional conse=
nt requests in addition to those already sent every 5th second.

Regards,

Christer

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: 10 October 2014 15:23
To: rtcweb@ietf.org
Subject: [rtcweb] Consent Freshness: Connection liveness/heartbeats

Hi,

I have some questions on the connection liveness/heartbeat section of the d=
raft.


Q1:
=3D=3D=3D

The text says:

        "Similarly, if packets haven't been received within a certain perio=
d,
        an application can request a consent check (heartbeat) be generated=
."

BUT, if an endpoint is anyway sending consent requests every 5th second, wh=
y aren't the those enough as hearbeats? As long as the endpoint receives re=
sponses, it knows the remote endpoint is alive (the text even says that an =
endpoint is considered alive as long as some data is received from it).

And, if the endpoint does NOT receive responses to the consent requests, co=
nsent will expire anyway.

I DO understand that an ICE lite may want to send a heartbeat request if it=
 does not receive any data, as it doesn't send consent requests.


Q2:
=3D=3D=3D

The text says:


        "Sending consent checks (heartbeats) at a high rate could allow a

        malicious application to generate congestion, so applications MUST

        NOT be able to send heartbeats at an average rate of more than 1 pe=
r

        second."



I don't think applications should be alowed to generate heartbeats this oft=
en, and I see no need for it.


Q3:
=3D=3D=3D

The spec is unclear on what happens is a response to a heartbeat request is=
 not received. I guess that is an application decision, but I think the spe=
c should be clear that it does NOT have any impact on consent - the "real" =
consent requests are used to determine.


Regards,

Christer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US">I suggest that we REMOVE the liveness/heartbeats feature from the cons=
ent refresh draft. As I said earlier, I see no need for sending additional =
consent requests in addition to those
 already sent every 5<sup>th</sup> second.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D;mso-fareast-language:EN=
-US">Christer<o:p></o:p></span></p>
<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"color:#1F=
497D;mso-fareast-language:EN-US"><o:p>&nbsp;</o:p></span></a></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">From:</span></b><span lang=
=3D"EN-US"> rtcweb [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> 10 October 2014 15:23<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Consent Freshness: Connection liveness/heartbeats<=
o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have some questions on the co=
nnection liveness/heartbeat section of the draft.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Q1:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The text says:<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&#8220;Similarly, if packets haven't been received within a certain period,=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; an ap=
plication can request a consent check (heartbeat) be generated.&#8221;<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">BUT, if an endpoint is anyway sending conse=
nt requests every 5<sup>th</sup> second, why aren&#8217;t the those enough =
as hearbeats? As long as the endpoint receives responses,
 it knows the remote endpoint is alive (the text even says that an endpoint=
 is considered alive as long as some data is received from it).
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">And, if the endpoint does NOT receive respo=
nses to the consent requests, consent will expire anyway.<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">I DO understand that an ICE lite may want t=
o send a heartbeat request if it does not receive any data, as it doesn&#82=
17;t send consent requests.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">Q2:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">The text says:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220=
;Sending consent checks (heartbeats) at a high rate could allow a<o:p></o:p=
></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; malicious a=
pplication to generate congestion, so applications MUST<o:p></o:p></span></=
pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; NOT be able=
 to send heartbeats at an average rate of more than 1 per<o:p></o:p></span>=
</pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; second.&#82=
21;<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">I don&#8217;t think applications should be alowed=
 to generate heartbeats this often, and I see no need for it.<o:p></o:p></s=
pan></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">Q3:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">The spec is unclear on what happens is a re=
sponse to a heartbeat request is not received. I guess that is an applicati=
on decision, but I think the spec should be clear
 that it does NOT have any impact on consent &#8211; the &#8220;real&#8221;=
 consent requests are used to determine.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D47B520ESESSMB209erics_--


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To: "Wyss, Felix" <Felix.Wyss@inin.com>,  "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
References: <20141010004836.12666.88765.idtracker@ietfa.amsl.com> <91953101b2634ec69d14e120ea62d929@CY1PR0501MB1579.namprd05.prod.outlook.com> <CABkgnnWE7czWqi4vQRb5d50N2k95joc_Zcvw6w6g7SOjU9b+9g@mail.gmail.com> <03a3bbbc282b4e6d88d587931b46b5f8@CY1PR0501MB1579.namprd05.prod.outlook.com> <57B40110-2F21-4893-B77C-54F46D18567F@lurchi.franken.de> <1DE35922-E890-40C2-87E9-C8C12235920E@phonefromhere.com> <E53B9B7E-37F0-495C-B1BA-DE0A48AC6D73@lurchi.franken.de> <abdfd3ef58dd40028e8d5e2248b5a995@CY1PR0501MB1579.namprd05.prod.outlook.com> <543A5570.7060208@alvestrand.no> <B53499C4-6B51-4E8E-87C2-C8E5C10DBC34@lurchi.franken.de> <543A9E11.2030605@alvestrand.no> <F5C54F5E-C301-4EC7-9536-E43EA3A16863@lurchi.franken.de> <543ABE69.8030104@alvestrand.no> <99078EA5-5FE7-431F-9C70-EEA882F4C3C6@lurchi.franken.de> <E1FE4C082A89A246A11D7F32A95A17828E5DA2AC@US70UWXCHMBA02.zam.alcatel-lucent.com> <71e2b21516cb496eb4d1ad2b26e53a29@CY1PR0501MB1579.namprd05.prod.outlook.com>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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On 10/13/2014 07:02 PM, Wyss, Felix wrote:
>>>> It has to be the same on both ends of the SCTP connection.
>>> This means you can't remap the streams. Or am I missing something?
>> [Raju] When middle box is in the bearer path each end will see middle box
>> as its peer.
>> If the DCEP is used then it gets terminated at middle box and re-
>> originated towards the other end after necessary stream id mapping.
>> One way to avoid such mapping is to use external negotiation of stream ids
>> as an alternative to inband DCEP.
> They would only see it as a peer if it were a full B2B user agent.  As an example, in VoIP the endpoints do not consider an SBC as a peer--yet it sits in the middle and may inspect the media passing through it.  
>
> With respect to DCEP: If we already define a protocol for establishing the channels in-band, why not make ID selection and conflict resolution an explicit part of it?  That certainly would beat the IMHO rather distasteful DTLS role hack.

When rethinking this, I've come to the conclusion that my original
conclusion (that there couldn't possibly be a point in preserving
channel numbers across a relay) was probably wrong. Please consider that
withdrawn.

I now see a tradeoff between allowing such non-mapping boxes (operating
below the SCTP layer but above the DTLS layer in our model) and avoiding
additional complexity (either extending DCEP with glare resolution -
something it was designed to avoid needing - or requiring such boxes to
not use actpass).

I agree that the DTLS hack is ugly. But glare resolution is ugly too.


>
> --Felix
>


-- 
Surveillance is pervasive. Go Dark.


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Date: Tue, 14 Oct 2014 14:02:34 +0530
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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Here are my thoughts:
- Liveness was originally included in the document as a value added
feature; the same messages used for consent can be used to also determine
liveness and notify the application so that is can take some action before
consent expires. In a sense it is not liveness per se, rather an early
warning to the application before the doomsday.
- The application is free to choose its liveness interval (of course,
choosing an interval greater than the consent period isn't really useful).
If a valid STUN binding response corresponding to one of the STUN requests
sent in that interval has not been received, the browser notifies the
application. This doesn't require new STUN requests to be sent or a change
to how often they are sent for consent (and hence doesn't impact consent in
any way).

I see two options that wouldn't delay this work further:
1. Retain this simple functionality and fix the text, if we think this is
useful/sufficient for applications.
2. Remove liveness entirely from the document, if we need a full-fledged
liveness functionality. There are different ways of determining liveness
depending on the use case (RTCP, DTLS heartbeat, TCP keepalive etc), so we
could spin off a new doc and start with the problem/requirements.

Feedback from the WG would help (either way works for me, though).

Muthu

On Tue, Oct 14, 2014 at 12:09 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> I suggest that we REMOVE the liveness/heartbeats feature from the consent
> refresh draft. As I said earlier, I see no need for sending additional
> consent requests in addition to those already sent every 5th second.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Christer
> Holmberg
> *Sent:* 10 October 2014 15:23
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Consent Freshness: Connection liveness/heartbeats
>
>
>
> Hi,
>
>
>
> I have some questions on the connection liveness/heartbeat section of the
> draft.
>
>
>
>
>
> Q1:
>
> =3D=3D=3D
>
>
>
> The text says:
>
>
>
>         =E2=80=9CSimilarly, if packets haven't been received within a cer=
tain
> period,
>
>         an application can request a consent check (heartbeat) be
> generated.=E2=80=9D
>
>
>
> BUT, if an endpoint is anyway sending consent requests every 5th second,
> why aren=E2=80=99t the those enough as hearbeats? As long as the endpoint=
 receives
> responses, it knows the remote endpoint is alive (the text even says that
> an endpoint is considered alive as long as some data is received from it)=
.
>
>
>
> And, if the endpoint does NOT receive responses to the consent requests,
> consent will expire anyway.
>
>
>
> I DO understand that an ICE lite may want to send a heartbeat request if
> it does not receive any data, as it doesn=E2=80=99t send consent requests=
.
>
>
>
>
>
> Q2:
>
> =3D=3D=3D
>
>
>
> The text says:
>
>
>
>         =E2=80=9CSending consent checks (heartbeats) at a high rate could=
 allow a
>
>         malicious application to generate congestion, so applications MUS=
T
>
>         NOT be able to send heartbeats at an average rate of more than 1 =
per
>
>         second.=E2=80=9D
>
>
>
> I don=E2=80=99t think applications should be alowed to generate heartbeat=
s this often, and I see no need for it.
>
>
>
>
>
> Q3:
>
> =3D=3D=3D
>
>
>
> The spec is unclear on what happens is a response to a heartbeat request
> is not received. I guess that is an application decision, but I think the
> spec should be clear that it does NOT have any impact on consent =E2=80=
=93 the
> =E2=80=9Creal=E2=80=9D consent requests are used to determine.
>
>
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--f46d043749ad62674e05055ddbbc
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">Here are my thoughts:</div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif;font-siz=
e:small"><div class=3D"gmail_default">- Liveness was originally included in=
 the document as a value added feature; the same messages used for consent =
can be used to also determine liveness and notify the application so that i=
s can take some action before consent expires. In a sense it is not livenes=
s per se, rather an early warning to the application before the doomsday.</=
div><div class=3D"gmail_default">- The application is free to choose its li=
veness interval (of course, choosing an interval greater than the consent p=
eriod isn&#39;t really useful). If a valid STUN binding response correspond=
ing to one of the STUN requests sent in that interval has not been received=
, the browser notifies the application. This doesn&#39;t require new STUN r=
equests to be sent or a change to how often they are sent for consent (and =
hence doesn&#39;t impact consent in any way).</div><div class=3D"gmail_defa=
ult"><br></div><div class=3D"gmail_default">I see two options that wouldn&#=
39;t delay this work further:</div><div class=3D"gmail_default">1. Retain t=
his simple functionality and fix the text, if we think this is useful/suffi=
cient for applications.</div><div class=3D"gmail_default">2. Remove livenes=
s entirely from the document, if we need a full-fledged liveness functional=
ity. There are different ways of determining liveness depending on the use =
case (RTCP, DTLS heartbeat, TCP keepalive etc), so we could spin off a new =
doc and start with the problem/requirements.</div><div class=3D"gmail_defau=
lt"><br></div><div class=3D"gmail_default">Feedback from the WG would help =
(either way works for me, though).</div><div class=3D"gmail_default"><br></=
div><div class=3D"gmail_default">Muthu</div></div><div class=3D"gmail_extra=
"><br><div class=3D"gmail_quote">On Tue, Oct 14, 2014 at 12:09 PM, Christer=
 Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsso=
n.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:s=
olid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)">Hi,<u></u><u></=
u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)"><u></u>=C2=A0<u=
></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)">I suggest that =
we REMOVE the liveness/heartbeats feature from the consent refresh draft. A=
s I said earlier, I see no need for sending additional consent requests in =
addition to those
 already sent every 5<sup>th</sup> second.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)"><u></u>=C2=A0<u=
></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)">Regards,<u></u>=
<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)"><u></u>=C2=A0<u=
></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125)">Christer<u></u>=
<u></u></span></p>
<p class=3D"MsoNormal"><a name=3D"1490d62819741b10__MailEndCompose"><span s=
tyle=3D"color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></a></p>
<div>
<div style=3D"border-style:solid none none;border-top-color:rgb(225,225,225=
);border-top-width:1pt;padding:3pt 0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">From:</span></b><span lang=
=3D"EN-US"> rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" targe=
t=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> 10 October 2014 15:23<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> [rtcweb] Consent Freshness: Connection liveness/heartbeats<=
u></u><u></u></span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have some questions on the co=
nnection liveness/heartbeat section of the draft.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Q1:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=3D=3D=3D<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The text says:<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =E2=
=80=9CSimilarly, if packets haven&#39;t been received within a certain peri=
od,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0 an applic=
ation can request a consent check (heartbeat) be generated.=E2=80=9D<u></u>=
<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">BUT, if an endpoint is anyway sending consent r=
equests every 5<sup>th</sup> second, why aren=E2=80=99t the those enough as=
 hearbeats? As long as the endpoint receives responses,
 it knows the remote endpoint is alive (the text even says that an endpoint=
 is considered alive as long as some data is received from it).
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">And, if the endpoint does NOT receive responses=
 to the consent requests, consent will expire anyway.<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">I DO understand that an ICE lite may want to se=
nd a heartbeat request if it does not receive any data, as it doesn=E2=80=
=99t send consent requests.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">Q2:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">=3D=3D=3D<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">The text says:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<pre><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =E2=80=
=9CSending consent checks (heartbeats) at a high rate could allow a<u></u><=
u></u></span></pre>
<pre><span lang=3D"EN-US">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0 malicious a=
pplication to generate congestion, so applications MUST<u></u><u></u></span=
></pre>
<pre><span lang=3D"EN-US">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0 NOT be able=
 to send heartbeats at an average rate of more than 1 per<u></u><u></u></sp=
an></pre>
<pre><span lang=3D"EN-US">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0 second.=E2=
=80=9D<u></u><u></u></span></pre>
<pre><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></pre>
<pre><span lang=3D"EN-US">I don=E2=80=99t think applications should be alow=
ed to generate heartbeats this often, and I see no need for it.<u></u><u></=
u></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">Q3:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">=3D=3D=3D<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">The spec is unclear on what happens is a respon=
se to a heartbeat request is not received. I guess that is an application d=
ecision, but I think the spec should be clear
 that it does NOT have any impact on consent =E2=80=93 the =E2=80=9Creal=E2=
=80=9D consent requests are used to determine.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">Regards,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10pt;font-fa=
mily:&#39;Courier New&#39;">Christer<u></u><u></u></span></p>
</div></div></div>
</div>

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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Thread-Topic: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Muthu Arul Mozhi Perumal" <muthu.arul@gmail.com>
Thread-Topic: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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Date: Tue, 14 Oct 2014 10:28:59 +0000
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Subject: Re: [rtcweb] Consent Freshness: Connection liveness/heartbeats
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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: ext Bernard Aboba <bernard.aboba@gmail.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Dates for rtcweb drafts
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+1


> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Bernard
> Aboba
> Sent: Friday, October 10, 2014 7:39 PM
> To: Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Dates for rtcweb drafts
>=20
> +1.
>=20
>=20
>=20
> > On Oct 10, 2014, at 10:13, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
> >
> > While you try to figure out who should ask the question, I'll give MY
> answer: I think the draft should be adopted as a WG item :)
> >
> > Regards,
> >
> > Christer
> >
> > -----Original Message-----
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
> > Sent: 10 October 2014 19:59
> > To: Martin Thomson
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Dates for rtcweb drafts
> >
> >> On 10/10/2014 06:35 PM, Martin Thomson wrote:
> >>> On 10 October 2014 03:52, Harald Alvestrand <harald@alvestrand.no>
> wrote:
> >>> I'd like the chairs to indicate whether or not they are going to
> >>> entertain -gateways for WG draft status too.
> >> Maybe you should instead ask the working group.
> >
> > That's the question I think it's the WG chairs' job to ask.
> >
> >
> > --
> > Surveillance is pervasive. Go Dark.
> >
> > _______________________________________________
> > 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Hi Harald,

<snip>
>>> 2) It is not required to be endpoint but it shall be middle box.
>> What do you mean by "middle box"? Again, that term is slippery.
> <Partha> I intent to say that the entity which is between two=20
> endpoints and it does not end any media session itself. Here, The=20
> confusion is that WebRTC compatible endpoint which is not an endpoint=20
> but it is a middle box. </Partha>

Seems that this entity (whatever it's called) isn't an endpoint at all, =
so defining terms for endpoints shouldn't be relevant to whatever this =
device is.

There's always more boxes in the middle..... although as long as they =
don't have the DTLS keys, it's limited what they can do to the packets.
<snip>

Could you please update the terminology as "WebRTC compatible device" =
instead of WebRTC compatible endpoint as the entity is not required to =
be endpoint.

Thanks
Partha.



> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Thursday, October 09, 2014 12:29 PM
> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> Subject: Re: [rtcweb] Definitions of WebRTC entities
>=20
> On 10/08/2014 06:39 PM, Parthasarathi R wrote:
> > Hi Harald,
> >
> > Please read inline.
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> >> Sent: Tuesday, October 07, 2014 10:16 PM
> >> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Definitions of WebRTC entities
> >>
> >> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
> >>> Hi Christer,
> >>>
> >>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
> >> endpoint.
> >>> I have bit trouble with WebRTC compatible endpoint as a entity =
name
> >> as
> >>> 1) It may pass SRTP/data channel
> >> What do you mean by "pass"? That's a slippery term.
> > <Partha> "relay" will be more appropriate term as mentioned in Sec 5
> Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
> >
> >>> 2) It is not required to be endpoint but it shall be middle box.
> >> What do you mean by "middle box"? Again, that term is slippery.
> > <Partha> I intent to say that the entity which is between two
> endpoints and it does not end any media session itself. Here, The
> confusion is that WebRTC compatible endpoint which is not an endpoint
> but it is a middle box. </Partha>
>=20
> Seems that this entity (whatever it's called) isn't an endpoint at =
all,
> so defining terms for endpoints shouldn't be relevant to whatever this
> device is.
>=20
> There's always more boxes in the middle..... although as long as they
> don't have the DTLS keys, it's limited what they can do to the =
packets.
>=20
>=20
>=20
> >
> >>> WebRTC gateway looks more appropriate entity name in those
> scenarios.
> >> As written in my proposal, a WebRTC gateway is a WebRTC compatible
> >> endpoint.
> >>
> > <Partha> As per your proposal, we need to define WebRTC compatible
> endpoint first which is super set of WebRTC gateway. Then, we need to
> clarify which kind of WebRTC compatible endpoint qualify as WebRTC
> gateway. But Christer wishes to have only two definition =
(Full/Subset).
> </Partha>
>=20
> And I don't agree with Christer, so then we're two :-)


From nobody Tue Oct 14 21:34:55 2014
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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On 10/15/2014 02:29 AM, Parthasarathi R wrote:
> Hi Harald,
>
> <snip>
>>>> 2) It is not required to be endpoint but it shall be middle box.
>>> What do you mean by "middle box"? Again, that term is slippery.
>> <Partha> I intent to say that the entity which is between two 
>> endpoints and it does not end any media session itself. Here, The 
>> confusion is that WebRTC compatible endpoint which is not an endpoint 
>> but it is a middle box. </Partha>
> Seems that this entity (whatever it's called) isn't an endpoint at all, so defining terms for endpoints shouldn't be relevant to whatever this device is.
>
> There's always more boxes in the middle..... although as long as they don't have the DTLS keys, it's limited what they can do to the packets.
> <snip>
>
> Could you please update the terminology as "WebRTC compatible device" instead of WebRTC compatible endpoint as the entity is not required to be endpoint.

No.

I don't know what the device is, I don't know how to describe it, and I
don't even know if I think it should exist.

I will not define terminology for that.


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> When rethinking this, I've come to the conclusion that my original conclu=
sion
> (that there couldn't possibly be a point in preserving channel numbers ac=
ross
> a relay) was probably wrong. Please consider that withdrawn.

Thank you!


> I now see a tradeoff between allowing such non-mapping boxes (operating
> below the SCTP layer but above the DTLS layer in our model) and avoiding
> additional complexity (either extending DCEP with glare resolution -
> something it was designed to avoid needing - or requiring such boxes to n=
ot
> use actpass).
>=20
> I agree that the DTLS hack is ugly. But glare resolution is ugly too.

Requiring middleboxes to assert a specific DTLS role instead of using actpa=
ss as offerer conflicts with RFC#5763.  That risks incompatibility with end=
points that assume they get the choice as answerer and blindly use active--=
or weren't tested by their vendor to be DTLS server as answerer and misbeha=
ve in other ways.  In our experience, interoperability is already challengi=
ng enough with how vendors interpret RFCs liberally and appear to use a "su=
nny weather" approach to test coverage.  End customers won't take "big vend=
or X doesn't implement an RFC correctly" as excuse why their system doesn't=
 work properly or won't interoperate.  They paid a lot of money to have a w=
orking solution--which then requires hacks and workarounds to make it work =
somehow.  Abstraction leaks and unexpected dependencies greatly increases t=
he risk of incorrect implementations and bugs. =20

Speaking from an implementer's perspective, I thus strongly recommend again=
st codification of conflicting requirements between RFCs and/or dubious dep=
endencies across layers. =20


--Felix


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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: ext Parthasarathi R <partha@parthasarathi.co.in>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Hi Harald,

I wonder whether we can simplify the set of terms by removing "WebRTC endpo=
int".

What was the reason for introducing the term "WebRTC endpoint"? Wouldn't it=
 be enough to have "WebRTC UA", "WebRTC device" and "WebRTC-compatible devi=
ce" (plus WebRTC gateway as special widespread case of compatible device)?


A slight change in the chain of terms in -overview-12 could reduce the numb=
er of definitions without losing meaning (I think):


1) A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) is som=
ething that conforms to both the protocol specification and the Javascript =
API defined above. A WebRTC User Agent conforms to both the protocol specif=
ication and the Javascript API.=20
--> keep as is

2) A WebRTC device is something that conforms to the protocol specification=
, but does not claim to implement the Javascript API.=20
--> replace "claim" by "have". This results in <WebRTC UA> IS-A <WebRTC dev=
ice> and we can pull out "WebRTC endpoint". (This is also better in line th=
an the current chain with the statement in the draft "All WebRTC browsers (=
UAs) are WebRTC devices, so any requirement on a WebRTC device also applies=
 to a WebRTC browser"

3) A WebRTC endpoint is either a WebRTC User Agent or a WebRTC device.
--> delete

4) A WebRTC-compatible endpoint is an endpoint that is capable of successfu=
lly communicating with a WebRTC endpoint, but may fail to meet some require=
ments of a WebRTC endpoint.  This may limit where in the network such an en=
dpoint can be attached, or may limit the security guarantees that it offers=
 to others.
--> replace "endpoint" by "device"

5) A WebRTC gateway is a WebRTC-compatible endpoint that mediates traffic t=
o non-WebRTC entities.
--> replace "endpoint" by "device"
=A0
Kind regards,
Uwe=20



> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext
> Parthasarathi R
> Sent: Wednesday, October 15, 2014 2:30 AM
> To: 'Harald Alvestrand'; 'Christer Holmberg'; rtcweb@ietf.org
> Subject: Re: [rtcweb] Definitions of WebRTC entities
>=20
> Hi Harald,
>=20
> <snip>
> >>> 2) It is not required to be endpoint but it shall be middle box.
> >> What do you mean by "middle box"? Again, that term is slippery.
> > <Partha> I intent to say that the entity which is between two
> > endpoints and it does not end any media session itself. Here, The
> > confusion is that WebRTC compatible endpoint which is not an endpoint
> > but it is a middle box. </Partha>
>=20
> Seems that this entity (whatever it's called) isn't an endpoint at all,
> so defining terms for endpoints shouldn't be relevant to whatever this
> device is.
>=20
> There's always more boxes in the middle..... although as long as they
> don't have the DTLS keys, it's limited what they can do to the packets.
> <snip>
>=20
> Could you please update the terminology as "WebRTC compatible device"
> instead of WebRTC compatible endpoint as the entity is not required to
> be endpoint.
>=20
> Thanks
> Partha.
>=20
>=20
>=20
> > -----Original Message-----
> > From: Harald Alvestrand [mailto:harald@alvestrand.no]
> > Sent: Thursday, October 09, 2014 12:29 PM
> > To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> > Subject: Re: [rtcweb] Definitions of WebRTC entities
> >
> > On 10/08/2014 06:39 PM, Parthasarathi R wrote:
> > > Hi Harald,
> > >
> > > Please read inline.
> > >
> > > Thanks
> > > Partha
> > >
> > >> -----Original Message-----
> > >> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> > >> Sent: Tuesday, October 07, 2014 10:16 PM
> > >> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> > >> Subject: Re: [rtcweb] Definitions of WebRTC entities
> > >>
> > >> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
> > >>> Hi Christer,
> > >>>
> > >>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
> > >> endpoint.
> > >>> I have bit trouble with WebRTC compatible endpoint as a entity
> name
> > >> as
> > >>> 1) It may pass SRTP/data channel
> > >> What do you mean by "pass"? That's a slippery term.
> > > <Partha> "relay" will be more appropriate term as mentioned in Sec
> 5
> > Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
> > >
> > >>> 2) It is not required to be endpoint but it shall be middle box.
> > >> What do you mean by "middle box"? Again, that term is slippery.
> > > <Partha> I intent to say that the entity which is between two
> > endpoints and it does not end any media session itself. Here, The
> > confusion is that WebRTC compatible endpoint which is not an endpoint
> > but it is a middle box. </Partha>
> >
> > Seems that this entity (whatever it's called) isn't an endpoint at
> all,
> > so defining terms for endpoints shouldn't be relevant to whatever
> this
> > device is.
> >
> > There's always more boxes in the middle..... although as long as they
> > don't have the DTLS keys, it's limited what they can do to the
> packets.
> >
> >
> >
> > >
> > >>> WebRTC gateway looks more appropriate entity name in those
> > scenarios.
> > >> As written in my proposal, a WebRTC gateway is a WebRTC compatible
> > >> endpoint.
> > >>
> > > <Partha> As per your proposal, we need to define WebRTC compatible
> > endpoint first which is super set of WebRTC gateway. Then, we need to
> > clarify which kind of WebRTC compatible endpoint qualify as WebRTC
> > gateway. But Christer wishes to have only two definition
> (Full/Subset).
> > </Partha>
> >
> > And I don't agree with Christer, so then we're two :-)
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Oct 15 02:39:59 2014
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Date: Wed, 15 Oct 2014 11:39:51 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>,  ext Parthasarathi R <partha@parthasarathi.co.in>, 'Christer Holmberg' <christer.holmberg@ericsson.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
References: <542E53D2.5040500@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465376@ESESSMB209.ericsson.se> <C45C84E3-FC63-4DF6-ABDE-701FC7584E3C@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465985@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D465A34@ESESSMB209.ericsson.se> <00f501cfe24a$b8515930$28f40b90$@co.in> <543418D5.8010509@alvestrand.no> <006301cfe316$6d3c5590$47b500b0$@co.in> <54363216.3060700@alvestrand.no> <010d01cfe80f$1c8e3930$55aaab90$@co.in> <56C2F665D49E0341B9DF5938005ACDF8194C0459@DEMUMBX005.nsn-intra.net>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/d3-u8fFHhlOGVdrNK-vKPi0WTi4
Subject: Re: [rtcweb] Definitions of WebRTC entities
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On 10/15/2014 10:47 AM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
> Hi Harald,
>
> I wonder whether we can simplify the set of terms by removing "WebRTC endpoint".

The reason for adding it was that it was already used in the RTCWEB RTP 
document, in harmony with the usage of "endpoint" in other RTP-releated 
specs.

I'll take others' input on whether it's worth removing the term or 
keeping it.

Note - from a strictly formalistic point of view, replacing all 
occurences of "WebRTC device" with "WebRTC endpoint" would have exactly 
the same effect. But the terms feel more comfortable to use in different 
contexts.

>
> What was the reason for introducing the term "WebRTC endpoint"? Wouldn't it be enough to have "WebRTC UA", "WebRTC device" and "WebRTC-compatible device" (plus WebRTC gateway as special widespread case of compatible device)?
>
>
> A slight change in the chain of terms in -overview-12 could reduce the number of definitions without losing meaning (I think):
>
>
> 1) A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) is something that conforms to both the protocol specification and the Javascript API defined above. A WebRTC User Agent conforms to both the protocol specification and the Javascript API.
> --> keep as is
>
> 2) A WebRTC device is something that conforms to the protocol specification, but does not claim to implement the Javascript API.
> --> replace "claim" by "have". This results in <WebRTC UA> IS-A <WebRTC device> and we can pull out "WebRTC endpoint". (This is also better in line than the current chain with the statement in the draft "All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a WebRTC device also applies to a WebRTC browser"
>
> 3) A WebRTC endpoint is either a WebRTC User Agent or a WebRTC device.
> --> delete
>
> 4) A WebRTC-compatible endpoint is an endpoint that is capable of successfully communicating with a WebRTC endpoint, but may fail to meet some requirements of a WebRTC endpoint.  This may limit where in the network such an endpoint can be attached, or may limit the security guarantees that it offers to others.
> --> replace "endpoint" by "device"
>
> 5) A WebRTC gateway is a WebRTC-compatible endpoint that mediates traffic to non-WebRTC entities.
> --> replace "endpoint" by "device"
>   
> Kind regards,
> Uwe
>
>
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext
>> Parthasarathi R
>> Sent: Wednesday, October 15, 2014 2:30 AM
>> To: 'Harald Alvestrand'; 'Christer Holmberg'; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> Hi Harald,
>>
>> <snip>
>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>> What do you mean by "middle box"? Again, that term is slippery.
>>> <Partha> I intent to say that the entity which is between two
>>> endpoints and it does not end any media session itself. Here, The
>>> confusion is that WebRTC compatible endpoint which is not an endpoint
>>> but it is a middle box. </Partha>
>> Seems that this entity (whatever it's called) isn't an endpoint at all,
>> so defining terms for endpoints shouldn't be relevant to whatever this
>> device is.
>>
>> There's always more boxes in the middle..... although as long as they
>> don't have the DTLS keys, it's limited what they can do to the packets.
>> <snip>
>>
>> Could you please update the terminology as "WebRTC compatible device"
>> instead of WebRTC compatible endpoint as the entity is not required to
>> be endpoint.
>>
>> Thanks
>> Partha.
>>
>>
>>
>>> -----Original Message-----
>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>> Sent: Thursday, October 09, 2014 12:29 PM
>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>
>>> On 10/08/2014 06:39 PM, Parthasarathi R wrote:
>>>> Hi Harald,
>>>>
>>>> Please read inline.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>>>> Sent: Tuesday, October 07, 2014 10:16 PM
>>>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>>>
>>>>> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
>>>>>> Hi Christer,
>>>>>>
>>>>>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
>>>>> endpoint.
>>>>>> I have bit trouble with WebRTC compatible endpoint as a entity
>> name
>>>>> as
>>>>>> 1) It may pass SRTP/data channel
>>>>> What do you mean by "pass"? That's a slippery term.
>>>> <Partha> "relay" will be more appropriate term as mentioned in Sec
>> 5
>>> Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
>>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>>> What do you mean by "middle box"? Again, that term is slippery.
>>>> <Partha> I intent to say that the entity which is between two
>>> endpoints and it does not end any media session itself. Here, The
>>> confusion is that WebRTC compatible endpoint which is not an endpoint
>>> but it is a middle box. </Partha>
>>>
>>> Seems that this entity (whatever it's called) isn't an endpoint at
>> all,
>>> so defining terms for endpoints shouldn't be relevant to whatever
>> this
>>> device is.
>>>
>>> There's always more boxes in the middle..... although as long as they
>>> don't have the DTLS keys, it's limited what they can do to the
>> packets.
>>>
>>>
>>>>>> WebRTC gateway looks more appropriate entity name in those
>>> scenarios.
>>>>> As written in my proposal, a WebRTC gateway is a WebRTC compatible
>>>>> endpoint.
>>>>>
>>>> <Partha> As per your proposal, we need to define WebRTC compatible
>>> endpoint first which is super set of WebRTC gateway. Then, we need to
>>> clarify which kind of WebRTC compatible endpoint qualify as WebRTC
>>> gateway. But Christer wishes to have only two definition
>> (Full/Subset).
>>> </Partha>
>>>
>>> And I don't agree with Christer, so then we're two :-)
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Oct 15 03:38:59 2014
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>, ext Parthasarathi R <partha@parthasarathi.co.in>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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References: <542E53D2.5040500@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465376@ESESSMB209.ericsson.se> <C45C84E3-FC63-4DF6-ABDE-701FC7584E3C@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465985@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D465A34@ESESSMB209.ericsson.se> <00f501cfe24a$b8515930$28f40b90$@co.in> <543418D5.8010509@alvestrand.no> <006301cfe316$6d3c5590$47b500b0$@co.in> <54363216.3060700@alvestrand.no> <010d01cfe80f$1c8e3930$55aaab90$@co.in> <56C2F665D49E0341B9DF5938005ACDF8194C0459@DEMUMBX005.nsn-intra.net> <543E40E7.4030609@alvestrand.no>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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I'm also reluctant using "endpoint" as qualifier for WebRTC entities becaus=
e in communication networks the notion of endpoint is fundamentally tight t=
o a protocol (layer) (see e.g. (N)-connection-endpoint in ITU-T X.200).
Now, "WebRTC" is not a single protocol, rather a suite of protocols in conc=
ert.
E.g., we got
(DTLS)-connection-endpoints,
(SCTP)-connection-endpoints,
(UDP)-connection-endpoints,
(RTP)-connection-endpoints,
etc=20
within an WebRTC entity.

Two cents from my side, of course you may always expand these OSI-ISO base =
terms towards an (WebRTC)-endpoint (by highlighting the service access poin=
t aspects and the "(WebRTC)-connection" concept).

-Albrecht



-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestran=
d
Sent: Mittwoch, 15. Oktober 2014 11:40
To: Rauschenbach, Uwe (NSN - DE/Munich); ext Parthasarathi R; 'Christer Hol=
mberg'; rtcweb@ietf.org
Subject: Re: [rtcweb] Definitions of WebRTC entities

On 10/15/2014 10:47 AM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
> Hi Harald,
>
> I wonder whether we can simplify the set of terms by removing "WebRTC end=
point".

The reason for adding it was that it was already used in the RTCWEB RTP doc=
ument, in harmony with the usage of "endpoint" in other RTP-releated specs.

I'll take others' input on whether it's worth removing the term or keeping =
it.

Note - from a strictly formalistic point of view, replacing all occurences =
of "WebRTC device" with "WebRTC endpoint" would have exactly the same effec=
t. But the terms feel more comfortable to use in different contexts.

>
> What was the reason for introducing the term "WebRTC endpoint"? Wouldn't =
it be enough to have "WebRTC UA", "WebRTC device" and "WebRTC-compatible de=
vice" (plus WebRTC gateway as special widespread case of compatible device)=
?
>
>
> A slight change in the chain of terms in -overview-12 could reduce the nu=
mber of definitions without losing meaning (I think):
>
>
> 1) A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) is s=
omething that conforms to both the protocol specification and the Javascrip=
t API defined above. A WebRTC User Agent conforms to both the protocol spec=
ification and the Javascript API.
> --> keep as is
>
> 2) A WebRTC device is something that conforms to the protocol specificati=
on, but does not claim to implement the Javascript API.
> --> replace "claim" by "have". This results in <WebRTC UA> IS-A <WebRTC d=
evice> and we can pull out "WebRTC endpoint". (This is also better in line =
than the current chain with the statement in the draft "All WebRTC browsers=
 (UAs) are WebRTC devices, so any requirement on a WebRTC device also appli=
es to a WebRTC browser"
>
> 3) A WebRTC endpoint is either a WebRTC User Agent or a WebRTC device.
> --> delete
>
> 4) A WebRTC-compatible endpoint is an endpoint that is capable of success=
fully communicating with a WebRTC endpoint, but may fail to meet some requi=
rements of a WebRTC endpoint.  This may limit where in the network such an =
endpoint can be attached, or may limit the security guarantees that it offe=
rs to others.
> --> replace "endpoint" by "device"
>
> 5) A WebRTC gateway is a WebRTC-compatible endpoint that mediates traffic=
 to non-WebRTC entities.
> --> replace "endpoint" by "device"
>  =20
> Kind regards,
> Uwe
>
>
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext=20
>> Parthasarathi R
>> Sent: Wednesday, October 15, 2014 2:30 AM
>> To: 'Harald Alvestrand'; 'Christer Holmberg'; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> Hi Harald,
>>
>> <snip>
>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>> What do you mean by "middle box"? Again, that term is slippery.
>>> <Partha> I intent to say that the entity which is between two=20
>>> endpoints and it does not end any media session itself. Here, The=20
>>> confusion is that WebRTC compatible endpoint which is not an=20
>>> endpoint but it is a middle box. </Partha>
>> Seems that this entity (whatever it's called) isn't an endpoint at=20
>> all, so defining terms for endpoints shouldn't be relevant to=20
>> whatever this device is.
>>
>> There's always more boxes in the middle..... although as long as they=20
>> don't have the DTLS keys, it's limited what they can do to the packets.
>> <snip>
>>
>> Could you please update the terminology as "WebRTC compatible device"
>> instead of WebRTC compatible endpoint as the entity is not required=20
>> to be endpoint.
>>
>> Thanks
>> Partha.
>>
>>
>>
>>> -----Original Message-----
>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>> Sent: Thursday, October 09, 2014 12:29 PM
>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>
>>> On 10/08/2014 06:39 PM, Parthasarathi R wrote:
>>>> Hi Harald,
>>>>
>>>> Please read inline.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>>>> Sent: Tuesday, October 07, 2014 10:16 PM
>>>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>>>
>>>>> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
>>>>>> Hi Christer,
>>>>>>
>>>>>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
>>>>> endpoint.
>>>>>> I have bit trouble with WebRTC compatible endpoint as a entity
>> name
>>>>> as
>>>>>> 1) It may pass SRTP/data channel
>>>>> What do you mean by "pass"? That's a slippery term.
>>>> <Partha> "relay" will be more appropriate term as mentioned in Sec
>> 5
>>> Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
>>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>>> What do you mean by "middle box"? Again, that term is slippery.
>>>> <Partha> I intent to say that the entity which is between two
>>> endpoints and it does not end any media session itself. Here, The=20
>>> confusion is that WebRTC compatible endpoint which is not an=20
>>> endpoint but it is a middle box. </Partha>
>>>
>>> Seems that this entity (whatever it's called) isn't an endpoint at
>> all,
>>> so defining terms for endpoints shouldn't be relevant to whatever
>> this
>>> device is.
>>>
>>> There's always more boxes in the middle..... although as long as=20
>>> they don't have the DTLS keys, it's limited what they can do to the
>> packets.
>>>
>>>
>>>>>> WebRTC gateway looks more appropriate entity name in those
>>> scenarios.
>>>>> As written in my proposal, a WebRTC gateway is a WebRTC compatible=20
>>>>> endpoint.
>>>>>
>>>> <Partha> As per your proposal, we need to define WebRTC compatible
>>> endpoint first which is super set of WebRTC gateway. Then, we need=20
>>> to clarify which kind of WebRTC compatible endpoint qualify as=20
>>> WebRTC gateway. But Christer wishes to have only two definition
>> (Full/Subset).
>>> </Partha>
>>>
>>> And I don't agree with Christer, so then we're two :-)
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

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


From nobody Wed Oct 15 07:21:11 2014
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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I surely missed a recent thread on it but, any plan to discuss MTI
Video Codec in HI?

I read in Jonathan's post it's part of the agenda?
http://blogs.cisco.com/collaboration/ciscos-openh264-now-part-of-firefox/

Cheers,
-Victor

On Wed, Sep 11, 2013 at 7:51 PM, Victor Pascual Avila
<victor.pascual.avila@gmail.com> wrote:
> Thanks for the feedback Ted!
>
> On Wed, Sep 11, 2013 at 7:48 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
>> Hi Victor,
>>
>> The chairs have been discussion the structure of the upcoming meeting, a=
nd
>> you should expect something from us soon.
>>
>> thanks,
>>
>> Ted
>>
>>
>> On Tue, Sep 10, 2013 at 6:47 AM, Victor Pascual Avila
>> <victor.pascual.avila@gmail.com> wrote:
>>>
>>> Dear WG Chairs,
>>>
>>> What's the plan for the MTI video codec discussion? Are we planning to
>>> discuss this during our next meeting? Is there a reason to delay this
>>> decision?
>>>
>>> My apologies if this has already been discussed but could not find it
>>> in the archives.
>>>
>>> Thanks,
>>> -Victor
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
>
> --
> Victor Pascual =C3=81vila



--=20
Victor Pascual =C3=81vila


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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: "ext Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>,  Harald Alvestrand <harald@alvestrand.no>, ext Parthasarathi R <partha@parthasarathi.co.in>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Definitions of WebRTC entities
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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Hi Harald, all,

I concur with Albrecht w.r.t. the meaning of "WebRTC endpoint". You mention=
 below that "endpoint" is included mainly because it is used in the rtp-usa=
ge draft. But an "RTP endpoint" (a source and/or destination of RTP streams=
) may be different from a "WebRTC endpoint" (e.g. a mixing device).

I believe that in the RTP usage draft, the meaning of "endpoint" is usually=
 "RTP endpoint" with the interpretation "RTP endpoint that supports WebRTC"=
. Terms used as well are "WebRTC implementations of RTP endpoints" and "Web=
RTC implementation". If we interpret "endpoint" that way, and maybe do some=
 editorial update of -rtp-usage replacing "WebRTC endpoint" with "WebRTC im=
plementation" or "WebRTC-aware endpoint" (I can volunteer to write a propos=
al), there will be no clash with the proposal to drop "WebRTC endpoint" fro=
m -overview.

The more neutral "device" for WebRTC entities allows to include entities th=
at may not be the original source or final sink of RTP media streams. "Enti=
ty" would be an alternative for "Device" but I seem to recall there was dis=
like for that term previously.=20

Kind regards,
Uwe=20


> -----Original Message-----
> From: ext Schwarz, Albrecht (Albrecht)
> [mailto:albrecht.schwarz@alcatel-lucent.com]
> Sent: Wednesday, October 15, 2014 12:38 PM
> To: Harald Alvestrand; Rauschenbach, Uwe (NSN - DE/Munich); ext
> Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> Subject: RE: [rtcweb] Definitions of WebRTC entities
>=20
> I'm also reluctant using "endpoint" as qualifier for WebRTC entities
> because in communication networks the notion of endpoint is
> fundamentally tight to a protocol (layer) (see e.g. (N)-connection-
> endpoint in ITU-T X.200).
> Now, "WebRTC" is not a single protocol, rather a suite of protocols in
> concert.
> E.g., we got
> (DTLS)-connection-endpoints,
> (SCTP)-connection-endpoints,
> (UDP)-connection-endpoints,
> (RTP)-connection-endpoints,
> etc
> within an WebRTC entity.
>=20
> Two cents from my side, of course you may always expand these OSI-ISO
> base terms towards an (WebRTC)-endpoint (by highlighting the service
> access point aspects and the "(WebRTC)-connection" concept).
>=20
> -Albrecht
>=20
>=20
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
> Sent: Mittwoch, 15. Oktober 2014 11:40
> To: Rauschenbach, Uwe (NSN - DE/Munich); ext Parthasarathi R; 'Christer
> Holmberg'; rtcweb@ietf.org
> Subject: Re: [rtcweb] Definitions of WebRTC entities
>=20
> On 10/15/2014 10:47 AM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
> > Hi Harald,
> >
> > I wonder whether we can simplify the set of terms by removing "WebRTC
> endpoint".
>=20
> The reason for adding it was that it was already used in the RTCWEB RTP
> document, in harmony with the usage of "endpoint" in other RTP-releated
> specs.
>=20
> I'll take others' input on whether it's worth removing the term or
> keeping it.
>=20
> Note - from a strictly formalistic point of view, replacing all
> occurences of "WebRTC device" with "WebRTC endpoint" would have exactly
> the same effect. But the terms feel more comfortable to use in
> different contexts.
>=20
> >
> > What was the reason for introducing the term "WebRTC endpoint"?
> Wouldn't it be enough to have "WebRTC UA", "WebRTC device" and "WebRTC-
> compatible device" (plus WebRTC gateway as special widespread case of
> compatible device)?
> >
> >
> > A slight change in the chain of terms in -overview-12 could reduce
> the number of definitions without losing meaning (I think):
> >
> >
> > 1) A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser)
> is something that conforms to both the protocol specification and the
> Javascript API defined above. A WebRTC User Agent conforms to both the
> protocol specification and the Javascript API.
> > --> keep as is
> >
> > 2) A WebRTC device is something that conforms to the protocol
> specification, but does not claim to implement the Javascript API.
> > --> replace "claim" by "have". This results in <WebRTC UA> IS-A
> <WebRTC device> and we can pull out "WebRTC endpoint". (This is also
> better in line than the current chain with the statement in the draft
> "All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a
> WebRTC device also applies to a WebRTC browser"
> >
> > 3) A WebRTC endpoint is either a WebRTC User Agent or a WebRTC
> device.
> > --> delete
> >
> > 4) A WebRTC-compatible endpoint is an endpoint that is capable of
> successfully communicating with a WebRTC endpoint, but may fail to meet
> some requirements of a WebRTC endpoint.  This may limit where in the
> network such an endpoint can be attached, or may limit the security
> guarantees that it offers to others.
> > --> replace "endpoint" by "device"
> >
> > 5) A WebRTC gateway is a WebRTC-compatible endpoint that mediates
> traffic to non-WebRTC entities.
> > --> replace "endpoint" by "device"
> >
> > Kind regards,
> > Uwe
> >
> >
> >
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext
> >> Parthasarathi R
> >> Sent: Wednesday, October 15, 2014 2:30 AM
> >> To: 'Harald Alvestrand'; 'Christer Holmberg'; rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Definitions of WebRTC entities
> >>
> >> Hi Harald,
> >>
> >> <snip>
> >>>>> 2) It is not required to be endpoint but it shall be middle box.
> >>>> What do you mean by "middle box"? Again, that term is slippery.
> >>> <Partha> I intent to say that the entity which is between two
> >>> endpoints and it does not end any media session itself. Here, The
> >>> confusion is that WebRTC compatible endpoint which is not an
> >>> endpoint but it is a middle box. </Partha>
> >> Seems that this entity (whatever it's called) isn't an endpoint at
> >> all, so defining terms for endpoints shouldn't be relevant to
> >> whatever this device is.
> >>
> >> There's always more boxes in the middle..... although as long as
> they
> >> don't have the DTLS keys, it's limited what they can do to the
> packets.
> >> <snip>
> >>
> >> Could you please update the terminology as "WebRTC compatible
> device"
> >> instead of WebRTC compatible endpoint as the entity is not required
> >> to be endpoint.
> >>
> >> Thanks
> >> Partha.
> >>
> >>
> >>
> >>> -----Original Message-----
> >>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> >>> Sent: Thursday, October 09, 2014 12:29 PM
> >>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> >>> Subject: Re: [rtcweb] Definitions of WebRTC entities
> >>>
> >>> On 10/08/2014 06:39 PM, Parthasarathi R wrote:
> >>>> Hi Harald,
> >>>>
> >>>> Please read inline.
> >>>>
> >>>> Thanks
> >>>> Partha
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> >>>>> Sent: Tuesday, October 07, 2014 10:16 PM
> >>>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
> >>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
> >>>>>
> >>>>> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
> >>>>>> Hi Christer,
> >>>>>>
> >>>>>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
> >>>>> endpoint.
> >>>>>> I have bit trouble with WebRTC compatible endpoint as a entity
> >> name
> >>>>> as
> >>>>>> 1) It may pass SRTP/data channel
> >>>>> What do you mean by "pass"? That's a slippery term.
> >>>> <Partha> "relay" will be more appropriate term as mentioned in Sec
> >> 5
> >>> Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
> >>>>>> 2) It is not required to be endpoint but it shall be middle box.
> >>>>> What do you mean by "middle box"? Again, that term is slippery.
> >>>> <Partha> I intent to say that the entity which is between two
> >>> endpoints and it does not end any media session itself. Here, The
> >>> confusion is that WebRTC compatible endpoint which is not an
> >>> endpoint but it is a middle box. </Partha>
> >>>
> >>> Seems that this entity (whatever it's called) isn't an endpoint at
> >> all,
> >>> so defining terms for endpoints shouldn't be relevant to whatever
> >> this
> >>> device is.
> >>>
> >>> There's always more boxes in the middle..... although as long as
> >>> they don't have the DTLS keys, it's limited what they can do to the
> >> packets.
> >>>
> >>>
> >>>>>> WebRTC gateway looks more appropriate entity name in those
> >>> scenarios.
> >>>>> As written in my proposal, a WebRTC gateway is a WebRTC
> compatible
> >>>>> endpoint.
> >>>>>
> >>>> <Partha> As per your proposal, we need to define WebRTC compatible
> >>> endpoint first which is super set of WebRTC gateway. Then, we need
> >>> to clarify which kind of WebRTC compatible endpoint qualify as
> >>> WebRTC gateway. But Christer wishes to have only two definition
> >> (Full/Subset).
> >>> </Partha>
> >>>
> >>> And I don't agree with Christer, so then we're two :-)
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Ted Hardie <ted.ietf@gmail.com>
To: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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Hi Victor,

As you may remember, we tabled discussion of video codecs back in January,
and said that the topic could be re-opened 6 weeks prior to IETF 91.  We do
anticipate discussing it at this meeting, likely in the Thursday slot
(since there is a related BoF earlier that day).

regards,

Ted Hardie

On Wed, Oct 15, 2014 at 7:20 AM, Victor Pascual Avila <
victor.pascual.avila@gmail.com> wrote:

> I surely missed a recent thread on it but, any plan to discuss MTI
> Video Codec in HI?
>
> I read in Jonathan's post it's part of the agenda?
> http://blogs.cisco.com/collaboration/ciscos-openh264-now-part-of-firefox/
>
> Cheers,
> -Victor
>
> On Wed, Sep 11, 2013 at 7:51 PM, Victor Pascual Avila
> <victor.pascual.avila@gmail.com> wrote:
> > Thanks for the feedback Ted!
> >
> > On Wed, Sep 11, 2013 at 7:48 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
> >> Hi Victor,
> >>
> >> The chairs have been discussion the structure of the upcoming meeting,
> and
> >> you should expect something from us soon.
> >>
> >> thanks,
> >>
> >> Ted
> >>
> >>
> >> On Tue, Sep 10, 2013 at 6:47 AM, Victor Pascual Avila
> >> <victor.pascual.avila@gmail.com> wrote:
> >>>
> >>> Dear WG Chairs,
> >>>
> >>> What's the plan for the MTI video codec discussion? Are we planning t=
o
> >>> discuss this during our next meeting? Is there a reason to delay this
> >>> decision?
> >>>
> >>> My apologies if this has already been discussed but could not find it
> >>> in the archives.
> >>>
> >>> Thanks,
> >>> -Victor
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >>
> >
> >
> >
> > --
> > Victor Pascual =C3=81vila
>
>
>
> --
> Victor Pascual =C3=81vila
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">Hi Victor,<br><br></div><div class=3D"gmail_defa=
ult" style=3D"font-family:tahoma,sans-serif;font-size:small">As you may rem=
ember, we tabled discussion of video codecs back in January, and said that =
the topic could be re-opened 6 weeks prior to IETF 91.=C2=A0 We do anticipa=
te discussing it at this meeting, likely in the Thursday slot (since there =
is a related BoF earlier that day).<br><br></div><div class=3D"gmail_defaul=
t" style=3D"font-family:tahoma,sans-serif;font-size:small">regards,<br><br>=
Ted Hardie<br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On Wed, Oct 15, 2014 at 7:20 AM, Victor Pascual Avila <span dir=3D=
"ltr">&lt;<a href=3D"mailto:victor.pascual.avila@gmail.com" target=3D"_blan=
k">victor.pascual.avila@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">I surely missed a recent thread on it but, any plan to discu=
ss MTI<br>
Video Codec in HI?<br>
<br>
I read in Jonathan&#39;s post it&#39;s part of the agenda?<br>
<a href=3D"http://blogs.cisco.com/collaboration/ciscos-openh264-now-part-of=
-firefox/" target=3D"_blank">http://blogs.cisco.com/collaboration/ciscos-op=
enh264-now-part-of-firefox/</a><br>
<br>
Cheers,<br>
-Victor<br>
<br>
On Wed, Sep 11, 2013 at 7:51 PM, Victor Pascual Avila<br>
<span class=3D"im HOEnZb">&lt;<a href=3D"mailto:victor.pascual.avila@gmail.=
com">victor.pascual.avila@gmail.com</a>&gt; wrote:<br>
&gt; Thanks for the feedback Ted!<br>
&gt;<br>
&gt; On Wed, Sep 11, 2013 at 7:48 PM, Ted Hardie &lt;<a href=3D"mailto:ted.=
ietf@gmail.com">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt;&gt; Hi Victor,<br>
&gt;&gt;<br>
&gt;&gt; The chairs have been discussion the structure of the upcoming meet=
ing, and<br>
&gt;&gt; you should expect something from us soon.<br>
&gt;&gt;<br>
&gt;&gt; thanks,<br>
&gt;&gt;<br>
&gt;&gt; Ted<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Tue, Sep 10, 2013 at 6:47 AM, Victor Pascual Avila<br>
&gt;&gt; &lt;<a href=3D"mailto:victor.pascual.avila@gmail.com">victor.pascu=
al.avila@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Dear WG Chairs,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What&#39;s the plan for the MTI video codec discussion? Are we=
 planning to<br>
&gt;&gt;&gt; discuss this during our next meeting? Is there a reason to del=
ay this<br>
&gt;&gt;&gt; decision?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; My apologies if this has already been discussed but could not =
find it<br>
&gt;&gt;&gt; in the archives.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Thanks,<br>
&gt;&gt;&gt; -Victor<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" targe=
t=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Victor Pascual =C3=81vila<br>
<br>
<br>
<br>
</span><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
Victor Pascual =C3=81vila<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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Date: Wed, 15 Oct 2014 19:26:41 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>,  "ext Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>, ext Parthasarathi R <partha@parthasarathi.co.in>,  'Christer Holmberg' <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
References: <542E53D2.5040500@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465376@ESESSMB209.ericsson.se> <C45C84E3-FC63-4DF6-ABDE-701FC7584E3C@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D465985@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D465A34@ESESSMB209.ericsson.se> <00f501cfe24a$b8515930$28f40b90$@co.in> <543418D5.8010509@alvestrand.no> <006301cfe316$6d3c5590$47b500b0$@co.in> <54363216.3060700@alvestrand.no> <010d01cfe80f$1c8e3930$55aaab90$@co.in> <56C2F665D49E0341B9DF5938005ACDF8194C0459@DEMUMBX005.nsn-intra.net> <543E40E7.4030609@alvestrand.no> <786615F3A85DF44AA2A76164A71FE1AC3418E0@FR711WXCHMBA03.zeu.alcatel-lucent.com> <56C2F665D49E0341B9DF5938005ACDF8194C1087@DEMUMBX005.nsn-intra.net>
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Subject: Re: [rtcweb] Definitions of WebRTC entities
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On 10/15/2014 06:55 PM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
> Hi Harald, all,
>
> I concur with Albrecht w.r.t. the meaning of "WebRTC endpoint". You mention below that "endpoint" is included mainly because it is used in the rtp-usage draft. But an "RTP endpoint" (a source and/or destination of RTP streams) may be different from a "WebRTC endpoint" (e.g. a mixing device).
To be more precise: The term "WebRTC endpoint" is used in the rtp-usage 
draft. Three times.
>
> I believe that in the RTP usage draft, the meaning of "endpoint" is usually "RTP endpoint" with the interpretation "RTP endpoint that supports WebRTC". Terms used as well are "WebRTC implementations of RTP endpoints" and "WebRTC implementation". If we interpret "endpoint" that way, and maybe do some editorial update of -rtp-usage replacing "WebRTC endpoint" with "WebRTC implementation" or "WebRTC-aware endpoint" (I can volunteer to write a proposal), there will be no clash with the proposal to drop "WebRTC endpoint" from -overview.

"WebRTC implementation" is defined nowhere, so I'm happy to have that 
term removed from service.

>
> The more neutral "device" for WebRTC entities allows to include entities that may not be the original source or final sink of RTP media streams. "Entity" would be an alternative for "Device" but I seem to recall there was dislike for that term previously.

I would be happy to exclude devices that are participating in, but not 
terminating, an RTP session (such as RTP translators and other 
quasi-multicast devices) from consideration in this set of documents.

I think we already have strong advice against them:

   "This limitation means that
    some of the RTP middlebox-based topologies are not suitable for use
    in the WebRTC environment.  Specifically:

    o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
       since they make the use of RTCP for congestion control and quality
       of service reports problematic (see Section 3.8 of
       [I-D.ietf-avtcore-rtp-topologies-update]).

    o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
       SHOULD NOT be used because its safe use requires a congestion
       control algorithm or RTP circuit breaker that handles point to
       multipoint, which has not yet been standardised.

"

If we were to start talking about them, this is a topic of far more 
wide-ranging debate than mere terminology.

>
> Kind regards,
> Uwe
>
>
>> -----Original Message-----
>> From: ext Schwarz, Albrecht (Albrecht)
>> [mailto:albrecht.schwarz@alcatel-lucent.com]
>> Sent: Wednesday, October 15, 2014 12:38 PM
>> To: Harald Alvestrand; Rauschenbach, Uwe (NSN - DE/Munich); ext
>> Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>> Subject: RE: [rtcweb] Definitions of WebRTC entities
>>
>> I'm also reluctant using "endpoint" as qualifier for WebRTC entities
>> because in communication networks the notion of endpoint is
>> fundamentally tight to a protocol (layer) (see e.g. (N)-connection-
>> endpoint in ITU-T X.200).
>> Now, "WebRTC" is not a single protocol, rather a suite of protocols in
>> concert.
>> E.g., we got
>> (DTLS)-connection-endpoints,
>> (SCTP)-connection-endpoints,
>> (UDP)-connection-endpoints,
>> (RTP)-connection-endpoints,
>> etc
>> within an WebRTC entity.
>>
>> Two cents from my side, of course you may always expand these OSI-ISO
>> base terms towards an (WebRTC)-endpoint (by highlighting the service
>> access point aspects and the "(WebRTC)-connection" concept).
>>
>> -Albrecht
>>
>>
>>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>> Alvestrand
>> Sent: Mittwoch, 15. Oktober 2014 11:40
>> To: Rauschenbach, Uwe (NSN - DE/Munich); ext Parthasarathi R; 'Christer
>> Holmberg'; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>
>> On 10/15/2014 10:47 AM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
>>> Hi Harald,
>>>
>>> I wonder whether we can simplify the set of terms by removing "WebRTC
>> endpoint".
>>
>> The reason for adding it was that it was already used in the RTCWEB RTP
>> document, in harmony with the usage of "endpoint" in other RTP-releated
>> specs.
>>
>> I'll take others' input on whether it's worth removing the term or
>> keeping it.
>>
>> Note - from a strictly formalistic point of view, replacing all
>> occurences of "WebRTC device" with "WebRTC endpoint" would have exactly
>> the same effect. But the terms feel more comfortable to use in
>> different contexts.
>>
>>> What was the reason for introducing the term "WebRTC endpoint"?
>> Wouldn't it be enough to have "WebRTC UA", "WebRTC device" and "WebRTC-
>> compatible device" (plus WebRTC gateway as special widespread case of
>> compatible device)?
>>>
>>> A slight change in the chain of terms in -overview-12 could reduce
>> the number of definitions without losing meaning (I think):
>>>
>>> 1) A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser)
>> is something that conforms to both the protocol specification and the
>> Javascript API defined above. A WebRTC User Agent conforms to both the
>> protocol specification and the Javascript API.
>>> --> keep as is
>>>
>>> 2) A WebRTC device is something that conforms to the protocol
>> specification, but does not claim to implement the Javascript API.
>>> --> replace "claim" by "have". This results in <WebRTC UA> IS-A
>> <WebRTC device> and we can pull out "WebRTC endpoint". (This is also
>> better in line than the current chain with the statement in the draft
>> "All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a
>> WebRTC device also applies to a WebRTC browser"
>>> 3) A WebRTC endpoint is either a WebRTC User Agent or a WebRTC
>> device.
>>> --> delete
>>>
>>> 4) A WebRTC-compatible endpoint is an endpoint that is capable of
>> successfully communicating with a WebRTC endpoint, but may fail to meet
>> some requirements of a WebRTC endpoint.  This may limit where in the
>> network such an endpoint can be attached, or may limit the security
>> guarantees that it offers to others.
>>> --> replace "endpoint" by "device"
>>>
>>> 5) A WebRTC gateway is a WebRTC-compatible endpoint that mediates
>> traffic to non-WebRTC entities.
>>> --> replace "endpoint" by "device"
>>>
>>> Kind regards,
>>> Uwe
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext
>>>> Parthasarathi R
>>>> Sent: Wednesday, October 15, 2014 2:30 AM
>>>> To: 'Harald Alvestrand'; 'Christer Holmberg'; rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>>
>>>> Hi Harald,
>>>>
>>>> <snip>
>>>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>>>> What do you mean by "middle box"? Again, that term is slippery.
>>>>> <Partha> I intent to say that the entity which is between two
>>>>> endpoints and it does not end any media session itself. Here, The
>>>>> confusion is that WebRTC compatible endpoint which is not an
>>>>> endpoint but it is a middle box. </Partha>
>>>> Seems that this entity (whatever it's called) isn't an endpoint at
>>>> all, so defining terms for endpoints shouldn't be relevant to
>>>> whatever this device is.
>>>>
>>>> There's always more boxes in the middle..... although as long as
>> they
>>>> don't have the DTLS keys, it's limited what they can do to the
>> packets.
>>>> <snip>
>>>>
>>>> Could you please update the terminology as "WebRTC compatible
>> device"
>>>> instead of WebRTC compatible endpoint as the entity is not required
>>>> to be endpoint.
>>>>
>>>> Thanks
>>>> Partha.
>>>>
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>>>> Sent: Thursday, October 09, 2014 12:29 PM
>>>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>>>
>>>>> On 10/08/2014 06:39 PM, Parthasarathi R wrote:
>>>>>> Hi Harald,
>>>>>>
>>>>>> Please read inline.
>>>>>>
>>>>>> Thanks
>>>>>> Partha
>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>>>>>>> Sent: Tuesday, October 07, 2014 10:16 PM
>>>>>>> To: Parthasarathi R; 'Christer Holmberg'; rtcweb@ietf.org
>>>>>>> Subject: Re: [rtcweb] Definitions of WebRTC entities
>>>>>>>
>>>>>>> On 10/07/2014 06:21 PM, Parthasarathi R wrote:
>>>>>>>> Hi Christer,
>>>>>>>>
>>>>>>>> I have no issue with WebRTC User Agent, WebRTC device, WebRTC
>>>>>>> endpoint.
>>>>>>>> I have bit trouble with WebRTC compatible endpoint as a entity
>>>> name
>>>>>>> as
>>>>>>>> 1) It may pass SRTP/data channel
>>>>>>> What do you mean by "pass"? That's a slippery term.
>>>>>> <Partha> "relay" will be more appropriate term as mentioned in Sec
>>>> 5
>>>>> Para 2 of draft-alvestrand-rtcweb-gateways. </Partha>
>>>>>>>> 2) It is not required to be endpoint but it shall be middle box.
>>>>>>> What do you mean by "middle box"? Again, that term is slippery.
>>>>>> <Partha> I intent to say that the entity which is between two
>>>>> endpoints and it does not end any media session itself. Here, The
>>>>> confusion is that WebRTC compatible endpoint which is not an
>>>>> endpoint but it is a middle box. </Partha>
>>>>>
>>>>> Seems that this entity (whatever it's called) isn't an endpoint at
>>>> all,
>>>>> so defining terms for endpoints shouldn't be relevant to whatever
>>>> this
>>>>> device is.
>>>>>
>>>>> There's always more boxes in the middle..... although as long as
>>>>> they don't have the DTLS keys, it's limited what they can do to the
>>>> packets.
>>>>>
>>>>>>>> WebRTC gateway looks more appropriate entity name in those
>>>>> scenarios.
>>>>>>> As written in my proposal, a WebRTC gateway is a WebRTC
>> compatible
>>>>>>> endpoint.
>>>>>>>
>>>>>> <Partha> As per your proposal, we need to define WebRTC compatible
>>>>> endpoint first which is super set of WebRTC gateway. Then, we need
>>>>> to clarify which kind of WebRTC compatible endpoint qualify as
>>>>> WebRTC gateway. But Christer wishes to have only two definition
>>>> (Full/Subset).
>>>>> </Partha>
>>>>>
>>>>> And I don't agree with Christer, so then we're two :-)
>>>> _______________________________________________
>>>> 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] Plan for MTI video codec?
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--089e010d7fb826414f05057a75bb
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Are we actually going to spend time discussing draft-ietf-rtcweb-video
(which will provide the details on what is necessary for video
implementations of any codec), or will yet another MTI discussion take up
all the energy (and perhaps time)?


On Wed, Oct 15, 2014 at 10:10 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Hi Victor,
>
> As you may remember, we tabled discussion of video codecs back in January=
,
> and said that the topic could be re-opened 6 weeks prior to IETF 91.  We =
do
> anticipate discussing it at this meeting, likely in the Thursday slot
> (since there is a related BoF earlier that day).
>
> regards,
>
> Ted Hardie
>
> On Wed, Oct 15, 2014 at 7:20 AM, Victor Pascual Avila <
> victor.pascual.avila@gmail.com> wrote:
>
>> I surely missed a recent thread on it but, any plan to discuss MTI
>> Video Codec in HI?
>>
>> I read in Jonathan's post it's part of the agenda?
>> http://blogs.cisco.com/collaboration/ciscos-openh264-now-part-of-firefox=
/
>>
>> Cheers,
>> -Victor
>>
>> On Wed, Sep 11, 2013 at 7:51 PM, Victor Pascual Avila
>> <victor.pascual.avila@gmail.com> wrote:
>> > Thanks for the feedback Ted!
>> >
>> > On Wed, Sep 11, 2013 at 7:48 PM, Ted Hardie <ted.ietf@gmail.com> wrote=
:
>> >> Hi Victor,
>> >>
>> >> The chairs have been discussion the structure of the upcoming meeting=
,
>> and
>> >> you should expect something from us soon.
>> >>
>> >> thanks,
>> >>
>> >> Ted
>> >>
>> >>
>> >> On Tue, Sep 10, 2013 at 6:47 AM, Victor Pascual Avila
>> >> <victor.pascual.avila@gmail.com> wrote:
>> >>>
>> >>> Dear WG Chairs,
>> >>>
>> >>> What's the plan for the MTI video codec discussion? Are we planning =
to
>> >>> discuss this during our next meeting? Is there a reason to delay thi=
s
>> >>> decision?
>> >>>
>> >>> My apologies if this has already been discussed but could not find i=
t
>> >>> in the archives.
>> >>>
>> >>> Thanks,
>> >>> -Victor
>> >>> _______________________________________________
>> >>> rtcweb mailing list
>> >>> rtcweb@ietf.org
>> >>> https://www.ietf.org/mailman/listinfo/rtcweb
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Victor Pascual =C3=81vila
>>
>>
>>
>> --
>> Victor Pascual =C3=81vila
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Are we actually going to spend time discussing draft-ietf-=
rtcweb-video (which will provide the details on what is necessary for video=
 implementations of any codec), or will yet another MTI discussion take up =
all the energy (and perhaps time)?=C2=A0<div><div><br></div></div></div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Oct 15, 2014=
 at 10:10 AM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@g=
mail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_default" s=
tyle=3D"font-family:tahoma,sans-serif;font-size:small">Hi Victor,<br><br></=
div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;fon=
t-size:small">As you may remember, we tabled discussion of video codecs bac=
k in January, and said that the topic could be re-opened 6 weeks prior to I=
ETF 91.=C2=A0 We do anticipate discussing it at this meeting, likely in the=
 Thursday slot (since there is a related BoF earlier that day).<br><br></di=
v><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-=
size:small">regards,<br><br>Ted Hardie<br></div></div><div class=3D"HOEnZb"=
><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Wed, Oct 15, 2014 at 7:20 AM, Victor Pascual Avila <span dir=3D"ltr">&=
lt;<a href=3D"mailto:victor.pascual.avila@gmail.com" target=3D"_blank">vict=
or.pascual.avila@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">I surely missed a recent thread on it but, any plan to discuss MTI<=
br>
Video Codec in HI?<br>
<br>
I read in Jonathan&#39;s post it&#39;s part of the agenda?<br>
<a href=3D"http://blogs.cisco.com/collaboration/ciscos-openh264-now-part-of=
-firefox/" target=3D"_blank">http://blogs.cisco.com/collaboration/ciscos-op=
enh264-now-part-of-firefox/</a><br>
<br>
Cheers,<br>
-Victor<br>
<br>
On Wed, Sep 11, 2013 at 7:51 PM, Victor Pascual Avila<br>
<span>&lt;<a href=3D"mailto:victor.pascual.avila@gmail.com" target=3D"_blan=
k">victor.pascual.avila@gmail.com</a>&gt; wrote:<br>
&gt; Thanks for the feedback Ted!<br>
&gt;<br>
&gt; On Wed, Sep 11, 2013 at 7:48 PM, Ted Hardie &lt;<a href=3D"mailto:ted.=
ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt;&gt; Hi Victor,<br>
&gt;&gt;<br>
&gt;&gt; The chairs have been discussion the structure of the upcoming meet=
ing, and<br>
&gt;&gt; you should expect something from us soon.<br>
&gt;&gt;<br>
&gt;&gt; thanks,<br>
&gt;&gt;<br>
&gt;&gt; Ted<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Tue, Sep 10, 2013 at 6:47 AM, Victor Pascual Avila<br>
&gt;&gt; &lt;<a href=3D"mailto:victor.pascual.avila@gmail.com" target=3D"_b=
lank">victor.pascual.avila@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Dear WG Chairs,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What&#39;s the plan for the MTI video codec discussion? Are we=
 planning to<br>
&gt;&gt;&gt; discuss this during our next meeting? Is there a reason to del=
ay this<br>
&gt;&gt;&gt; decision?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; My apologies if this has already been discussed but could not =
find it<br>
&gt;&gt;&gt; in the archives.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Thanks,<br>
&gt;&gt;&gt; -Victor<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" targe=
t=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Victor Pascual =C3=81vila<br>
<br>
<br>
<br>
</span><span><font color=3D"#888888">--<br>
Victor Pascual =C3=81vila<br>
</font></span><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>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--089e010d7fb826414f05057a75bb--


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From: Colin Perkins <csp@csperkins.org>
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On 15 Oct 2014, at 18:26, Harald Alvestrand <harald@alvestrand.no> =
wrote:
> On 10/15/2014 06:55 PM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
>> Hi Harald, all,
>>=20
>> I concur with Albrecht w.r.t. the meaning of "WebRTC endpoint". You =
mention below that "endpoint" is included mainly because it is used in =
the rtp-usage draft. But an "RTP endpoint" (a source and/or destination =
of RTP streams) may be different from a "WebRTC endpoint" (e.g. a mixing =
device).
> To be more precise: The term "WebRTC endpoint" is used in the =
rtp-usage draft. Three times.
>>=20
>> I believe that in the RTP usage draft, the meaning of "endpoint" is =
usually "RTP endpoint" with the interpretation "RTP endpoint that =
supports WebRTC". Terms used as well are "WebRTC implementations of RTP =
endpoints" and "WebRTC implementation". If we interpret "endpoint" that =
way, and maybe do some editorial update of -rtp-usage replacing "WebRTC =
endpoint" with "WebRTC implementation" or "WebRTC-aware endpoint" (I can =
volunteer to write a proposal), there will be no clash with the proposal =
to drop "WebRTC endpoint" from -overview.
>=20
> =93WebRTC implementation" is defined nowhere, so I'm happy to have =
that term removed from service.

Once we decide on the desired terms, we can align the RTP usage draft =
with them.=20

>> The more neutral "device" for WebRTC entities allows to include =
entities that may not be the original source or final sink of RTP media =
streams. "Entity" would be an alternative for "Device" but I seem to =
recall there was dislike for that term previously.
>=20
> I would be happy to exclude devices that are participating in, but not =
terminating, an RTP session (such as RTP translators and other =
quasi-multicast devices) from consideration in this set of documents.
>=20
> I think we already have strong advice against them:
>=20
>  "This limitation means that
>   some of the RTP middlebox-based topologies are not suitable for use
>   in the WebRTC environment.  Specifically:
>=20
>   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>      since they make the use of RTCP for congestion control and =
quality
>      of service reports problematic (see Section 3.8 of
>      [I-D.ietf-avtcore-rtp-topologies-update]).
>=20
>   o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>      SHOULD NOT be used because its safe use requires a congestion
>      control algorithm or RTP circuit breaker that handles point to
>      multipoint, which has not yet been standardised.
>=20
> "
>=20
> If we were to start talking about them, this is a topic of far more =
wide-ranging debate than mere terminology.

There are allowed topologies with RTP mixers and translators where a =
device participates in, but does not terminate, an RTP session, and the =
RTP usage draft needs to refer to such middleboxes in places. I=92m not =
sure we need a WebRTC-specific term for such a middlebox for the RTP =
usage draft, though.


--=20
Colin Perkins
https://csperkins.org/





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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Sean Turner <turners@ieca.com>,  Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] Agenda for the upcoming meeting
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During this morning chairs call, we discussed the agenda for the upcoming
meeting.  The two big items are JSEP and the video codec discussion, and
our current thought is to have the JSEP discussion on Monday and the video
codec discussion on Thursday.  There are a number of items we may need to
discuss; if there are particular issues you feel we need to cover, please
send a message to the list with the topic and the amount of time you feel
you need.

thanks,

Ted, Cullen, Sean

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">During this morning chairs call, we discussed th=
e agenda for the upcoming meeting.=C2=A0 The two big items are JSEP and the=
 video codec discussion, and our current thought is to have the JSEP discus=
sion on Monday and the video codec discussion on Thursday.=C2=A0 There are =
a number of items we may need to discuss; if there are particular issues yo=
u feel we need to cover, please send a message to the list with the topic a=
nd the amount of time you feel you need.<br><br>thanks,<br><br>Ted, Cullen,=
 Sean<br></div></div>

--bcaec5196bdd8ea2d605058d543e--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "Sean Turner" <turners@ieca.com>, Cullen Jennings <fluffy@cisco.com>
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From nobody Thu Oct 16 14:17:23 2014
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On 10/15/2014 10:10 AM, Ted Hardie wrote:
> Hi Victor,
>
> As you may remember, we tabled discussion of video codecs back in 
> January, and said that the topic could be re-opened 6 weeks prior to 
> IETF 91.  We do anticipate discussing it at this meeting, likely in 
> the Thursday slot (since there is a related BoF earlier that day).
>

And that's because something substantive has changed, or simply because 
wasting the WG time on this again is more entertaining than actually 
finishing a specification that can be independently implemented by all 
browser vendors? (A specification that we are nowhere near having, as 
far as I can tell)

Matthew Kaufman



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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 10/15/2014 10:10 AM, Ted Hardie
      wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMAZufR7gUrwkS7Tf5GOfg+ZtsZWGcn-8YLCvnmYnTgfFw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:tahoma,sans-serif;font-size:small">Hi
          Victor,<br>
          <br>
        </div>
        <div class="gmail_default"
          style="font-family:tahoma,sans-serif;font-size:small">As you
          may remember, we tabled discussion of video codecs back in
          January, and said that the topic could be re-opened 6 weeks
          prior to IETF 91.&nbsp; We do anticipate discussing it at this
          meeting, likely in the Thursday slot (since there is a related
          BoF earlier that day).<br>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
    And that's because something substantive has changed, or simply
    because wasting the WG time on this again is more entertaining than
    actually finishing a specification that can be independently
    implemented by all browser vendors? (A specification that we are
    nowhere near having, as far as I can tell)<br>
    <br>
    Matthew Kaufman<br>
    <br>
    <br>
  </body>
</html>

--------------050307000004080905020408--


From nobody Thu Oct 16 14:28:22 2014
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On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far as
> I can tell)

Personally, I've found the reprieve from this fight refreshing.  And
it would appear that we've made some real progress as a result.  I'd
suggest that if we don't have new information, we continue to spend
our time productively.  If we can't find topics to occupy our meeting
agenda time, then maybe we can free an agenda slot.


From nobody Thu Oct 16 14:35:50 2014
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On 10/16/14, 3:28 PM, Martin Thomson wrote:
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>> And that's because something substantive has changed, or simply because
>> wasting the WG time on this again is more entertaining than actually
>> finishing a specification that can be independently implemented by all
>> browser vendors? (A specification that we are nowhere near having, as far as
>> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.

+1

As I said here on January 28th:

"IMHO the state of video codec development needs to evolve before the WG 
can make a decision regarding an MTI video codec."

As far as I can see, it is still too early to take up the MTI video 
codec issue again, and I urge the WG to continue working on tractable 
problems.

Peter

-- 
Peter Saint-Andre
https://andyet.com/


From nobody Thu Oct 16 15:02:40 2014
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Thu, 16 Oct 2014 15:02:30 -0700
To: Martin Thomson <martin.thomson@gmail.com>
Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/bvW7qk0MTrUmwWlJLSsBatFrpnU
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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One thing we could do instead of wasting time on MTI is to actually make pro=
gress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actually=
 interoperate regardless of the codec.



> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com> wro=
te:
>=20
>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>> And that's because something substantive has changed, or simply because
>> wasting the WG time on this again is more entertaining than actually
>> finishing a specification that can be independently implemented by all
>> browser vendors? (A specification that we are nowhere near having, as far=
 as
>> I can tell)
>=20
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Oct 16 15:41:16 2014
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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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I completely agree.  MTI is wasting time, given we know the industry is spl=
it between H.xxx and VPxxx ... and given we know that H.264/VP8 will become=
 H.265/VP9 rapidly ... we need to ensure WebRTC works with various codecs a=
nd guide the industry towards VP8, H.264, VP9, H.265 (not some other varian=
t).  There will be some vendors who support both formats and some who don't=
.
-chris=20

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Bernard Aboba
Sent: Thursday, October 16, 2014 3:03 PM
To: Martin Thomson
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Plan for MTI video codec?

One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.



> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com> wr=
ote:
>=20
>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>> And that's because something substantive has changed, or simply=20
>> because wasting the WG time on this again is more entertaining than=20
>> actually finishing a specification that can be independently=20
>> implemented by all browser vendors? (A specification that we are=20
>> nowhere near having, as far as I can tell)
>=20
> Personally, I've found the reprieve from this fight refreshing.  And=20
> it would appear that we've made some real progress as a result.  I'd=20
> suggest that if we don't have new information, we continue to spend=20
> our time productively.  If we can't find topics to occupy our meeting=20
> agenda time, then maybe we can free an agenda slot.
>=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


From nobody Thu Oct 16 15:51:18 2014
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+ 1

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cavigioli, Chr=
is
> Sent: Thursday, October 16, 2014 5:41 PM
> To: Bernard Aboba; Martin Thomson
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Plan for MTI video codec?
>=20
> I completely agree.  MTI is wasting time, given we know the industry is
> split between H.xxx and VPxxx ... and given we know that H.264/VP8 will
> become H.265/VP9 rapidly ... we need to ensure WebRTC works with various
> codecs and guide the industry towards VP8, H.264, VP9, H.265 (not some ot=
her
> variant).  There will be some vendors who support both formats and some w=
ho
> don't.
> -chris
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Bernard Aboba
> Sent: Thursday, October 16, 2014 3:03 PM
> To: Martin Thomson
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Plan for MTI video codec?
>=20
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>=20
>=20
>=20
> > On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
> >
> >> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> >> And that's because something substantive has changed, or simply
> >> because wasting the WG time on this again is more entertaining than
> >> actually finishing a specification that can be independently
> >> implemented by all browser vendors? (A specification that we are
> >> nowhere near having, as far as I can tell)
> >
> > Personally, I've found the reprieve from this fight refreshing.  And
> > it would appear that we've made some real progress as a result.  I'd
> > suggest that if we don't have new information, we continue to spend
> > our time productively.  If we can't find topics to occupy our meeting
> > agenda time, then maybe we can free an agenda slot.
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Oct 16 15:54:16 2014
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Date: Thu, 16 Oct 2014 18:54:08 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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+1 as well. There is nothing of consequence that changed since this
discussion happened last time, so there is no possibility for the decision.

_____________
Roman Shpount

On Thu, Oct 16, 2014 at 6:50 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> + 1
>
> > -----Original Message-----
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cavigioli,
> Chris
> > Sent: Thursday, October 16, 2014 5:41 PM
> > To: Bernard Aboba; Martin Thomson
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Plan for MTI video codec?
> >
> > I completely agree.  MTI is wasting time, given we know the industry is
> > split between H.xxx and VPxxx ... and given we know that H.264/VP8 will
> > become H.265/VP9 rapidly ... we need to ensure WebRTC works with various
> > codecs and guide the industry towards VP8, H.264, VP9, H.265 (not some
> other
> > variant).  There will be some vendors who support both formats and some
> who
> > don't.
> > -chris
> >
> > -----Original Message-----
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Bernard Aboba
> > Sent: Thursday, October 16, 2014 3:03 PM
> > To: Martin Thomson
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Plan for MTI video codec?
> >
> > One thing we could do instead of wasting time on MTI is to actually make
> > progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> > actually interoperate regardless of the codec.
> >
> >
> >
> > > On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> > wrote:
> > >
> > >> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> > >> And that's because something substantive has changed, or simply
> > >> because wasting the WG time on this again is more entertaining than
> > >> actually finishing a specification that can be independently
> > >> implemented by all browser vendors? (A specification that we are
> > >> nowhere near having, as far as I can tell)
> > >
> > > Personally, I've found the reprieve from this fight refreshing.  And
> > > it would appear that we've made some real progress as a result.  I'd
> > > suggest that if we don't have new information, we continue to spend
> > > our time productively.  If we can't find topics to occupy our meeting
> > > agenda time, then maybe we can free an agenda slot.
> > >
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">+1 as well. There is nothing of consequence that changed s=
ince this discussion happened last time, so there is no possibility for the=
 decision.</div><div class=3D"gmail_extra"><br clear=3D"all"><div>_________=
____<br>Roman Shpount</div>
<br><div class=3D"gmail_quote">On Thu, Oct 16, 2014 at 6:50 PM, Makaraju, M=
aridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alc=
atel-lucent.com" target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex">+ 1<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb=
-bounces@ietf.org</a>] On Behalf Of Cavigioli, Chris<br>
&gt; Sent: Thursday, October 16, 2014 5:41 PM<br>
&gt; To: Bernard Aboba; Martin Thomson<br>
&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Plan for MTI video codec?<br>
&gt;<br>
&gt; I completely agree.=C2=A0 MTI is wasting time, given we know the indus=
try is<br>
&gt; split between H.xxx and VPxxx ... and given we know that H.264/VP8 wil=
l<br>
&gt; become H.265/VP9 rapidly ... we need to ensure WebRTC works with vario=
us<br>
&gt; codecs and guide the industry towards VP8, H.264, VP9, H.265 (not some=
 other<br>
&gt; variant).=C2=A0 There will be some vendors who support both formats an=
d some who<br>
&gt; don&#39;t.<br>
&gt; -chris<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb=
-bounces@ietf.org</a>] On Behalf Of Bernard Aboba<br>
&gt; Sent: Thursday, October 16, 2014 3:03 PM<br>
&gt; To: Martin Thomson<br>
&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Plan for MTI video codec?<br>
&gt;<br>
&gt; One thing we could do instead of wasting time on MTI is to actually ma=
ke<br>
&gt; progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could=
<br>
&gt; actually interoperate regardless of the codec.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &gt; On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto=
:martin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt;<br>
&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt;&gt; On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mail=
to:matthew@matthew.at">matthew@matthew.at</a>&gt; wrote:<br>
&gt; &gt;&gt; And that&#39;s because something substantive has changed, or =
simply<br>
&gt; &gt;&gt; because wasting the WG time on this again is more entertainin=
g than<br>
&gt; &gt;&gt; actually finishing a specification that can be independently<=
br>
&gt; &gt;&gt; implemented by all browser vendors? (A specification that we =
are<br>
&gt; &gt;&gt; nowhere near having, as far as I can tell)<br>
&gt; &gt;<br>
&gt; &gt; Personally, I&#39;ve found the reprieve from this fight refreshin=
g.=C2=A0 And<br>
&gt; &gt; it would appear that we&#39;ve made some real progress as a resul=
t.=C2=A0 I&#39;d<br>
&gt; &gt; suggest that if we don&#39;t have new information, we continue to=
 spend<br>
&gt; &gt; our time productively.=C2=A0 If we can&#39;t find topics to occup=
y our meeting<br>
&gt; &gt; agenda time, then maybe we can free an agenda slot.<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&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>
_______________________________________________<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>

--001a11339a6446161c05059220e2--


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On 10/14/2014 3:33 AM, Harald Alvestrand wrote:
> On 10/13/2014 07:02 PM, Wyss, Felix wrote:
>
>> They would only see it as a peer if it were a full B2B user agent.  As an example, in VoIP the endpoints do not consider an SBC as a peer--yet it sits in the middle and may inspect the media passing through it.
>>
>> With respect to DCEP: If we already define a protocol for establishing the channels in-band, why not make ID selection and conflict resolution an explicit part of it?  That certainly would beat the IMHO rather distasteful DTLS role hack.
> When rethinking this, I've come to the conclusion that my original
> conclusion (that there couldn't possibly be a point in preserving
> channel numbers across a relay) was probably wrong. Please consider that
> withdrawn.
>
> I now see a tradeoff between allowing such non-mapping boxes (operating
> below the SCTP layer but above the DTLS layer in our model) and avoiding
> additional complexity (either extending DCEP with glare resolution -
> something it was designed to avoid needing - or requiring such boxes to
> not use actpass).

I'll note that not only does it have to sit between the SCTP and DTLS 
layers, but also it has to be the offerer to both sides as well.  While 
certainly there are some use-cases for that, it's by no means the common 
case.  WebRTC applications designed to work with such a midpoint may 
need to fall back on "external" negotiation, perhaps running a short 
sequence at opening time to decide who will get even and odd (assuming 
that both sides actually open channels dynamically).

It means you may not be able to drop such a middlebox (and have it 
initiate offers to both sides) without a cooperative application (or one 
known to have a compatible usage patter) - but likely you need some 
level of known compatibility anyways, or more likely the entity 
deploying the middlebox also deploys or chooses the application - 
especially as we've explicitly said that signaling protocols and 
transmission (i.e. sending the offer, etc) is explicitly not defined by 
the WG, so the middlebox, in order to generate the offers to both sides 
*must* be in bed somehow with the applications - at least to where they 
all agree on signaling.

> I agree that the DTLS hack is ugly. But glare resolution is ugly too.

Quite so.  We had glare resolution in earlier versions. It sucked and 
caused lots of problems, especially with 0-rtt opens.

While DTLS role isn't wonderful, neither was anything else proposed.  
Offerer/answerer status has the same problem.  An SDP parameter would 
allow a middlebox to specify to the endpoints compatible (complementary) 
values; however from a DataChannels perspective, this is almost the same 
as external negotiation if the middlebox and applications are 'matched'.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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To: Roman Shpount <roman@telurix.com>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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What I am seeing is a better awareness of APIs for accessing embedded codec=
 hardware in the client, and independently of any MTI codec discussion, I t=
hink the codec document should address the existing of these, and how they =
might be used. At least it should not ignore the issue entirely.

But I do agree that I do not really want an MTI codec discussion (unless we=
 find a late night bar to do it in).

Keith

________________________________
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Roman Shpount
Sent: 16 October 2014 23:54
To: Makaraju, Maridi Raju (Raju)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Plan for MTI video codec?

+1 as well. There is nothing of consequence that changed since this discuss=
ion happened last time, so there is no possibility for the decision.

_____________
Roman Shpount

On Thu, Oct 16, 2014 at 6:50 PM, Makaraju, Maridi Raju (Raju) <Raju.Makaraj=
u@alcatel-lucent.com<mailto:Raju.Makaraju@alcatel-lucent.com>> wrote:
+ 1

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.o=
rg>] On Behalf Of Cavigioli, Chris
> Sent: Thursday, October 16, 2014 5:41 PM
> To: Bernard Aboba; Martin Thomson
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: Re: [rtcweb] Plan for MTI video codec?
>
> I completely agree.  MTI is wasting time, given we know the industry is
> split between H.xxx and VPxxx ... and given we know that H.264/VP8 will
> become H.265/VP9 rapidly ... we need to ensure WebRTC works with various
> codecs and guide the industry towards VP8, H.264, VP9, H.265 (not some ot=
her
> variant).  There will be some vendors who support both formats and some w=
ho
> don't.
> -chris
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.o=
rg>] On Behalf Of Bernard Aboba
> Sent: Thursday, October 16, 2014 3:03 PM
> To: Martin Thomson
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: Re: [rtcweb] Plan for MTI video codec?
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
>
> > On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com<m=
ailto:martin.thomson@gmail.com>>
> wrote:
> >
> >> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at<mailto:m=
atthew@matthew.at>> wrote:
> >> And that's because something substantive has changed, or simply
> >> because wasting the WG time on this again is more entertaining than
> >> actually finishing a specification that can be independently
> >> implemented by all browser vendors? (A specification that we are
> >> nowhere near having, as far as I can tell)
> >
> > Personally, I've found the reprieve from this fight refreshing.  And
> > it would appear that we've made some real progress as a result.  I'd
> > suggest that if we don't have new information, we continue to spend
> > our time productively.  If we can't find topics to occupy our meeting
> > agenda time, then maybe we can free an agenda slot.
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<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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>
<meta content=3D"MSHTML 6.00.2900.6550" name=3D"GENERATOR">
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<body>
<div dir=3D"ltr" align=3D"left"><font face=3D"Arial" color=3D"#0000ff" size=
=3D"2"><span class=3D"177122811-17102014">What I am seeing is a better awar=
eness of APIs for accessing embedded codec hardware in the client, and inde=
pendently of any MTI codec discussion, I think
 the codec document should address the existing of these, and how they migh=
t be used. At least it should not ignore the issue entirely.</span></font><=
/div>
<div dir=3D"ltr" align=3D"left"><font face=3D"Arial" color=3D"#0000ff" size=
=3D"2"><span class=3D"177122811-17102014"></span></font>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><font face=3D"Arial" color=3D"#0000ff" size=
=3D"2"><span class=3D"177122811-17102014">But I do agree that I do not real=
ly want an MTI codec discussion (unless we find a late night bar to do it i=
n).</span></font></div>
<div dir=3D"ltr" align=3D"left"><font face=3D"Arial" color=3D"#0000ff" size=
=3D"2"><span class=3D"177122811-17102014"></span></font>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><font face=3D"Arial" color=3D"#0000ff" size=
=3D"2"><span class=3D"177122811-17102014">Keith</span></font></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDE=
R-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<div class=3D"OutlookMessageHeader" lang=3D"en-us" dir=3D"ltr" align=3D"lef=
t">
<hr tabindex=3D"-1">
<font face=3D"Tahoma" size=3D"2"><b>From:</b> rtcweb [mailto:rtcweb-bounces=
@ietf.org]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> 16 October 2014 23:54<br>
<b>To:</b> Makaraju, Maridi Raju (Raju)<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</font><br>
</div>
<div></div>
<div dir=3D"ltr">&#43;1 as well. There is nothing of consequence that chang=
ed since this discussion happened last time, so there is no possibility for=
 the decision.</div>
<div class=3D"gmail_extra"><br clear=3D"all">
<div>_____________<br>
Roman Shpount</div>
<br>
<div class=3D"gmail_quote">On Thu, Oct 16, 2014 at 6:50 PM, Makaraju, Marid=
i Raju (Raju)
<span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" t=
arget=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
&#43; 1<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb=
-bounces@ietf.org</a>] On Behalf Of Cavigioli, Chris<br>
&gt; Sent: Thursday, October 16, 2014 5:41 PM<br>
&gt; To: Bernard Aboba; Martin Thomson<br>
&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Plan for MTI video codec?<br>
&gt;<br>
&gt; I completely agree.&nbsp; MTI is wasting time, given we know the indus=
try is<br>
&gt; split between H.xxx and VPxxx ... and given we know that H.264/VP8 wil=
l<br>
&gt; become H.265/VP9 rapidly ... we need to ensure WebRTC works with vario=
us<br>
&gt; codecs and guide the industry towards VP8, H.264, VP9, H.265 (not some=
 other<br>
&gt; variant).&nbsp; There will be some vendors who support both formats an=
d some who<br>
&gt; don't.<br>
&gt; -chris<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb=
-bounces@ietf.org</a>] On Behalf Of Bernard Aboba<br>
&gt; Sent: Thursday, October 16, 2014 3:03 PM<br>
&gt; To: Martin Thomson<br>
&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Plan for MTI video codec?<br>
&gt;<br>
&gt; One thing we could do instead of wasting time on MTI is to actually ma=
ke<br>
&gt; progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could=
<br>
&gt; actually interoperate regardless of the codec.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &gt; On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto=
:martin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt;<br>
&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt;&gt; On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mail=
to:matthew@matthew.at">matthew@matthew.at</a>&gt; wrote:<br>
&gt; &gt;&gt; And that's because something substantive has changed, or simp=
ly<br>
&gt; &gt;&gt; because wasting the WG time on this again is more entertainin=
g than<br>
&gt; &gt;&gt; actually finishing a specification that can be independently<=
br>
&gt; &gt;&gt; implemented by all browser vendors? (A specification that we =
are<br>
&gt; &gt;&gt; nowhere near having, as far as I can tell)<br>
&gt; &gt;<br>
&gt; &gt; Personally, I've found the reprieve from this fight refreshing.&n=
bsp; And<br>
&gt; &gt; it would appear that we've made some real progress as a result.&n=
bsp; I'd<br>
&gt; &gt; suggest that if we don't have new information, we continue to spe=
nd<br>
&gt; &gt; our time productively.&nbsp; If we can't find topics to occupy ou=
r meeting<br>
&gt; &gt; agenda time, then maybe we can free an agenda slot.<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&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>
_______________________________________________<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>
</blockquote>
</body>
</html>

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On 10/17/2014 12:02 AM, Bernard Aboba wrote:
> One thing we could do instead of wasting time on MTI is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actually interoperate regardless of the codec.

The big argument for an MTI is actually the one stated in -overview: It 
guards against interoperability failure.

I would like to have an ecosystem where one can field a box, connect it 
to everything else, and run well for *some* level of "well" - and I 
would prefer that ecosystem to be one where it's possible to field the 
box without making prior arrangements with anyone about licenses.

This argument hasn't changed one whit since last time we discussed it. 
And I don't see much movement on the specifics of the proposals either.

We'll have to interoperate well with the codecs we field. So using some 
time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 
isn't finished either. There's one sentence that needs to be removed.)

I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd 
prefer to re-discuss based on the knowledge that some significant 
players have actually changed their minds.

At the moment, I don't see signs that any of the poll respondents have 
said "I have changed my mind".

Harald


>
>
>
>> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com> wrote:
>>
>>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>>> And that's because something substantive has changed, or simply because
>>> wasting the WG time on this again is more entertaining than actually
>>> finishing a specification that can be independently implemented by all
>>> browser vendors? (A specification that we are nowhere near having, as far as
>>> I can tell)
>> Personally, I've found the reprieve from this fight refreshing.  And
>> it would appear that we've made some real progress as a result.  I'd
>> suggest that if we don't have new information, we continue to spend
>> our time productively.  If we can't find topics to occupy our meeting
>> agenda time, then maybe we can free an agenda slot.
>>
>> _______________________________________________
>> 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] Agenda for the upcoming meeting
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On 10/16/2014 07:10 PM, Ted Hardie wrote:
> During this morning chairs call, we discussed the agenda for the 
> upcoming meeting.  The two big items are JSEP and the video codec 
> discussion, and our current thought is to have the JSEP discussion on 
> Monday and the video codec discussion on Thursday.  There are a number 
> of items we may need to discuss; if there are particular issues you 
> feel we need to cover, please send a message to the list with the 
> topic and the amount of time you feel you need.

We have not had face to face discussion on 
draft-alvestrand-rtcweb-gateways (and not much discussion on the mailing 
list either).

I'd like to have a review of that document on the agenda.

Note: In my opinion, the chairs can choose to decide on whether the 
group should adopt the draft before, during or after the meeting, with 
or without calling for consensus of the WG.

I would like to have the chairs say what they intend to do about this 
document.

>
> thanks,
>
> Ted, Cullen, Sean
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 10/16/2014 07:10 PM, Ted Hardie
      wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMBQobeA_6aiMZffA6hHNkySK+CZptJU0XYzDyxGF4xE2A@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:tahoma,sans-serif;font-size:small">During
          this morning chairs call, we discussed the agenda for the
          upcoming meeting.  The two big items are JSEP and the video
          codec discussion, and our current thought is to have the JSEP
          discussion on Monday and the video codec discussion on
          Thursday.  There are a number of items we may need to discuss;
          if there are particular issues you feel we need to cover,
          please send a message to the list with the topic and the
          amount of time you feel you need.<br>
        </div>
      </div>
    </blockquote>
    <br>
    We have not had face to face discussion on
    draft-alvestrand-rtcweb-gateways (and not much discussion on the
    mailing list either).<br>
    <br>
    I'd like to have a review of that document on the agenda.<br>
    <br>
    Note: In my opinion, the chairs can choose to decide on whether the
    group should adopt the draft before, during or after the meeting,
    with or without calling for consensus of the WG.<br>
    <br>
    I would like to have the chairs say what they intend to do about
    this document.<br>
    <br>
    <blockquote
cite="mid:CA+9kkMBQobeA_6aiMZffA6hHNkySK+CZptJU0XYzDyxGF4xE2A@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:tahoma,sans-serif;font-size:small"><br>
          thanks,<br>
          <br>
          Ted, Cullen, Sean<br>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------080706010007090109000504--


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> > I now see a tradeoff between allowing such non-mapping boxes
> > (operating below the SCTP layer but above the DTLS layer in our model)
> > and avoiding additional complexity (either extending DCEP with glare
> > resolution - something it was designed to avoid needing - or requiring
> > such boxes to not use actpass).
>=20
> I'll note that not only does it have to sit between the SCTP and DTLS lay=
ers,
> but also it has to be the offerer to both sides as well.=20

This is a problem even if the middlebox is the answerer on one side and the=
 offerer on the other (relay).  As offerer it cannot control what role it w=
ill play without violating RFC#5763.  As Indicated in my response to Harald=
 from the 15th, I do not believe allowing the offerer to offer anything but=
 actpass to be a desirable approach. =20


> While certainly there are some use-cases for that, it's by no means the c=
ommon case. =20
> WebRTC applications designed to work with such a midpoint may need to fal=
l back on
> "external" negotiation, perhaps running a short sequence at opening time =
to
> decide who will get even and odd (assuming that both sides actually open
> channels dynamically).
> It means you may not be able to drop such a middlebox (and have it initia=
te
> offers to both sides) without a cooperative application (or one known to
> have a compatible usage patter) - but likely you need some level of known
> compatibility anyways, or more likely the entity deploying the middlebox =
also
> deploys or chooses the application - especially as we've explicitly said =
that
> signaling protocols and transmission (i.e. sending the offer, etc) is exp=
licitly
> not defined by the WG, so the middlebox, in order to generate the offers =
to
> both sides
> *must* be in bed somehow with the applications - at least to where they a=
ll
> agree on signaling.

Well, then why not leave the ID generation and conflict resolution complete=
ly up to the application?   That takes the DTLS role hack out of the pictur=
e too.  Of course, leaving it to the application how it determines the ID r=
isks that poorly written applications just pick it at random or some other =
criteria and don't worry about collisions ("well, it worked in my tests"). =
=20


> > I agree that the DTLS hack is ugly. But glare resolution is ugly too.
>=20
> Quite so.  We had glare resolution in earlier versions. It sucked and cau=
sed
> lots of problems, especially with 0-rtt opens.
>=20
> While DTLS role isn't wonderful, neither was anything else proposed.
> Offerer/answerer status has the same problem. =20

Even the offerer/answerer role would be better than the DTLS role.  At leas=
t that would work in the case where the middlebox relays the offer and answ=
er.  I think that's still too limiting as I'd rather allow the middlebox to=
 be offerer on both sides, but I'll take it over the DTLS hack any day :-)


> An SDP parameter would
> allow a middlebox to specify to the endpoints compatible (complementary)
> values; however from a DataChannels perspective, this is almost the same =
as
> external negotiation if the middlebox and applications are 'matched'.

If in-band negotiation and collision resolution is not an option, adding an=
 SDP attribute would be my second choice.  If it's not present, use offerer=
/answerer role.=20


Thanks,
--Felix


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From: Justin Uberti <juberti@google.com>
Date: Fri, 17 Oct 2014 15:49:02 -0700
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Subject: Re: [rtcweb] Agenda for the upcoming meeting
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--089e011849c40fc0170505a62de7
Content-Type: text/plain; charset=UTF-8

I would like to discuss enterprise WebRTC proxies, a previously unaddressed
requirement which now has a proposal in
https://tools.ietf.org/html/draft-schwartz-rtcweb-return-03

On Fri, Oct 17, 2014 at 6:25 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>  On 10/16/2014 07:10 PM, Ted Hardie wrote:
>
>  During this morning chairs call, we discussed the agenda for the
> upcoming meeting.  The two big items are JSEP and the video codec
> discussion, and our current thought is to have the JSEP discussion on
> Monday and the video codec discussion on Thursday.  There are a number of
> items we may need to discuss; if there are particular issues you feel we
> need to cover, please send a message to the list with the topic and the
> amount of time you feel you need.
>
>
> We have not had face to face discussion on
> draft-alvestrand-rtcweb-gateways (and not much discussion on the mailing
> list either).
>
> I'd like to have a review of that document on the agenda.
>
> Note: In my opinion, the chairs can choose to decide on whether the group
> should adopt the draft before, during or after the meeting, with or without
> calling for consensus of the WG.
>
> I would like to have the chairs say what they intend to do about this
> document.
>
>
> thanks,
>
> Ted, Cullen, Sean
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I would like to discuss enterprise WebRTC proxies, a previ=
ously unaddressed requirement which now has a proposal in<div><a href=3D"ht=
tps://tools.ietf.org/html/draft-schwartz-rtcweb-return-03">https://tools.ie=
tf.org/html/draft-schwartz-rtcweb-return-03</a><br></div></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Oct 17, 2014 at 6:2=
5 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alve=
strand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
    <div>On 10/16/2014 07:10 PM, Ted Hardie
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">During
          this morning chairs call, we discussed the agenda for the
          upcoming meeting.=C2=A0 The two big items are JSEP and the video
          codec discussion, and our current thought is to have the JSEP
          discussion on Monday and the video codec discussion on
          Thursday.=C2=A0 There are a number of items we may need to discus=
s;
          if there are particular issues you feel we need to cover,
          please send a message to the list with the topic and the
          amount of time you feel you need.<br>
        </div>
      </div>
    </blockquote>
    <br></span>
    We have not had face to face discussion on
    draft-alvestrand-rtcweb-gateways (and not much discussion on the
    mailing list either).<br>
    <br>
    I&#39;d like to have a review of that document on the agenda.<br>
    <br>
    Note: In my opinion, the chairs can choose to decide on whether the
    group should adopt the draft before, during or after the meeting,
    with or without calling for consensus of the WG.<br>
    <br>
    I would like to have the chairs say what they intend to do about
    this document.<br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small"><br>
          thanks,<br>
          <br>
          Ted, Cullen, Sean<br>
        </div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </div>

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

--089e011849c40fc0170505a62de7--


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On 10/17/2014 4:00 PM, Wyss, Felix wrote:
>>> I now see a tradeoff between allowing such non-mapping boxes
>>> (operating below the SCTP layer but above the DTLS layer in our model)
>>> and avoiding additional complexity (either extending DCEP with glare
>>> resolution - something it was designed to avoid needing - or requiring
>>> such boxes to not use actpass).
>> I'll note that not only does it have to sit between the SCTP and DTLS layers,
>> but also it has to be the offerer to both sides as well.


So if we use SDP, the issues become one of ensuring the two sides always 
have different values.

For direct calls, no problem (offerer sets mine=x, answerer sets mine=!x)
For direct intermediated calls, the same (offer is just forwarded)

For start-in-the-middle calls (3pcc) the middlebox sends offers to both 
sides, but can use complimentary values:

offer->A mine=x, offer->B mine=!x (makes assumptions about how A and B 
will answer! & likely demands use of TURN server or media gateway)
or
offer->A mine=x,  A->answer mine=!x,  "offer"->B mine=!x, B->answer 
mine=x (and then the middlebox will start a renegotiation if for some 
reason B's answer is incompatible with A's response).  May require some 
careful ICE candidate handling to avoid TURN.

Note: I hate this sort of setup in SDP; it almost guarantees issues when 
you don't have tightly-coupled endpoints and thus know how they'll 
respond to the SDP.

Note that in SIP-land-style you'd generally (always?) do it this way to 
avoid re-negotiation and avoid the tight coupling:

offer->A with no SDP; A->"answer" (really createOffer()), offer->B, 
B->answer, answer->A

(In SIP this would be OFFER->A (null SDP), A->200 OK(offer-SDP), 
OFFER->B(offer-SDP), B->200 OK (answer-SDP), ACK->A (answer-SDP), ACK->B 
(null)

Note here that both sides agree who offered and answered, unlike the 
above, AND I believe that DTLS role use would work.

> This is a problem even if the middlebox is the answerer on one side and the offerer on the other (relay).  As offerer it cannot control what role it will play without violating RFC#5763.  As Indicated in my response to Harald from the 15th, I do not believe allowing the offerer to offer anything but actpass to be a desirable approach.

Per above, if it's simply asking the application for an offer, then 
forwarding it, there's no issue: the offer is a true offer, you never 
have to "convert" an SDP offer into an answer or vice versa, and the 
offer will be actpass and roles will be consistent (assuming the 
middlebox isn't trying to intercept/sniff the decrypted DataChannel data).

So, if the middle box uses sip-style "empty offers" (ask the app to make 
an offer), then I think there are no issues with DTLS role.  If it tries 
to avoid the 3-way handshake style, or work with apps that don't 
understand it then there'd be an issue (but in webrtc signaling isn't 
specified, so the middlebox MUST be designed to be compatible with the 
signaling used by the apps).

Also, if the middlebox acts as an "answerer-in-the-middle", DTLS roles 
should work:
A->middlebox offer, B->middlebox offer, middlebox->A answer, 
middlebox->B answer (where the answers are derived from the offer on the 
other side).  This again requires that the two Offers be safely 
transformable by the middlebox into answers without a renegotiation. The 
middlebox would select the roles in the answers to be complimentary.

>> While certainly there are some use-cases for that, it's by no means the common case.
>> WebRTC applications designed to work with such a midpoint may need to fall back on
>> "external" negotiation, perhaps running a short sequence at opening time to
>> decide who will get even and odd (assuming that both sides actually open
>> channels dynamically).
>> It means you may not be able to drop such a middlebox (and have it initiate
>> offers to both sides) without a cooperative application (or one known to
>> have a compatible usage patter) - but likely you need some level of known
>> compatibility anyways, or more likely the entity deploying the middlebox also
>> deploys or chooses the application - especially as we've explicitly said that
>> signaling protocols and transmission (i.e. sending the offer, etc) is explicitly
>> not defined by the WG, so the middlebox, in order to generate the offers to
>> both sides
>> *must* be in bed somehow with the applications - at least to where they all
>> agree on signaling.
> Well, then why not leave the ID generation and conflict resolution completely up to the application?   That takes the DTLS role hack out of the picture too.  Of course, leaving it to the application how it determines the ID risks that poorly written applications just pick it at random or some other criteria and don't worry about collisions ("well, it worked in my tests").

As per above, an application can *always* do that, it never has to use 
the DTLS-role-reliant allocations.

Webrtc doesn't specify the signaling side; any middlebox must agree with 
the apps on signaling, and thus you can put requirements on how that 
interaction works.  Inserting a middlebox into existing application 
channels that were not designed for a middlebox at most requires you to 
allow the application to be told to generate an offer, OR for the 
application to use it's own generation/conflict resolution.

What is not possible is to insert a middlebox between arbitrary WebRTC 
apps that are not designed with that in mind, AND to try to initiate 
calls from the "middle" (without the SDP solution). However, initiation 
from the middle in that way is fraught with other issues and need to 
renegotiate after the first pass anyways.


>>> I agree that the DTLS hack is ugly. But glare resolution is ugly too.
>> Quite so.  We had glare resolution in earlier versions. It sucked and caused
>> lots of problems, especially with 0-rtt opens.
>>
>> While DTLS role isn't wonderful, neither was anything else proposed.
>> Offerer/answerer status has the same problem.
> Even the offerer/answerer role would be better than the DTLS role.  At least that would work in the case where the middlebox relays the offer and answer.  I think that's still too limiting as I'd rather allow the middlebox to be offerer on both sides, but I'll take it over the DTLS hack any day :-)

I disagree that role won't work if the middlebox is just forwarding 
offers and answers - that's what every WebRTC signaling server is doing 
today.   However, as I think you implied, you're also terminating DTLS 
but forwarding SCTP (and media).  This a) implies packet-relay; b) 
implicit decryption of all media at the server (NOTE this is a 
security/privacy/MITM issue).

Still, if A->middlebox (offer) is actpass, and middlebox->B (offer) is 
actpass (and they would be), then B chooses the role, and then when 
forwarding the answer to A the middlebox can choose the same role B 
did.  So I still don't see a problem here....

Also, assuming some bits of my analysis are incorrect, or there are edge 
cases we care about we don't handle (the non-SIP-like start-in-middle 
case), then (and only then) as a working group, we need to decide if 
this is a usecase we support.  Obviously B2BUA's (which this 
more-or-less is) are always possible, at varying levels of ease.  The 
question (partly) is should the protocol be designed to ease the use of 
a B2BUA with datachannels *assuming* it doesn't work per above.


>> An SDP parameter would
>> allow a middlebox to specify to the endpoints compatible (complementary)
>> values; however from a DataChannels perspective, this is almost the same as
>> external negotiation if the middlebox and applications are 'matched'.
> If in-band negotiation and collision resolution is not an option, adding an SDP attribute would be my second choice.  If it's not present, use offerer/answerer role.

If we *cannot* safely use DTLS role for middlebox cases, and care, then 
we should allow SDP to override the DTLS role (not use the SDP offer 
state as the base).  If an offerer doesn't specify even/odd in the SDP, 
but the answer does specify, then the offerer must use the SDP of the 
answerer to select even/odd (i.e. SDP overrides role).

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Subject: Re: [rtcweb] Plan for MTI video codec?
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+1 to not having MTI codec discussion unless some progress has been made on
VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there was
no change on the members position.

On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
>> One thing we could do instead of wasting time on MTI is to actually make
>> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>> actually interoperate regardless of the codec.
>>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it to
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. And
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>>
>>
>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>>> wrote:
>>>
>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>>>> And that's because something substantive has changed, or simply because
>>>> wasting the WG time on this again is more entertaining than actually
>>>> finishing a specification that can be independently implemented by all
>>>> browser vendors? (A specification that we are nowhere near having, as
>>>> far as
>>>> I can tell)
>>>>
>>> Personally, I've found the reprieve from this fight refreshing.  And
>>> it would appear that we've made some real progress as a result.  I'd
>>> suggest that if we don't have new information, we continue to spend
>>> our time productively.  If we can't find topics to occupy our meeting
>>> agenda time, then maybe we can free an agenda slot.
>>>
>>> _______________________________________________
>>> 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
>



-- 
Alex. Gouaillard, PhD, PhD, MBA
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<div dir=3D"ltr">+1 to not having MTI codec discussion unless some progress=
 has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing haral=
d&#39;s =C2=A0feeling that there was no change on the members position.=C2=
=A0</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, =
Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D=
"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</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"><span class=3D"">On 10/17=
/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span class=3D"HOEnZb"><font color=
=3D"#888888"><br>
<br>
Harald</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>

--001a113ad9f090fbc80505b8e028--


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Subject: [rtcweb] VP8 in ISO (Re:  Plan for MTI video codec?)
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Changing the subject line, since this is information that more people
may be interested in having called out clearly:

On 10/18/2014 11:08 PM, Alexandre GOUAILLARD wrote:
> +1 to not having MTI codec discussion unless some progress has been
> made on VP8 at MPEG. Any news on that?

The VCB spec (VP8 in ISO) is in DIS ballot, and the ballot closes in
January.
ISO processes are complex, and I don't always understand them, but if
the ballot results are positive, the standard should be approved.

So far, we have not seen any official feedback, but all the unofficial
feedback I've seen has been more or less editorial in nature.

There are no technical changes that would cause any issues in
compatibility between VP8 and VCB.


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--001a11c29d5e77256c0505bc9866
Content-Type: text/plain; charset=UTF-8

Thanks harald for the update.

I'm tempted to say that this is the only event I can think off, based on
the results of the previous pool, that would motivate some from changing
their mind. Hence putting the MTI discussion back on the table before any
progress on VP8 in IOS is, IMHO, useless.

Alex.

On Sun, Oct 19, 2014 at 5:25 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>
> Changing the subject line, since this is information that more people
> may be interested in having called out clearly:
>
> On 10/18/2014 11:08 PM, Alexandre GOUAILLARD wrote:
> > +1 to not having MTI codec discussion unless some progress has been
> > made on VP8 at MPEG. Any news on that?
>
> The VCB spec (VP8 in ISO) is in DIS ballot, and the ballot closes in
> January.
> ISO processes are complex, and I don't always understand them, but if
> the ballot results are positive, the standard should be approved.
>
> So far, we have not seen any official feedback, but all the unofficial
> feedback I've seen has been more or less editorial in nature.
>
> There are no technical changes that would cause any issues in
> compatibility between VP8 and VCB.
>
>


-- 
Alex. Gouaillard, PhD, PhD, MBA
------------------------------------------------------------------------------------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
------------------------------------------------------------------------------------
sg.linkedin.com/agouaillard

   -

--001a11c29d5e77256c0505bc9866
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<div dir=3D"ltr">Thanks harald for the update.<div><br></div><div>I&#39;m t=
empted to say that this is the only event I can think off, based on the res=
ults of the previous pool, that would motivate some from changing their min=
d. Hence putting the MTI discussion back on the table before any progress o=
n VP8 in IOS is, IMHO, useless.=C2=A0</div><div><br></div><div>Alex.</div><=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Oct =
19, 2014 at 5:25 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><br>
Changing the subject line, since this is information that more people<br>
may be interested in having called out clearly:<br>
<br>
On 10/18/2014 11:08 PM, Alexandre GOUAILLARD wrote:<br>
&gt; +1 to not having MTI codec discussion unless some progress has been<br=
>
&gt; made on VP8 at MPEG. Any news on that?<br>
<br>
The VCB spec (VP8 in ISO) is in DIS ballot, and the ballot closes in<br>
January.<br>
ISO processes are complex, and I don&#39;t always understand them, but if<b=
r>
the ballot results are positive, the standard should be approved.<br>
<br>
So far, we have not seen any official feedback, but all the unofficial<br>
feedback I&#39;ve seen has been more or less editorial in nature.<br>
<br>
There are no technical changes that would cause any issues in<br>
compatibility between VP8 and VCB.<br>
<br>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><div dir=3D"=
ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------------------=
--------------------------------------------------</div><div>CTO - Temasys =
Communications, S&#39;pore / Mountain View</div><div>President - CoSMo Soft=
ware, Cambridge, MA</div><div>---------------------------------------------=
---------------------------------------</div><div><a href=3D"http://sg.link=
edin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a></di=
v><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;outline:0px;f=
ont-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-align:baselin=
e;list-style:none;line-height:17px;display:table-cell;width:504px;color:rgb=
(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;out=
line:0px;font-style:inherit;font-size:11px;font-family:inherit;vertical-ali=
gn:baseline;font-variant:inherit;line-height:1.2em"><dl style=3D"margin:0px=
;padding:0px;border:0px;outline:0px;font-style:inherit;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:inherit;word-wrap:=
break-word"><br></dl></li></ul></div></div>
</div>

--001a11c29d5e77256c0505bc9866--


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From: Jonathan Rosenberg <jdrosen@jdrosen.net>
To: Alexandre GOUAILLARD <agouaillard@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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I'm in favor of taking another run at this.

The working group has repeatedly said that an MTI codec is something we
need to produce. And its one of the issues holding up wider adoption of the
technology (not the only one for sure).

And things have changed since the last meeting, a year ago now (November in
Vancouver). Cisco's open264 plugin shipped and now just recently is
integrated into Firefox. iOS8 shipped with APIs for H264. There are other
things worth noting. Will this change the minds of everyone? Surely not.
Will it sway enough for us to achieve rough consensus? Maybe. IMHO - worth
finding out.

On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <agouaillard@gmail.com
> wrote:

> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there
> was no change on the members position.
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>
>>> One thing we could do instead of wasting time on MTI is to actually make
>>> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>> actually interoperate regardless of the codec.
>>>
>>
>> The big argument for an MTI is actually the one stated in -overview: It
>> guards against interoperability failure.
>>
>> I would like to have an ecosystem where one can field a box, connect it
>> to everything else, and run well for *some* level of "well" - and I would
>> prefer that ecosystem to be one where it's possible to field the box
>> without making prior arrangements with anyone about licenses.
>>
>> This argument hasn't changed one whit since last time we discussed it.
>> And I don't see much movement on the specifics of the proposals either.
>>
>> We'll have to interoperate well with the codecs we field. So using some
>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
>> finished either. There's one sentence that needs to be removed.)
>>
>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>> prefer to re-discuss based on the knowledge that some significant players
>> have actually changed their minds.
>>
>> At the moment, I don't see signs that any of the poll respondents have
>> said "I have changed my mind".
>>
>> Harald
>>
>>
>>
>>
>>>
>>>
>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>>>> wrote:
>>>>
>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>>>>> And that's because something substantive has changed, or simply because
>>>>> wasting the WG time on this again is more entertaining than actually
>>>>> finishing a specification that can be independently implemented by all
>>>>> browser vendors? (A specification that we are nowhere near having, as
>>>>> far as
>>>>> I can tell)
>>>>>
>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>> it would appear that we've made some real progress as a result.  I'd
>>>> suggest that if we don't have new information, we continue to spend
>>>> our time productively.  If we can't find topics to occupy our meeting
>>>> agenda time, then maybe we can free an agenda slot.
>>>>
>>>> _______________________________________________
>>>> 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
>>
>
>
>
> --
> Alex. Gouaillard, PhD, PhD, MBA
>
> ------------------------------------------------------------------------------------
> CTO - Temasys Communications, S'pore / Mountain View
> President - CoSMo Software, Cambridge, MA
>
> ------------------------------------------------------------------------------------
> sg.linkedin.com/agouaillard
>
>    -
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net
http://www.jdrosen.net

--047d7b343d30bb22180505c80eab
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I&#39;m in favor of taking another run at this.<div><br></=
div><div>The working group has repeatedly said that an MTI codec is somethi=
ng we need to produce. And its one of the issues holding up wider adoption =
of the technology (not the only one for sure).</div><div><br></div><div>And=
 things have changed since the last meeting, a year ago now (November in Va=
ncouver). Cisco&#39;s open264 plugin shipped and now just recently is integ=
rated into Firefox. iOS8 shipped with APIs for H264. There are other things=
 worth noting. Will this change the minds of everyone? Surely not. Will it =
sway enough for us to achieve rough consensus? Maybe. IMHO - worth finding =
out.</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;=
<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmai=
l.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr">+1 to not having MTI codec discussion unless some progress has been mad=
e on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing harald&#39;s =C2=
=A0feeling that there was no change on the members position.=C2=A0</div><di=
v class=3D"gmail_extra"><div><div class=3D"h5"><br><div class=3D"gmail_quot=
e">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&lt=
;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestran=
d.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On 10/17=
/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span class=3D"HOEnZb"><font color=3D"#888888">-- <br><div dir=3D"ltr"=
>Alex. Gouaillard, PhD, PhD, MBA<div>--------------------------------------=
----------------------------------------------</div><div>CTO - Temasys Comm=
unications, S&#39;pore / Mountain View</div><div>President - CoSMo Software=
, Cambridge, MA</div><div>-------------------------------------------------=
-----------------------------------</div><div><a href=3D"http://sg.linkedin=
.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a></div><d=
iv><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-=
size:12px;font-family:Helvetica,Arial,sans-serif;vertical-align:baseline;li=
st-style:none;line-height:17px;display:table-cell;width:504px;color:rgb(51,=
51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline=
:0px;font-style:inherit;font-size:11px;font-family:inherit;vertical-align:b=
aseline;font-variant:inherit;line-height:1.2em"><dl style=3D"margin:0px;pad=
ding:0px;border:0px;outline:0px;font-style:inherit;font-family:inherit;vert=
ical-align:baseline;font-variant:inherit;line-height:inherit;word-wrap:brea=
k-word"><br></dl></li></ul></div></div>
</font></span></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><div dir=
=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdrosen.net=
" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://www.jdrose=
n.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>

--047d7b343d30bb22180505c80eab--


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"And its one of the issues holding up wider adoption of the technology"

[BA] Specifying an MTI encoder/decoder is not sufficient for
interoperability.  It is also necessary to specify the transport in enough
detail to allow independent implementations that interoperate well enough
to be usable in a wide variety of scenarios, including wireless networks
where loss is commonly experienced.

We made the mistake of having an MTI discussion previously with not enough
details on that subject, particularly with respect to H.264.
draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.

So if we are to have the discussion again, it should occur in the context
of detailed specifications and interoperability reports.





On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
wrote:

> I'm in favor of taking another run at this.
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of the
> technology (not the only one for sure).
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - worth
> finding out.
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
>> +1 to not having MTI codec discussion unless some progress has been made
>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there
>> was no change on the members position.
>>
>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
>> wrote:
>>
>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>
>>>> One thing we could do instead of wasting time on MTI is to actually
>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>> actually interoperate regardless of the codec.
>>>>
>>>
>>> The big argument for an MTI is actually the one stated in -overview: It
>>> guards against interoperability failure.
>>>
>>> I would like to have an ecosystem where one can field a box, connect it
>>> to everything else, and run well for *some* level of "well" - and I would
>>> prefer that ecosystem to be one where it's possible to field the box
>>> without making prior arrangements with anyone about licenses.
>>>
>>> This argument hasn't changed one whit since last time we discussed it.
>>> And I don't see much movement on the specifics of the proposals either.
>>>
>>> We'll have to interoperate well with the codecs we field. So using some
>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
>>> finished either. There's one sentence that needs to be removed.)
>>>
>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>> prefer to re-discuss based on the knowledge that some significant players
>>> have actually changed their minds.
>>>
>>> At the moment, I don't see signs that any of the poll respondents have
>>> said "I have changed my mind".
>>>
>>> Harald
>>>
>>>
>>>
>>>
>>>>
>>>>
>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>>>>> wrote:
>>>>>
>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>>>>>> And that's because something substantive has changed, or simply
>>>>>> because
>>>>>> wasting the WG time on this again is more entertaining than actually
>>>>>> finishing a specification that can be independently implemented by all
>>>>>> browser vendors? (A specification that we are nowhere near having, as
>>>>>> far as
>>>>>> I can tell)
>>>>>>
>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>> suggest that if we don't have new information, we continue to spend
>>>>> our time productively.  If we can't find topics to occupy our meeting
>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>
>>
>>
>>
>> --
>> Alex. Gouaillard, PhD, PhD, MBA
>>
>> ------------------------------------------------------------------------------------
>> CTO - Temasys Communications, S'pore / Mountain View
>> President - CoSMo Software, Cambridge, MA
>>
>> ------------------------------------------------------------------------------------
>> sg.linkedin.com/agouaillard
>>
>>    -
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> --
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> http://www.jdrosen.net
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c28eccf61aaa0505c8767a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">&quot;And its one of the issues holding up wider adoption =
of the technology&quot;<div><br></div><div>[BA] Specifying an MTI encoder/d=
ecoder is not sufficient for interoperability.=C2=A0 It is also necessary t=
o specify the transport in enough detail to allow independent implementatio=
ns that interoperate well enough to be usable in a wide variety of scenario=
s, including wireless networks where loss is commonly experienced. =C2=A0</=
div><div><br></div><div>We made the mistake of having an MTI discussion pre=
viously with not enough details on that subject, particularly with respect =
to H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best=
. =C2=A0</div><div><br></div><div>So if we are to have the discussion again=
, it should occur in the context of detailed specifications and interoperab=
ility reports.</div><div><br></div><div>=C2=A0</div><div><br></div><div>=C2=
=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">I&#=
39;m in favor of taking another run at this.<div><br></div><div>The working=
 group has repeatedly said that an MTI codec is something we need to produc=
e. And its one of the issues holding up wider adoption of the technology (n=
ot the only one for sure).</div><div><br></div><div>And things have changed=
 since the last meeting, a year ago now (November in Vancouver). Cisco&#39;=
s open264 plugin shipped and now just recently is integrated into Firefox. =
iOS8 shipped with APIs for H264. There are other things worth noting. Will =
this change the minds of everyone? Surely not. Will it sway enough for us t=
o achieve rough consensus? Maybe. IMHO - worth finding out.</div></div><div=
 class=3D"gmail_extra"><div><div class=3D"h5"><br><div class=3D"gmail_quote=
">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&=
lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@g=
mail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr">+1 to not having MTI codec discussion unless some progress has bee=
n made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing harald&#39;s =
=C2=A0feeling that there was no change on the members position.=C2=A0</div>=
<div class=3D"gmail_extra"><div><div><br><div class=3D"gmail_quote">On Fri,=
 Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=
=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</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"><span>On 10/17/2014 12=
:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c28eccf61aaa0505c8767a--


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--001a1140a9be9e52360505c8e061
Content-Type: text/plain; charset=UTF-8

@jonathan,

while you are right and availability of 264 implementation or hardware
acceleration has improved, it has never been reported as a problem in the
previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
AFAIR, the main reasons used by individuals to justify their positions.
Today, nothing has changed with respect to those two items (even though
providing open264 royalties and absorbing the license cost for some
platforms might have been a set in the right direction). This is why I
think the conditions are not met for a consensus to be reached.

Alex.


On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> "And its one of the issues holding up wider adoption of the technology"
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enough
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
> We made the mistake of having an MTI discussion previously with not enough
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
>> I'm in favor of taking another run at this.
>>
>> The working group has repeatedly said that an MTI codec is something we
>> need to produce. And its one of the issues holding up wider adoption of the
>> technology (not the only one for sure).
>>
>> And things have changed since the last meeting, a year ago now (November
>> in Vancouver). Cisco's open264 plugin shipped and now just recently is
>> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
>> things worth noting. Will this change the minds of everyone? Surely not.
>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - worth
>> finding out.
>>
>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>> agouaillard@gmail.com> wrote:
>>
>>> +1 to not having MTI codec discussion unless some progress has been made
>>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there
>>> was no change on the members position.
>>>
>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no
>>> > wrote:
>>>
>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>
>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>> actually interoperate regardless of the codec.
>>>>>
>>>>
>>>> The big argument for an MTI is actually the one stated in -overview: It
>>>> guards against interoperability failure.
>>>>
>>>> I would like to have an ecosystem where one can field a box, connect it
>>>> to everything else, and run well for *some* level of "well" - and I would
>>>> prefer that ecosystem to be one where it's possible to field the box
>>>> without making prior arrangements with anyone about licenses.
>>>>
>>>> This argument hasn't changed one whit since last time we discussed it.
>>>> And I don't see much movement on the specifics of the proposals either.
>>>>
>>>> We'll have to interoperate well with the codecs we field. So using some
>>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
>>>> finished either. There's one sentence that needs to be removed.)
>>>>
>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>> prefer to re-discuss based on the knowledge that some significant players
>>>> have actually changed their minds.
>>>>
>>>> At the moment, I don't see signs that any of the poll respondents have
>>>> said "I have changed my mind".
>>>>
>>>> Harald
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>> wrote:
>>>>>>> And that's because something substantive has changed, or simply
>>>>>>> because
>>>>>>> wasting the WG time on this again is more entertaining than actually
>>>>>>> finishing a specification that can be independently implemented by
>>>>>>> all
>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>> as far as
>>>>>>> I can tell)
>>>>>>>
>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>> our time productively.  If we can't find topics to occupy our meeting
>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Alex. Gouaillard, PhD, PhD, MBA
>>>
>>> ------------------------------------------------------------------------------------
>>> CTO - Temasys Communications, S'pore / Mountain View
>>> President - CoSMo Software, Cambridge, MA
>>>
>>> ------------------------------------------------------------------------------------
>>> sg.linkedin.com/agouaillard
>>>
>>>    -
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>>
>> --
>> Jonathan Rosenberg, Ph.D.
>> jdrosen@jdrosen.net
>> http://www.jdrosen.net
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>


-- 
Alex. Gouaillard, PhD, PhD, MBA
------------------------------------------------------------------------------------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
------------------------------------------------------------------------------------
sg.linkedin.com/agouaillard

   -

--001a1140a9be9e52360505c8e061
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">@jonathan,<div><br></div><div>while you are right and avai=
lability of 264 implementation or hardware acceleration has improved, it ha=
s never been reported as a problem in the previous pool by anyone. The 264 =
royalties, and the VP8 IP risks were, AFAIR, the main reasons used by indiv=
iduals to justify their positions. Today, nothing has changed with respect =
to those two items (even though providing open264 royalties and absorbing t=
he license cost for some platforms might have been a set in the right direc=
tion). This is why I think the conditions are not met for a consensus to be=
 reached.</div><div><br></div><div>Alex.</div><div><br></div></div><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 1=
1:43 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.abob=
a@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span class=3D"">&quot=
;And its one of the issues holding up wider adoption of the technology&quot=
;<div><br></div></span><div>[BA] Specifying an MTI encoder/decoder is not s=
ufficient for interoperability.=C2=A0 It is also necessary to specify the t=
ransport in enough detail to allow independent implementations that interop=
erate well enough to be usable in a wide variety of scenarios, including wi=
reless networks where loss is commonly experienced. =C2=A0</div><div><br></=
div><div>We made the mistake of having an MTI discussion previously with no=
t enough details on that subject, particularly with respect to H.264. draft=
-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best. =C2=A0</div><=
div><br></div><div>So if we are to have the discussion again, it should occ=
ur in the context of detailed specifications and interoperability reports.<=
/div><div><br></div><div>=C2=A0</div><div><br></div><div>=C2=A0</div></div>=
<div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
<span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_bla=
nk">jdrosen@jdrosen.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"><div dir=3D"ltr">I&#39;m in favor of taking another run at this.<div><=
br></div><div>The working group has repeatedly said that an MTI codec is so=
mething we need to produce. And its one of the issues holding up wider adop=
tion of the technology (not the only one for sure).</div><div><br></div><di=
v>And things have changed since the last meeting, a year ago now (November =
in Vancouver). Cisco&#39;s open264 plugin shipped and now just recently is =
integrated into Firefox. iOS8 shipped with APIs for H264. There are other t=
hings worth noting. Will this change the minds of everyone? Surely not. Wil=
l it sway enough for us to achieve rough consensus? Maybe. IMHO - worth fin=
ding out.</div></div><div class=3D"gmail_extra"><div><div><br><div class=3D=
"gmail_quote">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">a=
gouaillard@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
"><div dir=3D"ltr">+1 to not having MTI codec discussion unless some progre=
ss has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing har=
ald&#39;s =C2=A0feeling that there was no change on the members position.=
=C2=A0</div><div class=3D"gmail_extra"><div><div><br><div class=3D"gmail_qu=
ote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</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"><span>On 10/=
17/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>

--001a1140a9be9e52360505c8e061--


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On Sun, Oct 19, 2014 at 9:13 AM, Alexandre GOUAILLARD
<agouaillard@gmail.com> wrote:
> @jonathan,
>
> while you are right and availability of 264 implementation or hardware
> acceleration has improved, it has never been reported as a problem in the
> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
> AFAIR, the main reasons used by individuals to justify their positions.
> Today, nothing has changed with respect to those two items (even though
> providing open264 royalties and absorbing the license cost for some
> platforms might have been a set in the right direction). This is why I think
> the conditions are not met for a consensus to be reached.

But now VP8 is going through ISO, and the people claiming patents on
VP8 have had time to sue, and haven't. That's evidence that some
concerns are overblown.

>
> Alex.
>
>
> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>>
>> "And its one of the issues holding up wider adoption of the technology"
>>
>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>> interoperability.  It is also necessary to specify the transport in enough
>> detail to allow independent implementations that interoperate well enough to
>> be usable in a wide variety of scenarios, including wireless networks where
>> loss is commonly experienced.
>>
>> We made the mistake of having an MTI discussion previously with not enough
>> details on that subject, particularly with respect to H.264.
>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>
>> So if we are to have the discussion again, it should occur in the context
>> of detailed specifications and interoperability reports.
>>
>>
>>
>>
>>
>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
>> wrote:
>>>
>>> I'm in favor of taking another run at this.
>>>
>>> The working group has repeatedly said that an MTI codec is something we
>>> need to produce. And its one of the issues holding up wider adoption of the
>>> technology (not the only one for sure).
>>>
>>> And things have changed since the last meeting, a year ago now (November
>>> in Vancouver). Cisco's open264 plugin shipped and now just recently is
>>> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
>>> things worth noting. Will this change the minds of everyone? Surely not.
>>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - worth
>>> finding out.
>>>
>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD
>>> <agouaillard@gmail.com> wrote:
>>>>
>>>> +1 to not having MTI codec discussion unless some progress has been made
>>>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there
>>>> was no change on the members position.
>>>>
>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand
>>>> <harald@alvestrand.no> wrote:
>>>>>
>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>
>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>>> actually interoperate regardless of the codec.
>>>>>
>>>>>
>>>>> The big argument for an MTI is actually the one stated in -overview: It
>>>>> guards against interoperability failure.
>>>>>
>>>>> I would like to have an ecosystem where one can field a box, connect it
>>>>> to everything else, and run well for *some* level of "well" - and I would
>>>>> prefer that ecosystem to be one where it's possible to field the box without
>>>>> making prior arrangements with anyone about licenses.
>>>>>
>>>>> This argument hasn't changed one whit since last time we discussed it.
>>>>> And I don't see much movement on the specifics of the proposals either.
>>>>>
>>>>> We'll have to interoperate well with the codecs we field. So using some
>>>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
>>>>> finished either. There's one sentence that needs to be removed.)
>>>>>
>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>> prefer to re-discuss based on the knowledge that some significant players
>>>>> have actually changed their minds.
>>>>>
>>>>> At the moment, I don't see signs that any of the poll respondents have
>>>>> said "I have changed my mind".
>>>>>
>>>>> Harald
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> On Oct 16, 2014, at 2:28 PM, Martin Thomson
>>>>>>> <martin.thomson@gmail.com> wrote:
>>>>>>>
>>>>>>>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>> wrote:
>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>> because
>>>>>>>> wasting the WG time on this again is more entertaining than actually
>>>>>>>> finishing a specification that can be independently implemented by
>>>>>>>> all
>>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>>> as far as
>>>>>>>> I can tell)
>>>>>>>
>>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>>> our time productively.  If we can't find topics to occupy our meeting
>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>> President - CoSMo Software, Cambridge, MA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> sg.linkedin.com/agouaillard
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>>
>>>
>>> --
>>> Jonathan Rosenberg, Ph.D.
>>> jdrosen@jdrosen.net
>>> http://www.jdrosen.net
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>
>
>
> --
> Alex. Gouaillard, PhD, PhD, MBA
> ------------------------------------------------------------------------------------
> CTO - Temasys Communications, S'pore / Mountain View
> President - CoSMo Software, Cambridge, MA
> ------------------------------------------------------------------------------------
> sg.linkedin.com/agouaillard
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



-- 
"Those who would give up Essential Liberty to purchase a little
Temporary Safety deserve neither  Liberty nor Safety."
-- Benjamin Franklin


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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Plan for MTI video codec?
Thread-Index: AQHP669o0YA0VVXvZkG/wkm6G20Ux5w3tZ/C
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Subject: Re: [rtcweb] Plan for MTI video codec?
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I am hearing a lot of folks outside this group losing faith and beginning t=
o talk about writing off WebRTC for lack of consensus (including analysts).

I am therefore in favor of a very finite amount of "new-news" discussion an=
d another attempt at consensus. We either do or don't have it.


On Oct 19, 2014, at 11:15 AM, Jonathan Rosenberg <jdrosen@jdrosen.net<mailt=
o:jdrosen@jdrosen.net>> wrote:

I'm in favor of taking another run at this.

The working group has repeatedly said that an MTI codec is something we nee=
d to produce. And its one of the issues holding up wider adoption of the te=
chnology (not the only one for sure).

And things have changed since the last meeting, a year ago now (November in=
 Vancouver). Cisco's open264 plugin shipped and now just recently is integr=
ated into Firefox. iOS8 shipped with APIs for H264. There are other things =
worth noting. Will this change the minds of everyone? Surely not. Will it s=
way enough for us to achieve rough consensus? Maybe. IMHO - worth finding o=
ut.

On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <agouaillard@gmail.co=
m<mailto:agouaillard@gmail.com>> wrote:
+1 to not having MTI codec discussion unless some progress has been made on=
 VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there wa=
s no change on the members position.

On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
On 10/17/2014 12:02 AM, Bernard Aboba wrote:
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.

The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.

I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of "well" - and I would pref=
er that ecosystem to be one where it's possible to field the box without ma=
king prior arrangements with anyone about licenses.

This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.

We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)

I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.

At the moment, I don't see signs that any of the poll respondents have said=
 "I have changed my mind".

Harald






On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com<mailt=
o:martin.thomson@gmail.com>> wrote:

On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at<mailto:matthe=
w@matthew.at>> wrote:
And that's because something substantive has changed, or simply because
wasting the WG time on this again is more entertaining than actually
finishing a specification that can be independently implemented by all
browser vendors? (A specification that we are nowhere near having, as far a=
s
I can tell)
Personally, I've found the reprieve from this fight refreshing.  And
it would appear that we've made some real progress as a result.  I'd
suggest that if we don't have new information, we continue to spend
our time productively.  If we can't find topics to occupy our meeting
agenda time, then maybe we can free an agenda slot.

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

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



--
Alex. Gouaillard, PhD, PhD, MBA
---------------------------------------------------------------------------=
---------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
---------------------------------------------------------------------------=
---------
sg.linkedin.com/agouaillard<http://sg.linkedin.com/agouaillard>

  *


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




--
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net<mailto:jdrosen@jdrosen.net>
http://www.jdrosen.net
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

--_000_16C290A2EB7C4D778871C7E82B374C48ciscocom_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>I am hearing a lot of folks outside this group losing faith and beginn=
ing to talk about writing off WebRTC for lack of consensus (including analy=
sts).</div>
<div><br>
</div>
<div>I am therefore in favor of a very finite amount of &quot;new-news&quot=
; discussion and another attempt at consensus. We either do or don't have i=
t.<br>
<br>
</div>
<div><br>
On Oct 19, 2014, at 11:15 AM, Jonathan Rosenberg &lt;<a href=3D"mailto:jdro=
sen@jdrosen.net">jdrosen@jdrosen.net</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">I'm in favor of taking another run at this.
<div><br>
</div>
<div>The working group has repeatedly said that an MTI codec is something w=
e need to produce. And its one of the issues holding up wider adoption of t=
he technology (not the only one for sure).</div>
<div><br>
</div>
<div>And things have changed since the last meeting, a year ago now (Novemb=
er in Vancouver). Cisco's open264 plugin shipped and now just recently is i=
ntegrated into Firefox. iOS8 shipped with APIs for H264. There are other th=
ings worth noting. Will this change
 the minds of everyone? Surely not. Will it sway enough for us to achieve r=
ough consensus? Maybe. IMHO - worth finding out.</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAI=
LLARD <span dir=3D"ltr">
&lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@=
gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">&#43;1 to not having MTI codec discussion unless some prog=
ress has been made on VP8 at MPEG.&nbsp;Any news on that? I'm sharing haral=
d's &nbsp;feeling that there was no change on the members position.&nbsp;</=
div>
<div class=3D"gmail_extra">
<div>
<div class=3D"h5"><br>
<div class=3D"gmail_quote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestra=
nd <span dir=3D"ltr">
&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvest=
rand.no</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">
<span>On 10/17/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br>
</span>The big argument for an MTI is actually the one stated in -overview:=
 It guards against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it's possible to field the box =
without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.<br>
<br>
We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)<br>
<br>
I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.<br>
<br>
At the moment, I don't see signs that any of the poll respondents have said=
 &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span>
<div>
<div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that's because something substantive has changed, or simply because<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I've found the reprieve from this fight refreshing.&nbsp; And<b=
r>
it would appear that we've made some real progress as a result.&nbsp; I'd<b=
r>
suggest that if we don't have new information, we continue to spend<br>
our time productively.&nbsp; If we can't find topics to occupy our meeting<=
br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div>
</div>
</blockquote>
</div>
<br>
<br clear=3D"all">
<div><br>
</div>
</div>
</div>
<span class=3D"HOEnZb"><font color=3D"#888888">-- <br>
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA
<div>----------------------------------------------------------------------=
--------------</div>
<div>CTO - Temasys Communications, S'pore / Mountain View</div>
<div>President - CoSMo Software, Cambridge, MA</div>
<div>----------------------------------------------------------------------=
--------------</div>
<div><a href=3D"http://sg.linkedin.com/agouaillard" target=3D"_blank">sg.li=
nkedin.com/agouaillard</a></div>
<div>
<ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-siz=
e:12px;font-family:Helvetica,Arial,sans-serif;vertical-align:baseline;list-=
style:none;line-height:17px;display:table-cell;width:504px;color:rgb(51,51,=
51)">
<li style=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;fon=
t-style:inherit;font-size:11px;font-family:inherit;vertical-align:baseline;=
font-variant:inherit;line-height:1.2em">
<dl style=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inher=
it;font-family:inherit;vertical-align:baseline;font-variant:inherit;line-he=
ight:inherit;word-wrap:break-word">
<br>
</dl>
</li></ul>
</div>
</div>
</font></span></div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
<br clear=3D"all">
<div><br>
</div>
-- <br>
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a></div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_16C290A2EB7C4D778871C7E82B374C48ciscocom_--


From nobody Sun Oct 19 11:13:41 2014
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From: Jonathan Rosenberg <jdrosen@jdrosen.net>
To: Alexandre GOUAILLARD <agouaillard@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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@Alexandre - you say "Today, nothing has changed with respect to those two
items (even though providing open264 royalties and absorbing the license
cost for some platforms might have been a set in the right direction). ".
But, as you say, the availability of Firefox with H264 is a change
(previously it was not yet available); the fact that Cisco has in fact
fronted the cost is a change (at the last meeting some were skeptical this
would happen, but it has). The other big news was IOS8, which now enables
apps to access H264 and Apple pays the cost. Last time, the lack of a
solution on IOS was a big deal. That is also now, no longer an issue. As
such I think there are material changes.


On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <
agouaillard@gmail.com> wrote:

> @jonathan,
>
> while you are right and availability of 264 implementation or hardware
> acceleration has improved, it has never been reported as a problem in the
> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
> AFAIR, the main reasons used by individuals to justify their positions.
> Today, nothing has changed with respect to those two items (even though
> providing open264 royalties and absorbing the license cost for some
> platforms might have been a set in the right direction). This is why I
> think the conditions are not met for a consensus to be reached.
>
> Alex.
>
>
> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> "And its one of the issues holding up wider adoption of the technology"
>>
>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>> interoperability.  It is also necessary to specify the transport in enough
>> detail to allow independent implementations that interoperate well enough
>> to be usable in a wide variety of scenarios, including wireless networks
>> where loss is commonly experienced.
>>
>> We made the mistake of having an MTI discussion previously with not
>> enough details on that subject, particularly with respect to H.264.
>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>
>> So if we are to have the discussion again, it should occur in the context
>> of detailed specifications and interoperability reports.
>>
>>
>>
>>
>>
>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
>> wrote:
>>
>>> I'm in favor of taking another run at this.
>>>
>>> The working group has repeatedly said that an MTI codec is something we
>>> need to produce. And its one of the issues holding up wider adoption of the
>>> technology (not the only one for sure).
>>>
>>> And things have changed since the last meeting, a year ago now (November
>>> in Vancouver). Cisco's open264 plugin shipped and now just recently is
>>> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
>>> things worth noting. Will this change the minds of everyone? Surely not.
>>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - worth
>>> finding out.
>>>
>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>>> agouaillard@gmail.com> wrote:
>>>
>>>> +1 to not having MTI codec discussion unless some progress has been
>>>> made on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>> there was no change on the members position.
>>>>
>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <
>>>> harald@alvestrand.no> wrote:
>>>>
>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>
>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>>> actually interoperate regardless of the codec.
>>>>>>
>>>>>
>>>>> The big argument for an MTI is actually the one stated in -overview:
>>>>> It guards against interoperability failure.
>>>>>
>>>>> I would like to have an ecosystem where one can field a box, connect
>>>>> it to everything else, and run well for *some* level of "well" - and I
>>>>> would prefer that ecosystem to be one where it's possible to field the box
>>>>> without making prior arrangements with anyone about licenses.
>>>>>
>>>>> This argument hasn't changed one whit since last time we discussed it.
>>>>> And I don't see much movement on the specifics of the proposals either.
>>>>>
>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>> some time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1
>>>>> isn't finished either. There's one sentence that needs to be removed.)
>>>>>
>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>> prefer to re-discuss based on the knowledge that some significant players
>>>>> have actually changed their minds.
>>>>>
>>>>> At the moment, I don't see signs that any of the poll respondents have
>>>>> said "I have changed my mind".
>>>>>
>>>>> Harald
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <
>>>>>>> martin.thomson@gmail.com> wrote:
>>>>>>>
>>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>> wrote:
>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>> because
>>>>>>>> wasting the WG time on this again is more entertaining than actually
>>>>>>>> finishing a specification that can be independently implemented by
>>>>>>>> all
>>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>>> as far as
>>>>>>>> I can tell)
>>>>>>>>
>>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>>> our time productively.  If we can't find topics to occupy our meeting
>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>> President - CoSMo Software, Cambridge, MA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> sg.linkedin.com/agouaillard
>>>>
>>>>    -
>>>>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>
>>>
>>> --
>>> Jonathan Rosenberg, Ph.D.
>>> jdrosen@jdrosen.net
>>> http://www.jdrosen.net
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>
>
> --
> Alex. Gouaillard, PhD, PhD, MBA
>
> ------------------------------------------------------------------------------------
> CTO - Temasys Communications, S'pore / Mountain View
> President - CoSMo Software, Cambridge, MA
>
> ------------------------------------------------------------------------------------
> sg.linkedin.com/agouaillard
>
>    -
>
>


-- 
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net
http://www.jdrosen.net

--047d7b343d30c2041f0505ca8d8e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">@Alexandre - you say &quot;<span style=3D"font-family:aria=
l,sans-serif;font-size:13px">Today, nothing has changed with respect to tho=
se two items (even though providing open264 royalties and absorbing the lic=
ense cost for some platforms might have been a set in the right direction).=
=C2=A0&quot;. But, as you say, the availability of Firefox with H264 is a c=
hange (previously it was not yet available); the fact that Cisco has in fac=
t fronted the cost is a change (at the last meeting some were skeptical thi=
s would happen, but it has). The other big news was IOS8, which now enables=
 apps to access H264 and Apple pays the cost. Last time, the lack of a solu=
tion on IOS was a big deal. That is also now, no longer an issue. As such I=
 think there are material changes.</span><div><span style=3D"font-family:ar=
ial,sans-serif;font-size:13px"><br></span></div></div><div class=3D"gmail_e=
xtra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 12:13 PM, Alex=
andre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.=
com" target=3D"_blank">agouaillard@gmail.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"ltr">@jonathan,<div><br></div><div>wh=
ile you are right and availability of 264 implementation or hardware accele=
ration has improved, it has never been reported as a problem in the previou=
s pool by anyone. The 264 royalties, and the VP8 IP risks were, AFAIR, the =
main reasons used by individuals to justify their positions. Today, nothing=
 has changed with respect to those two items (even though providing open264=
 royalties and absorbing the license cost for some platforms might have bee=
n a set in the right direction). This is why I think the conditions are not=
 met for a consensus to be reached.</div><div><br></div><div>Alex.</div><di=
v><br></div></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gma=
il_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 11:43 PM, =
Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.c=
om" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><div dir=3D"ltr"><span>&quot;And its one of the =
issues holding up wider adoption of the technology&quot;<div><br></div></sp=
an><div>[BA] Specifying an MTI encoder/decoder is not sufficient for intero=
perability.=C2=A0 It is also necessary to specify the transport in enough d=
etail to allow independent implementations that interoperate well enough to=
 be usable in a wide variety of scenarios, including wireless networks wher=
e loss is commonly experienced. =C2=A0</div><div><br></div><div>We made the=
 mistake of having an MTI discussion previously with not enough details on =
that subject, particularly with respect to H.264. draft-ietf-rtcweb-video s=
ections 4.2 - 4.4 remain sketchy at best. =C2=A0</div><div><br></div><div>S=
o if we are to have the discussion again, it should occur in the context of=
 detailed specifications and interoperability reports.</div><div><br></div>=
<div>=C2=A0</div><div><br></div><div>=C2=A0</div></div><div><div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 8:1=
4 AM, Jonathan Rosenberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jd=
rosen.net" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">I&#39;m in favor of taking a=
nother run at this.<div><br></div><div>The working group has repeatedly sai=
d that an MTI codec is something we need to produce. And its one of the iss=
ues holding up wider adoption of the technology (not the only one for sure)=
.</div><div><br></div><div>And things have changed since the last meeting, =
a year ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped =
and now just recently is integrated into Firefox. iOS8 shipped with APIs fo=
r H264. There are other things worth noting. Will this change the minds of =
everyone? Surely not. Will it sway enough for us to achieve rough consensus=
? Maybe. IMHO - worth finding out.</div></div><div class=3D"gmail_extra"><d=
iv><div><br><div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:08 PM, Ale=
xandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail=
.com" target=3D"_blank">agouaillard@gmail.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><div dir=3D"ltr">+1 to not having MTI codec disc=
ussion unless some progress has been made on VP8 at MPEG.=C2=A0Any news on =
that? I&#39;m sharing harald&#39;s =C2=A0feeling that there was no change o=
n the members position.=C2=A0</div><div class=3D"gmail_extra"><div><div><br=
><div class=3D"gmail_quote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestr=
and <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D=
"_blank">harald@alvestrand.no</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"><span>On 10/17/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdr=
osen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://ww=
w.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>

--047d7b343d30c2041f0505ca8d8e--


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Date: Mon, 20 Oct 2014 02:46:41 +0800
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From: Alexandre GOUAILLARD <agouaillard@gmail.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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@watson

I would prefer the ISO process to see through before deciding. That's why I
keep bothering harald for news at every meeting ;-) If for any reason it
does not happen, we're back where we were when we did the last pool.
Nobody has sued .... yet, does not mean there is no IP risk. Actually,
there were at least 3 counts of nokia suing, but the three cases were
negotiated and the result is not available to the public. (here
<http://www.fosspatents.com/2013/05/nokia-files-third-patent-infringement.html>
)

@jonathan

Point taken, I hadn't thought about the implications of H264 hardware
acceleration. If indeed it leverages the need to get a license, that makes
the situation better. Would that be enough ...? We could map all the cases
and see what is left to be dealt with. In any case, deciding about 264
without looking at 265 seems shortsighted, and I am not aware of any
binary, or hardware acceleration support available for 265 yet (even though
iPhone 6 claims to supports 265 acc in its specs, I'm not aware of API that
would allow native app to leverage this capacity).

Correct me if I'm wrong or if I'm missing anything (anyone):

-- desktop browsers --
- firefox, let's assume the 264 binary is used. It also supports VP8.
- chrome on desktop supports vp8 but not 264 (as far as webrtc is
concerned, the media element supports 264 playback).
- what about IE / Safari ? I don't think apple will make a statement, but I
remember reading that microsoft would support 264 (at least for media
element playback) if it is available on the system. Bernard / shijun, any
comment about which codec would be used for ORTC/webRTC?

-- mobile browsers --
- no firefox on iOS. What about the android version, what does it support?
- chrome supports only VP8, and can't support webRTC on iOS anyway (even
though justin is working on a audio-only, API-injection-in-webkit solution).
- opera?

-- mobile native --
- iOS 8+ has system API for hw acc 264
- android has system API for both hw acc 264 and VP8 (is it true?)
- the webrtc.org stack does not support 264 even though kaiduan die from
blackberry provided a patch for this
- the ericson's openwebrtc stack supports 264 in multiple ways (does it
support hw acc?) but lack support for Windows and data channel.

-- Others --
- lots of MCUs / SFUs / Gateways would need 264 in more cases than just the
interoperability between webrtc and SIP, and I'm not sure the cisco
binaries can be used there.

On Mon, Oct 20, 2014 at 2:13 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
wrote:

> @Alexandre - you say "Today, nothing has changed with respect to those
> two items (even though providing open264 royalties and absorbing the
> license cost for some platforms might have been a set in the right
> direction). ". But, as you say, the availability of Firefox with H264 is a
> change (previously it was not yet available); the fact that Cisco has in
> fact fronted the cost is a change (at the last meeting some were skeptical
> this would happen, but it has). The other big news was IOS8, which now
> enables apps to access H264 and Apple pays the cost. Last time, the lack of
> a solution on IOS was a big deal. That is also now, no longer an issue. As
> such I think there are material changes.
>
>
> On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
>> @jonathan,
>>
>> while you are right and availability of 264 implementation or hardware
>> acceleration has improved, it has never been reported as a problem in the
>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
>> AFAIR, the main reasons used by individuals to justify their positions.
>> Today, nothing has changed with respect to those two items (even though
>> providing open264 royalties and absorbing the license cost for some
>> platforms might have been a set in the right direction). This is why I
>> think the conditions are not met for a consensus to be reached.
>>
>> Alex.
>>
>>
>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com>
>> wrote:
>>
>>> "And its one of the issues holding up wider adoption of the technology"
>>>
>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>>> interoperability.  It is also necessary to specify the transport in enough
>>> detail to allow independent implementations that interoperate well enough
>>> to be usable in a wide variety of scenarios, including wireless networks
>>> where loss is commonly experienced.
>>>
>>> We made the mistake of having an MTI discussion previously with not
>>> enough details on that subject, particularly with respect to H.264.
>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>>
>>> So if we are to have the discussion again, it should occur in the
>>> context of detailed specifications and interoperability reports.
>>>
>>>
>>>
>>>
>>>
>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net
>>> > wrote:
>>>
>>>> I'm in favor of taking another run at this.
>>>>
>>>> The working group has repeatedly said that an MTI codec is something we
>>>> need to produce. And its one of the issues holding up wider adoption of the
>>>> technology (not the only one for sure).
>>>>
>>>> And things have changed since the last meeting, a year ago now
>>>> (November in Vancouver). Cisco's open264 plugin shipped and now just
>>>> recently is integrated into Firefox. iOS8 shipped with APIs for H264. There
>>>> are other things worth noting. Will this change the minds of everyone?
>>>> Surely not. Will it sway enough for us to achieve rough consensus? Maybe.
>>>> IMHO - worth finding out.
>>>>
>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>>>> agouaillard@gmail.com> wrote:
>>>>
>>>>> +1 to not having MTI codec discussion unless some progress has been
>>>>> made on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>>> there was no change on the members position.
>>>>>
>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <
>>>>> harald@alvestrand.no> wrote:
>>>>>
>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>
>>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>>>> actually interoperate regardless of the codec.
>>>>>>>
>>>>>>
>>>>>> The big argument for an MTI is actually the one stated in -overview:
>>>>>> It guards against interoperability failure.
>>>>>>
>>>>>> I would like to have an ecosystem where one can field a box, connect
>>>>>> it to everything else, and run well for *some* level of "well" - and I
>>>>>> would prefer that ecosystem to be one where it's possible to field the box
>>>>>> without making prior arrangements with anyone about licenses.
>>>>>>
>>>>>> This argument hasn't changed one whit since last time we discussed
>>>>>> it. And I don't see much movement on the specifics of the proposals either.
>>>>>>
>>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>>> some time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1
>>>>>> isn't finished either. There's one sentence that needs to be removed.)
>>>>>>
>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>>> prefer to re-discuss based on the knowledge that some significant players
>>>>>> have actually changed their minds.
>>>>>>
>>>>>> At the moment, I don't see signs that any of the poll respondents
>>>>>> have said "I have changed my mind".
>>>>>>
>>>>>> Harald
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <
>>>>>>>> martin.thomson@gmail.com> wrote:
>>>>>>>>
>>>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>>> wrote:
>>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>>> because
>>>>>>>>> wasting the WG time on this again is more entertaining than
>>>>>>>>> actually
>>>>>>>>> finishing a specification that can be independently implemented by
>>>>>>>>> all
>>>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>>>> as far as
>>>>>>>>> I can tell)
>>>>>>>>>
>>>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>>>> our time productively.  If we can't find topics to occupy our
>>>>>>>> meeting
>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>
>>>>> ------------------------------------------------------------------------------------
>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>> President - CoSMo Software, Cambridge, MA
>>>>>
>>>>> ------------------------------------------------------------------------------------
>>>>> sg.linkedin.com/agouaillard
>>>>>
>>>>>    -
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Jonathan Rosenberg, Ph.D.
>>>> jdrosen@jdrosen.net
>>>> http://www.jdrosen.net
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>
>>
>>
>> --
>> Alex. Gouaillard, PhD, PhD, MBA
>>
>> ------------------------------------------------------------------------------------
>> CTO - Temasys Communications, S'pore / Mountain View
>> President - CoSMo Software, Cambridge, MA
>>
>> ------------------------------------------------------------------------------------
>> sg.linkedin.com/agouaillard
>>
>>    -
>>
>>
>
>
> --
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> http://www.jdrosen.net
>



-- 
Alex. Gouaillard, PhD, PhD, MBA
------------------------------------------------------------------------------------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
------------------------------------------------------------------------------------
sg.linkedin.com/agouaillard

   -

--089e013c5d2cdc53cb0505cb049b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">@watson<div><br></div><div>I would prefer the ISO process =
to see through before deciding. That&#39;s why I keep bothering harald for =
news at every meeting ;-) If for any reason it does not happen, we&#39;re b=
ack where we were when we did the last pool.</div><div>Nobody has sued ....=
 yet, does not mean there is no IP risk. Actually, there were at least 3 co=
unts of nokia suing, but the three cases were negotiated and the result is =
not available to the public. (<a href=3D"http://www.fosspatents.com/2013/05=
/nokia-files-third-patent-infringement.html">here</a>)</div><div><br></div>=
<div>@jonathan</div><div><br></div><div>Point taken, I hadn&#39;t thought a=
bout the implications of H264 hardware acceleration. If indeed it leverages=
 the need to get a license, that makes the situation better. Would that be =
enough ...? We could map all the cases and see what is left to be dealt wit=
h. In any case, deciding about 264 without looking at 265 seems shortsighte=
d, and I am not aware of any binary, or hardware acceleration support avail=
able for 265 yet (even though iPhone 6 claims to supports 265 acc in its sp=
ecs, I&#39;m not aware of API that would allow native app to leverage this =
capacity).=C2=A0</div><div><br></div><div>Correct me if I&#39;m wrong or if=
 I&#39;m missing anything (anyone):<br></div><div><br></div><div>-- desktop=
 browsers --</div><div>- firefox, let&#39;s assume the 264 binary is used. =
It also supports VP8.</div><div>- chrome on desktop supports vp8 but not 26=
4 (as far as webrtc is concerned, the media element supports 264 playback).=
</div><div>- what about IE / Safari ? I don&#39;t think apple will make a s=
tatement, but I remember reading that microsoft would support 264 (at least=
 for media element playback) if it is available on the system. Bernard / sh=
ijun, any comment about which codec would be used for ORTC/webRTC?</div><di=
v><br></div><div>-- mobile browsers --</div><div>- no firefox on iOS. What =
about the android version, what does it support?</div><div>- chrome support=
s only VP8, and can&#39;t support webRTC on iOS anyway (even though justin =
is working on a audio-only, API-injection-in-webkit solution).</div><div>- =
opera?</div><div><br></div><div>-- mobile native --</div><div>- iOS 8+ has =
system API for hw acc 264</div><div>- android has system API for both hw ac=
c 264 and VP8 (is it true?)</div><div>- the <a href=3D"http://webrtc.org">w=
ebrtc.org</a> stack does not support 264 even though kaiduan die from black=
berry provided a patch for this</div><div>- the ericson&#39;s openwebrtc st=
ack supports 264 in multiple ways (does it support hw acc?) but lack suppor=
t for Windows and data channel.</div><div><br></div><div>-- Others --</div>=
<div>- lots of MCUs / SFUs / Gateways would need 264 in more cases than jus=
t the interoperability between webrtc and SIP, and I&#39;m not sure the cis=
co binaries can be used there.=C2=A0</div></div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 2:13 AM, Jonathan Ro=
senberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=
=3D"_blank">jdrosen@jdrosen.net</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 dir=3D"ltr">@Alexandre - you say &quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">Today, nothing has changed with =
respect to those two items (even though providing open264 royalties and abs=
orbing the license cost for some platforms might have been a set in the rig=
ht direction).=C2=A0&quot;. But, as you say, the availability of Firefox wi=
th H264 is a change (previously it was not yet available); the fact that Ci=
sco has in fact fronted the cost is a change (at the last meeting some were=
 skeptical this would happen, but it has). The other big news was IOS8, whi=
ch now enables apps to access H264 and Apple pays the cost. Last time, the =
lack of a solution on IOS was a big deal. That is also now, no longer an is=
sue. As such I think there are material changes.</span><div><span style=3D"=
font-family:arial,sans-serif;font-size:13px"><br></span></div></div><div cl=
ass=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blan=
k">agouaillard@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div dir=3D"ltr">@jonathan,<div><br></div><div>while you are right an=
d availability of 264 implementation or hardware acceleration has improved,=
 it has never been reported as a problem in the previous pool by anyone. Th=
e 264 royalties, and the VP8 IP risks were, AFAIR, the main reasons used by=
 individuals to justify their positions. Today, nothing has changed with re=
spect to those two items (even though providing open264 royalties and absor=
bing the license cost for some platforms might have been a set in the right=
 direction). This is why I think the conditions are not met for a consensus=
 to be reached.</div><div><br></div><div>Alex.</div><div><br></div></div><d=
iv><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, O=
ct 19, 2014 at 11:43 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span>=
&quot;And its one of the issues holding up wider adoption of the technology=
&quot;<div><br></div></span><div>[BA] Specifying an MTI encoder/decoder is =
not sufficient for interoperability.=C2=A0 It is also necessary to specify =
the transport in enough detail to allow independent implementations that in=
teroperate well enough to be usable in a wide variety of scenarios, includi=
ng wireless networks where loss is commonly experienced. =C2=A0</div><div><=
br></div><div>We made the mistake of having an MTI discussion previously wi=
th not enough details on that subject, particularly with respect to H.264. =
draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best. =C2=A0</=
div><div><br></div><div>So if we are to have the discussion again, it shoul=
d occur in the context of detailed specifications and interoperability repo=
rts.</div><div><br></div><div>=C2=A0</div><div><br></div><div>=C2=A0</div><=
/div><div><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">I&#=
39;m in favor of taking another run at this.<div><br></div><div>The working=
 group has repeatedly said that an MTI codec is something we need to produc=
e. And its one of the issues holding up wider adoption of the technology (n=
ot the only one for sure).</div><div><br></div><div>And things have changed=
 since the last meeting, a year ago now (November in Vancouver). Cisco&#39;=
s open264 plugin shipped and now just recently is integrated into Firefox. =
iOS8 shipped with APIs for H264. There are other things worth noting. Will =
this change the minds of everyone? Surely not. Will it sway enough for us t=
o achieve rough consensus? Maybe. IMHO - worth finding out.</div></div><div=
 class=3D"gmail_extra"><div><div><br><div class=3D"gmail_quote">On Sat, Oct=
 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D=
"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmail.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">+1 to =
not having MTI codec discussion unless some progress has been made on VP8 a=
t MPEG.=C2=A0Any news on that? I&#39;m sharing harald&#39;s =C2=A0feeling t=
hat there was no change on the members position.=C2=A0</div><div class=3D"g=
mail_extra"><div><div><br><div class=3D"gmail_quote">On Fri, Oct 17, 2014 a=
t 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald=
@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><span>On 10/17/2014 12:02 AM, Bernard A=
boba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdr=
osen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://ww=
w.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>

--089e013c5d2cdc53cb0505cb049b--


From nobody Sun Oct 19 12:39:47 2014
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Date: Sun, 19 Oct 2014 21:39:44 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: [rtcweb] Scheduling a separate slot for MTI VC Disuccions (Was: Plan for MTI video codec?)
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Given how:

a) a big number of people have indicated their concern that codec 
discussions would steal time away from other important topics that need 
to be progressed and how

b) a second group of people seem to consider it important to have a 
discussion on developments around deployment, adoption and 
standardization of VP8 and H.264

then wouldn't it be reasonable to try and allocate an extra slot for an 
ad hoc meeting on the MTI Video Codec topic?

Emil

-- 
https://jitsi.org


From nobody Sun Oct 19 13:31:07 2014
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Sun, 19 Oct 2014 13:30:34 -0700
Message-ID: <CAOW+2dv7Wd=SHKDnh3cv9MUA-e9dMzuYojYsinHRz=j9QAm_uQ@mail.gmail.com>
To: Alexandre GOUAILLARD <agouaillard@gmail.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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Alex said:

"I don't think apple will make a statement, but I remember reading that
microsoft would support 264 (at least for media element playback) if it is
available on the system.  Bernard / shijun, any comment about which codec
would be used for ORTC/WebRTC?"

[BA] Microsoft, Apple, Cisco, Blackberry, Ericsson and others have
previously co-authored an H.264 MTI proposal:
https://tools.ietf.org/html/draft-burman-rtcweb-h264-proposal

Even though that proposal expires next week, we still believe it is
possible for an H.264 encoder/decoder to serve as a foundation for
interoperability
between independent implementations.   However, choosing an MTI codec and
specifying aspects of the codec is not enough.  On wireless networks
(bursty) packet loss is common, so it is important to specify transport
aspects and robustness measures.  This would include an MTI FEC scheme and
bandwidth estimation techniques, as well as use of codec-specific feedback
messages.

In terms of what needs to be done, there is work needed on the rtp-usage
document (such as addressing the MTI FEC scheme),  draft-ietf-rtcweb-video,
etc.  Focussing on another MTI debate at this time could make it difficult
to make major progress before IETF 92.


If instead of tackling the hard problems, we spend our precious time on yet
another MTI debate, this will not satisfy customers who have burned
before.  From past experience, they know that if interoperability issues
are not addressed *early on*, patching things up after the fact can be
difficult and time consuming.  Yet that is *exactly* where we are headed.

On Sun, Oct 19, 2014 at 11:46 AM, Alexandre GOUAILLARD <
agouaillard@gmail.com> wrote:

> @watson
>
> I would prefer the ISO process to see through before deciding. That's why
> I keep bothering harald for news at every meeting ;-) If for any reason it
> does not happen, we're back where we were when we did the last pool.
> Nobody has sued .... yet, does not mean there is no IP risk. Actually,
> there were at least 3 counts of nokia suing, but the three cases were
> negotiated and the result is not available to the public. (here
> <http://www.fosspatents.com/2013/05/nokia-files-third-patent-infringement.html>
> )
>
> @jonathan
>
> Point taken, I hadn't thought about the implications of H264 hardware
> acceleration. If indeed it leverages the need to get a license, that makes
> the situation better. Would that be enough ...? We could map all the cases
> and see what is left to be dealt with. In any case, deciding about 264
> without looking at 265 seems shortsighted, and I am not aware of any
> binary, or hardware acceleration support available for 265 yet (even though
> iPhone 6 claims to supports 265 acc in its specs, I'm not aware of API that
> would allow native app to leverage this capacity).
>
> Correct me if I'm wrong or if I'm missing anything (anyone):
>
> -- desktop browsers --
> - firefox, let's assume the 264 binary is used. It also supports VP8.
> - chrome on desktop supports vp8 but not 264 (as far as webrtc is
> concerned, the media element supports 264 playback).
> - what about IE / Safari ? I don't think apple will make a statement, but
> I remember reading that microsoft would support 264 (at least for media
> element playback) if it is available on the system. Bernard / shijun, any
> comment about which codec would be used for ORTC/webRTC?
>
> -- mobile browsers --
> - no firefox on iOS. What about the android version, what does it support?
> - chrome supports only VP8, and can't support webRTC on iOS anyway (even
> though justin is working on a audio-only, API-injection-in-webkit solution).
> - opera?
>
> -- mobile native --
> - iOS 8+ has system API for hw acc 264
> - android has system API for both hw acc 264 and VP8 (is it true?)
> - the webrtc.org stack does not support 264 even though kaiduan die from
> blackberry provided a patch for this
> - the ericson's openwebrtc stack supports 264 in multiple ways (does it
> support hw acc?) but lack support for Windows and data channel.
>
> -- Others --
> - lots of MCUs / SFUs / Gateways would need 264 in more cases than just
> the interoperability between webrtc and SIP, and I'm not sure the cisco
> binaries can be used there.
>
> On Mon, Oct 20, 2014 at 2:13 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
>> @Alexandre - you say "Today, nothing has changed with respect to those
>> two items (even though providing open264 royalties and absorbing the
>> license cost for some platforms might have been a set in the right
>> direction). ". But, as you say, the availability of Firefox with H264 is a
>> change (previously it was not yet available); the fact that Cisco has in
>> fact fronted the cost is a change (at the last meeting some were skeptical
>> this would happen, but it has). The other big news was IOS8, which now
>> enables apps to access H264 and Apple pays the cost. Last time, the lack of
>> a solution on IOS was a big deal. That is also now, no longer an issue. As
>> such I think there are material changes.
>>
>>
>> On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <
>> agouaillard@gmail.com> wrote:
>>
>>> @jonathan,
>>>
>>> while you are right and availability of 264 implementation or hardware
>>> acceleration has improved, it has never been reported as a problem in the
>>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
>>> AFAIR, the main reasons used by individuals to justify their positions.
>>> Today, nothing has changed with respect to those two items (even though
>>> providing open264 royalties and absorbing the license cost for some
>>> platforms might have been a set in the right direction). This is why I
>>> think the conditions are not met for a consensus to be reached.
>>>
>>> Alex.
>>>
>>>
>>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com
>>> > wrote:
>>>
>>>> "And its one of the issues holding up wider adoption of the technology"
>>>>
>>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>>>> interoperability.  It is also necessary to specify the transport in enough
>>>> detail to allow independent implementations that interoperate well enough
>>>> to be usable in a wide variety of scenarios, including wireless networks
>>>> where loss is commonly experienced.
>>>>
>>>> We made the mistake of having an MTI discussion previously with not
>>>> enough details on that subject, particularly with respect to H.264.
>>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>>>
>>>> So if we are to have the discussion again, it should occur in the
>>>> context of detailed specifications and interoperability reports.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <
>>>> jdrosen@jdrosen.net> wrote:
>>>>
>>>>> I'm in favor of taking another run at this.
>>>>>
>>>>> The working group has repeatedly said that an MTI codec is something
>>>>> we need to produce. And its one of the issues holding up wider adoption of
>>>>> the technology (not the only one for sure).
>>>>>
>>>>> And things have changed since the last meeting, a year ago now
>>>>> (November in Vancouver). Cisco's open264 plugin shipped and now just
>>>>> recently is integrated into Firefox. iOS8 shipped with APIs for H264. There
>>>>> are other things worth noting. Will this change the minds of everyone?
>>>>> Surely not. Will it sway enough for us to achieve rough consensus? Maybe.
>>>>> IMHO - worth finding out.
>>>>>
>>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>>>>> agouaillard@gmail.com> wrote:
>>>>>
>>>>>> +1 to not having MTI codec discussion unless some progress has been
>>>>>> made on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>>>> there was no change on the members position.
>>>>>>
>>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <
>>>>>> harald@alvestrand.no> wrote:
>>>>>>
>>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>>
>>>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>>>>> actually interoperate regardless of the codec.
>>>>>>>>
>>>>>>>
>>>>>>> The big argument for an MTI is actually the one stated in -overview:
>>>>>>> It guards against interoperability failure.
>>>>>>>
>>>>>>> I would like to have an ecosystem where one can field a box, connect
>>>>>>> it to everything else, and run well for *some* level of "well" - and I
>>>>>>> would prefer that ecosystem to be one where it's possible to field the box
>>>>>>> without making prior arrangements with anyone about licenses.
>>>>>>>
>>>>>>> This argument hasn't changed one whit since last time we discussed
>>>>>>> it. And I don't see much movement on the specifics of the proposals either.
>>>>>>>
>>>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>>>> some time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1
>>>>>>> isn't finished either. There's one sentence that needs to be removed.)
>>>>>>>
>>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>>>> prefer to re-discuss based on the knowledge that some significant players
>>>>>>> have actually changed their minds.
>>>>>>>
>>>>>>> At the moment, I don't see signs that any of the poll respondents
>>>>>>> have said "I have changed my mind".
>>>>>>>
>>>>>>> Harald
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <
>>>>>>>>> martin.thomson@gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>>>> wrote:
>>>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>>>> because
>>>>>>>>>> wasting the WG time on this again is more entertaining than
>>>>>>>>>> actually
>>>>>>>>>> finishing a specification that can be independently implemented
>>>>>>>>>> by all
>>>>>>>>>> browser vendors? (A specification that we are nowhere near
>>>>>>>>>> having, as far as
>>>>>>>>>> I can tell)
>>>>>>>>>>
>>>>>>>>> Personally, I've found the reprieve from this fight refreshing.
>>>>>>>>> And
>>>>>>>>> it would appear that we've made some real progress as a result.
>>>>>>>>> I'd
>>>>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>>>>> our time productively.  If we can't find topics to occupy our
>>>>>>>>> meeting
>>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>>
>>>>>> ------------------------------------------------------------------------------------
>>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>>> President - CoSMo Software, Cambridge, MA
>>>>>>
>>>>>> ------------------------------------------------------------------------------------
>>>>>> sg.linkedin.com/agouaillard
>>>>>>
>>>>>>    -
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> rtcweb mailing list
>>>>>> rtcweb@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Jonathan Rosenberg, Ph.D.
>>>>> jdrosen@jdrosen.net
>>>>> http://www.jdrosen.net
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>
>>>
>>>
>>> --
>>> Alex. Gouaillard, PhD, PhD, MBA
>>>
>>> ------------------------------------------------------------------------------------
>>> CTO - Temasys Communications, S'pore / Mountain View
>>> President - CoSMo Software, Cambridge, MA
>>>
>>> ------------------------------------------------------------------------------------
>>> sg.linkedin.com/agouaillard
>>>
>>>    -
>>>
>>>
>>
>>
>> --
>> Jonathan Rosenberg, Ph.D.
>> jdrosen@jdrosen.net
>> http://www.jdrosen.net
>>
>
>
>
> --
> Alex. Gouaillard, PhD, PhD, MBA
>
> ------------------------------------------------------------------------------------
> CTO - Temasys Communications, S'pore / Mountain View
> President - CoSMo Software, Cambridge, MA
>
> ------------------------------------------------------------------------------------
> sg.linkedin.com/agouaillard
>
>    -
>
>

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<div dir=3D"ltr">







<p class=3D""><span class=3D"">Alex said:</span></p>
<p class=3D""><span class=3D"">&quot;I don&#39;t think apple will make a st=
atement, but I remember reading that microsoft would support 264 (at least =
for media element playback)=C2=A0</span>if it is available on the system.=
=C2=A0 Bernard / shijun, any comment about which codec would be used for OR=
TC/WebRTC?&quot;<br><span class=3D""></span></p>
<p class=3D"">[BA] Microsoft, Apple, Cisco, Blackberry, Ericsson and others=
 have previously co-authored an H.264 MTI proposal: <a href=3D"https://tool=
s.ietf.org/html/draft-burman-rtcweb-h264-proposal">https://tools.ietf.org/h=
tml/draft-burman-rtcweb-h264-proposal</a></p>
<p class=3D""><span class=3D"">Even though that proposal expires next week,=
 we still believe it is possible for an H.264 encoder/decoder to serve as a=
 foundation for </span>interoperability between independent implementations=
. =C2=A0 However, choosing an MTI codec and specifying aspects of the codec=
 is not enough.=C2=A0 On wireless networks (bursty) packet loss is common, =
so it is important to specify transport aspects and robustness measures.=C2=
=A0 This would include an MTI FEC scheme and bandwidth estimation technique=
s, as well as use of codec-specific feedback messages.=C2=A0</p>
<p class=3D"">In terms of what needs to be done, there is work needed on th=
e rtp-usage document (such as addressing the MTI FEC scheme), =C2=A0draft-i=
etf-rtcweb-video, etc.=C2=A0 Focussing on another MTI debate at this time c=
ould make it difficult to make major progress before IETF 92.=C2=A0=C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=C2=A0</p>
<p class=3D"">If instead of tackling the hard problems, we spend our precio=
us time on yet another MTI debate, this will not satisfy customers who have=
 burned before.=C2=A0 From past experience, they know that if interoperabil=
ity issues are not addressed *early on*, patching things up after the fact =
can be difficult and time consuming.=C2=A0 Yet that is *exactly* where we a=
re headed.=C2=A0</p></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On Sun, Oct 19, 2014 at 11:46 AM, Alexandre GOUAILLARD <span dir=
=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">ago=
uaillard@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
<div dir=3D"ltr">@watson<div><br></div><div>I would prefer the ISO process =
to see through before deciding. That&#39;s why I keep bothering harald for =
news at every meeting ;-) If for any reason it does not happen, we&#39;re b=
ack where we were when we did the last pool.</div><div>Nobody has sued ....=
 yet, does not mean there is no IP risk. Actually, there were at least 3 co=
unts of nokia suing, but the three cases were negotiated and the result is =
not available to the public. (<a href=3D"http://www.fosspatents.com/2013/05=
/nokia-files-third-patent-infringement.html" target=3D"_blank">here</a>)</d=
iv><div><br></div><div>@jonathan</div><div><br></div><div>Point taken, I ha=
dn&#39;t thought about the implications of H264 hardware acceleration. If i=
ndeed it leverages the need to get a license, that makes the situation bett=
er. Would that be enough ...? We could map all the cases and see what is le=
ft to be dealt with. In any case, deciding about 264 without looking at 265=
 seems shortsighted, and I am not aware of any binary, or hardware accelera=
tion support available for 265 yet (even though iPhone 6 claims to supports=
 265 acc in its specs, I&#39;m not aware of API that would allow native app=
 to leverage this capacity).=C2=A0</div><div><br></div><div>Correct me if I=
&#39;m wrong or if I&#39;m missing anything (anyone):<br></div><div><br></d=
iv><div>-- desktop browsers --</div><div>- firefox, let&#39;s assume the 26=
4 binary is used. It also supports VP8.</div><div>- chrome on desktop suppo=
rts vp8 but not 264 (as far as webrtc is concerned, the media element suppo=
rts 264 playback).</div><div>- what about IE / Safari ? I don&#39;t think a=
pple will make a statement, but I remember reading that microsoft would sup=
port 264 (at least for media element playback) if it is available on the sy=
stem. Bernard / shijun, any comment about which codec would be used for ORT=
C/webRTC?</div><div><br></div><div>-- mobile browsers --</div><div>- no fir=
efox on iOS. What about the android version, what does it support?</div><di=
v>- chrome supports only VP8, and can&#39;t support webRTC on iOS anyway (e=
ven though justin is working on a audio-only, API-injection-in-webkit solut=
ion).</div><div>- opera?</div><div><br></div><div>-- mobile native --</div>=
<div>- iOS 8+ has system API for hw acc 264</div><div>- android has system =
API for both hw acc 264 and VP8 (is it true?)</div><div>- the <a href=3D"ht=
tp://webrtc.org" target=3D"_blank">webrtc.org</a> stack does not support 26=
4 even though kaiduan die from blackberry provided a patch for this</div><d=
iv>- the ericson&#39;s openwebrtc stack supports 264 in multiple ways (does=
 it support hw acc?) but lack support for Windows and data channel.</div><d=
iv><br></div><div>-- Others --</div><div>- lots of MCUs / SFUs / Gateways w=
ould need 264 in more cases than just the interoperability between webrtc a=
nd SIP, and I&#39;m not sure the cisco binaries can be used there.=C2=A0</d=
iv></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"=
><br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 2:13 AM, Jonathan R=
osenberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen.net" targe=
t=3D"_blank">jdrosen@jdrosen.net</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 dir=3D"ltr">@Alexandre - you say &quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">Today, nothing has changed with =
respect to those two items (even though providing open264 royalties and abs=
orbing the license cost for some platforms might have been a set in the rig=
ht direction).=C2=A0&quot;. But, as you say, the availability of Firefox wi=
th H264 is a change (previously it was not yet available); the fact that Ci=
sco has in fact fronted the cost is a change (at the last meeting some were=
 skeptical this would happen, but it has). The other big news was IOS8, whi=
ch now enables apps to access H264 and Apple pays the cost. Last time, the =
lack of a solution on IOS was a big deal. That is also now, no longer an is=
sue. As such I think there are material changes.</span><div><span style=3D"=
font-family:arial,sans-serif;font-size:13px"><br></span></div></div><div><d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19=
, 2014 at 12:13 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmail.com</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">@jonatha=
n,<div><br></div><div>while you are right and availability of 264 implement=
ation or hardware acceleration has improved, it has never been reported as =
a problem in the previous pool by anyone. The 264 royalties, and the VP8 IP=
 risks were, AFAIR, the main reasons used by individuals to justify their p=
ositions. Today, nothing has changed with respect to those two items (even =
though providing open264 royalties and absorbing the license cost for some =
platforms might have been a set in the right direction). This is why I thin=
k the conditions are not met for a consensus to be reached.</div><div><br><=
/div><div>Alex.</div><div><br></div></div><div><div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 11:43 PM, Bernar=
d Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" ta=
rget=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr"><span>&quot;And its one of the issues=
 holding up wider adoption of the technology&quot;<div><br></div></span><di=
v>[BA] Specifying an MTI encoder/decoder is not sufficient for interoperabi=
lity.=C2=A0 It is also necessary to specify the transport in enough detail =
to allow independent implementations that interoperate well enough to be us=
able in a wide variety of scenarios, including wireless networks where loss=
 is commonly experienced. =C2=A0</div><div><br></div><div>We made the mista=
ke of having an MTI discussion previously with not enough details on that s=
ubject, particularly with respect to H.264. draft-ietf-rtcweb-video section=
s 4.2 - 4.4 remain sketchy at best. =C2=A0</div><div><br></div><div>So if w=
e are to have the discussion again, it should occur in the context of detai=
led specifications and interoperability reports.</div><div><br></div><div>=
=C2=A0</div><div><br></div><div>=C2=A0</div></div><div><div><div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 8:14 AM,=
 Jonathan Rosenberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen=
.net" target=3D"_blank">jdrosen@jdrosen.net</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"><div dir=3D"ltr">I&#39;m in favor of taking anothe=
r run at this.<div><br></div><div>The working group has repeatedly said tha=
t an MTI codec is something we need to produce. And its one of the issues h=
olding up wider adoption of the technology (not the only one for sure).</di=
v><div><br></div><div>And things have changed since the last meeting, a yea=
r ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and n=
ow just recently is integrated into Firefox. iOS8 shipped with APIs for H26=
4. There are other things worth noting. Will this change the minds of every=
one? Surely not. Will it sway enough for us to achieve rough consensus? May=
be. IMHO - worth finding out.</div></div><div class=3D"gmail_extra"><div><d=
iv><br><div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:08 PM, Alexandr=
e GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.com"=
 target=3D"_blank">agouaillard@gmail.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><div dir=3D"ltr">+1 to not having MTI codec discussio=
n unless some progress has been made on VP8 at MPEG.=C2=A0Any news on that?=
 I&#39;m sharing harald&#39;s =C2=A0feeling that there was no change on the=
 members position.=C2=A0</div><div class=3D"gmail_extra"><div><div><br><div=
 class=3D"gmail_quote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <=
span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_bla=
nk">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><span>On 10/17/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdr=
osen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://ww=
w.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br></div>

--f46d04428170983c440505cc793d--


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On 10/19/2014 05:43 PM, Bernard Aboba wrote:
> "And its one of the issues holding up wider adoption of the technology"
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in
> enough detail to allow independent implementations that interoperate
> well enough to be usable in a wide variety of scenarios, including
> wireless networks where loss is commonly experienced. 

Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome,
and usage of various mechanisms for congestion control and repair of
packet loss being applied in live services.

So it can't be all bad.....

>
> We made the mistake of having an MTI discussion previously with not
> enough details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.  
>
> So if we are to have the discussion again, it should occur in the
> context of detailed specifications and interoperability reports.
>
>  
>
>  
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg
> <jdrosen@jdrosen.net <mailto:jdrosen@jdrosen.net>> wrote:
>
>     I'm in favor of taking another run at this.
>
>     The working group has repeatedly said that an MTI codec is
>     something we need to produce. And its one of the issues holding up
>     wider adoption of the technology (not the only one for sure).
>
>     And things have changed since the last meeting, a year ago now
>     (November in Vancouver). Cisco's open264 plugin shipped and now
>     just recently is integrated into Firefox. iOS8 shipped with APIs
>     for H264. There are other things worth noting. Will this change
>     the minds of everyone? Surely not. Will it sway enough for us to
>     achieve rough consensus? Maybe. IMHO - worth finding out.
>
>     On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD
>     <agouaillard@gmail.com <mailto:agouaillard@gmail.com>> wrote:
>
>         +1 to not having MTI codec discussion unless some progress has
>         been made on VP8 at MPEG. Any news on that? I'm sharing
>         harald's  feeling that there was no change on the members
>         position. 
>
>         On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand
>         <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>             On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
>                 One thing we could do instead of wasting time on MTI
>                 is to actually make progress on Sections 4.2 - 4.4 of
>                 draft-IETF-RTCWEB-video, so we could actually
>                 interoperate regardless of the codec.
>
>
>             The big argument for an MTI is actually the one stated in
>             -overview: It guards against interoperability failure.
>
>             I would like to have an ecosystem where one can field a
>             box, connect it to everything else, and run well for
>             *some* level of "well" - and I would prefer that ecosystem
>             to be one where it's possible to field the box without
>             making prior arrangements with anyone about licenses.
>
>             This argument hasn't changed one whit since last time we
>             discussed it. And I don't see much movement on the
>             specifics of the proposals either.
>
>             We'll have to interoperate well with the codecs we field.
>             So using some time to discuss draft-ietf-rtcweb-video
>             seems to make sense. (And 4.1 isn't finished either.
>             There's one sentence that needs to be removed.)
>
>             I wouldn't say I'd be happy to not discuss this in
>             Honolulu. But I'd prefer to re-discuss based on the
>             knowledge that some significant players have actually
>             changed their minds.
>
>             At the moment, I don't see signs that any of the poll
>             respondents have said "I have changed my mind".
>
>             Harald
>
>
>
>
>
>
>                     On Oct 16, 2014, at 2:28 PM, Martin Thomson
>                     <martin.thomson@gmail.com
>                     <mailto:martin.thomson@gmail.com>> wrote:
>
>                         On 16 October 2014 14:17, Matthew Kaufman
>                         <matthew@matthew.at
>                         <mailto:matthew@matthew.at>> wrote:
>                         And that's because something substantive has
>                         changed, or simply because
>                         wasting the WG time on this again is more
>                         entertaining than actually
>                         finishing a specification that can be
>                         independently implemented by all
>                         browser vendors? (A specification that we are
>                         nowhere near having, as far as
>                         I can tell)
>
>                     Personally, I've found the reprieve from this
>                     fight refreshing.  And
>                     it would appear that we've made some real progress
>                     as a result.  I'd
>                     suggest that if we don't have new information, we
>                     continue to spend
>                     our time productively.  If we can't find topics to
>                     occupy our meeting
>                     agenda time, then maybe we can free an agenda slot.
>
>                     _______________________________________________
>                     rtcweb mailing list
>                     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>                     https://www.ietf.org/mailman/listinfo/rtcweb
>
>                 _______________________________________________
>                 rtcweb mailing list
>                 rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>                 https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>             _______________________________________________
>             rtcweb mailing list
>             rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>             https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>         -- 
>         Alex. Gouaillard, PhD, PhD, MBA
>         ------------------------------------------------------------------------------------
>         CTO - Temasys Communications, S'pore / Mountain View
>         President - CoSMo Software, Cambridge, MA
>         ------------------------------------------------------------------------------------
>         sg.linkedin.com/agouaillard <http://sg.linkedin.com/agouaillard>
>
>          *
>
>
>
>         _______________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>     -- 
>     Jonathan Rosenberg, Ph.D.
>     jdrosen@jdrosen.net <mailto:jdrosen@jdrosen.net>
>     http://www.jdrosen.net
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 10/19/2014 05:43 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOW+2dugTtfLhk0VuJOk7OPEonGBApMjY93EZocH90RbX6X22w@mail.gmail.com"
      type="cite">
      <div dir="ltr">"And its one of the issues holding up wider
        adoption of the technology"
        <div><br>
        </div>
        <div>[BA] Specifying an MTI encoder/decoder is not sufficient
          for interoperability.  It is also necessary to specify the
          transport in enough detail to allow independent
          implementations that interoperate well enough to be usable in
          a wide variety of scenarios, including wireless networks where
          loss is commonly experienced.  <br>
        </div>
      </div>
    </blockquote>
    <br>
    Bernard,<br>
    <br>
    I think this is, to a large degree, codec independent.<br>
    <br>
    We have demonstrated interoperability on VP8 between Firefox and
    Chrome, and usage of various mechanisms for congestion control and
    repair of packet loss being applied in live services.<br>
    <br>
    So it can't be all bad.....<br>
    <br>
    <blockquote
cite="mid:CAOW+2dugTtfLhk0VuJOk7OPEonGBApMjY93EZocH90RbX6X22w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div><br>
        </div>
        <div>We made the mistake of having an MTI discussion previously
          with not enough details on that subject, particularly with
          respect to H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4
          remain sketchy at best.  </div>
        <div><br>
        </div>
        <div>So if we are to have the discussion again, it should occur
          in the context of detailed specifications and interoperability
          reports.</div>
        <div><br>
        </div>
        <div> </div>
        <div><br>
        </div>
        <div> </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Sun, Oct 19, 2014 at 8:14 AM,
          Jonathan Rosenberg <span dir="ltr">&lt;<a
              moz-do-not-send="true" href="mailto:jdrosen@jdrosen.net"
              target="_blank">jdrosen@jdrosen.net</a>&gt;</span> wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div dir="ltr">I'm in favor of taking another run at this.
              <div><br>
              </div>
              <div>The working group has repeatedly said that an MTI
                codec is something we need to produce. And its one of
                the issues holding up wider adoption of the technology
                (not the only one for sure).</div>
              <div><br>
              </div>
              <div>And things have changed since the last meeting, a
                year ago now (November in Vancouver). Cisco's open264
                plugin shipped and now just recently is integrated into
                Firefox. iOS8 shipped with APIs for H264. There are
                other things worth noting. Will this change the minds of
                everyone? Surely not. Will it sway enough for us to
                achieve rough consensus? Maybe. IMHO - worth finding
                out.</div>
            </div>
            <div class="gmail_extra">
              <div>
                <div class="h5"><br>
                  <div class="gmail_quote">On Sat, Oct 18, 2014 at 5:08
                    PM, Alexandre GOUAILLARD <span dir="ltr">&lt;<a
                        moz-do-not-send="true"
                        href="mailto:agouaillard@gmail.com"
                        target="_blank">agouaillard@gmail.com</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir="ltr">+1 to not having MTI codec
                        discussion unless some progress has been made on
                        VP8 at MPEG. Any news on that? I'm sharing
                        harald's  feeling that there was no change on
                        the members position. </div>
                      <div class="gmail_extra">
                        <div>
                          <div><br>
                            <div class="gmail_quote">On Fri, Oct 17,
                              2014 at 9:22 PM, Harald Alvestrand <span
                                dir="ltr">&lt;<a moz-do-not-send="true"
                                  href="mailto:harald@alvestrand.no"
                                  target="_blank">harald@alvestrand.no</a>&gt;</span>
                              wrote:<br>
                              <blockquote class="gmail_quote"
                                style="margin:0 0 0 .8ex;border-left:1px
                                #ccc solid;padding-left:1ex"><span>On
                                  10/17/2014 12:02 AM, Bernard Aboba
                                  wrote:<br>
                                  <blockquote class="gmail_quote"
                                    style="margin:0 0 0
                                    .8ex;border-left:1px #ccc
                                    solid;padding-left:1ex">
                                    One thing we could do instead of
                                    wasting time on MTI is to actually
                                    make progress on Sections 4.2 - 4.4
                                    of draft-IETF-RTCWEB-video, so we
                                    could actually interoperate
                                    regardless of the codec.<br>
                                  </blockquote>
                                  <br>
                                </span>
                                The big argument for an MTI is actually
                                the one stated in -overview: It guards
                                against interoperability failure.<br>
                                <br>
                                I would like to have an ecosystem where
                                one can field a box, connect it to
                                everything else, and run well for *some*
                                level of "well" - and I would prefer
                                that ecosystem to be one where it's
                                possible to field the box without making
                                prior arrangements with anyone about
                                licenses.<br>
                                <br>
                                This argument hasn't changed one whit
                                since last time we discussed it. And I
                                don't see much movement on the specifics
                                of the proposals either.<br>
                                <br>
                                We'll have to interoperate well with the
                                codecs we field. So using some time to
                                discuss draft-ietf-rtcweb-video seems to
                                make sense. (And 4.1 isn't finished
                                either. There's one sentence that needs
                                to be removed.)<br>
                                <br>
                                I wouldn't say I'd be happy to not
                                discuss this in Honolulu. But I'd prefer
                                to re-discuss based on the knowledge
                                that some significant players have
                                actually changed their minds.<br>
                                <br>
                                At the moment, I don't see signs that
                                any of the poll respondents have said "I
                                have changed my mind".<span><font
                                    color="#888888"><br>
                                    <br>
                                    Harald</font></span>
                                <div>
                                  <div><br>
                                    <br>
                                    <br>
                                    <blockquote class="gmail_quote"
                                      style="margin:0 0 0
                                      .8ex;border-left:1px #ccc
                                      solid;padding-left:1ex">
                                      <br>
                                      <br>
                                      <br>
                                      <blockquote class="gmail_quote"
                                        style="margin:0 0 0
                                        .8ex;border-left:1px #ccc
                                        solid;padding-left:1ex">
                                        On Oct 16, 2014, at 2:28 PM,
                                        Martin Thomson &lt;<a
                                          moz-do-not-send="true"
                                          href="mailto:martin.thomson@gmail.com"
                                          target="_blank">martin.thomson@gmail.com</a>&gt;
                                        wrote:<br>
                                        <br>
                                        <blockquote class="gmail_quote"
                                          style="margin:0 0 0
                                          .8ex;border-left:1px #ccc
                                          solid;padding-left:1ex">
                                          On 16 October 2014 14:17,
                                          Matthew Kaufman &lt;<a
                                            moz-do-not-send="true"
                                            href="mailto:matthew@matthew.at"
                                            target="_blank">matthew@matthew.at</a>&gt;
                                          wrote:<br>
                                          And that's because something
                                          substantive has changed, or
                                          simply because<br>
                                          wasting the WG time on this
                                          again is more entertaining
                                          than actually<br>
                                          finishing a specification that
                                          can be independently
                                          implemented by all<br>
                                          browser vendors? (A
                                          specification that we are
                                          nowhere near having, as far as<br>
                                          I can tell)<br>
                                        </blockquote>
                                        Personally, I've found the
                                        reprieve from this fight
                                        refreshing.  And<br>
                                        it would appear that we've made
                                        some real progress as a result. 
                                        I'd<br>
                                        suggest that if we don't have
                                        new information, we continue to
                                        spend<br>
                                        our time productively.  If we
                                        can't find topics to occupy our
                                        meeting<br>
                                        agenda time, then maybe we can
                                        free an agenda slot.<br>
                                        <br>
                                        _______________________________________________<br>
                                        rtcweb mailing list<br>
                                        <a moz-do-not-send="true"
                                          href="mailto:rtcweb@ietf.org"
                                          target="_blank">rtcweb@ietf.org</a><br>
                                        <a moz-do-not-send="true"
                                          href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                          target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                                      </blockquote>
                                      _______________________________________________<br>
                                      rtcweb mailing list<br>
                                      <a moz-do-not-send="true"
                                        href="mailto:rtcweb@ietf.org"
                                        target="_blank">rtcweb@ietf.org</a><br>
                                      <a moz-do-not-send="true"
                                        href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                        target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                                    </blockquote>
                                    <br>
                                    _______________________________________________<br>
                                    rtcweb mailing list<br>
                                    <a moz-do-not-send="true"
                                      href="mailto:rtcweb@ietf.org"
                                      target="_blank">rtcweb@ietf.org</a><br>
                                    <a moz-do-not-send="true"
                                      href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                      target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                                  </div>
                                </div>
                              </blockquote>
                            </div>
                            <br>
                            <br clear="all">
                            <div><br>
                            </div>
                          </div>
                        </div>
                        <span><font color="#888888">-- <br>
                            <div dir="ltr">Alex. Gouaillard, PhD, PhD,
                              MBA
                              <div>------------------------------------------------------------------------------------</div>
                              <div>CTO - Temasys Communications, S'pore
                                / Mountain View</div>
                              <div>President - CoSMo Software,
                                Cambridge, MA</div>
                              <div>------------------------------------------------------------------------------------</div>
                              <div><a moz-do-not-send="true"
                                  href="http://sg.linkedin.com/agouaillard"
                                  target="_blank">sg.linkedin.com/agouaillard</a></div>
                              <div>
                                <ul style="margin:0px;padding:0px 0px
8px;border:0px;outline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;line-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)">
                                  <li style="margin:0px;padding:8px 12px
                                    2px
0px;border:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-variant:inherit;line-height:1.2em">
                                    <dl
style="margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseline;font-variant:inherit;line-height:inherit;word-wrap:break-word">
                                      <br>
                                    </dl>
                                  </li>
                                </ul>
                              </div>
                            </div>
                          </font></span></div>
                      <br>
                      _______________________________________________<br>
                      rtcweb mailing list<br>
                      <a moz-do-not-send="true"
                        href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                      <a moz-do-not-send="true"
                        href="https://www.ietf.org/mailman/listinfo/rtcweb"
                        target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                      <br>
                    </blockquote>
                  </div>
                  <br>
                  <br clear="all">
                  <div><br>
                  </div>
                  -- <br>
                </div>
              </div>
              <div dir="ltr">Jonathan Rosenberg, Ph.D.<br>
                <a moz-do-not-send="true"
                  href="mailto:jdrosen@jdrosen.net" target="_blank">jdrosen@jdrosen.net</a><br>
                <a moz-do-not-send="true" href="http://www.jdrosen.net"
                  target="_blank">http://www.jdrosen.net</a></div>
            </div>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
            <br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Sun, 19 Oct 2014 14:29:05 -0700
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Harald said:

"Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome,
and usage of various mechanisms for congestion control and repair of packet
loss being applied in live services.

So it can't be all bad....."

[BA]  I agree that the lack of interoperability is largely "codec
independent":   that is, implementations based on different mechanisms for
congestion control, repair, etc. will exhibit interoperability problems,
regardless of the codec chosen.   The real test for RTCWEB will be to move
beyond "interoperability of implementations sharing source code"  to
"interoperability of independent implementations, based on open standards".


On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>  On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
> "And its one of the issues holding up wider adoption of the technology"
>
>  [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enough
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of packet
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>  We made the mistake of having an MTI discussion previously with not
> enough details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>  So if we are to have the discussion again, it should occur in the
> context of detailed specifications and interoperability reports.
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
>> I'm in favor of taking another run at this.
>>
>>  The working group has repeatedly said that an MTI codec is something we
>> need to produce. And its one of the issues holding up wider adoption of the
>> technology (not the only one for sure).
>>
>>  And things have changed since the last meeting, a year ago now
>> (November in Vancouver). Cisco's open264 plugin shipped and now just
>> recently is integrated into Firefox. iOS8 shipped with APIs for H264. There
>> are other things worth noting. Will this change the minds of everyone?
>> Surely not. Will it sway enough for us to achieve rough consensus? Maybe.
>> IMHO - worth finding out.
>>
>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>> agouaillard@gmail.com> wrote:
>>
>>> +1 to not having MTI codec discussion unless some progress has been made
>>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there
>>> was no change on the members position.
>>>
>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no
>>> > wrote:
>>>
>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>
>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>> actually interoperate regardless of the codec.
>>>>>
>>>>
>>>>  The big argument for an MTI is actually the one stated in -overview:
>>>> It guards against interoperability failure.
>>>>
>>>> I would like to have an ecosystem where one can field a box, connect it
>>>> to everything else, and run well for *some* level of "well" - and I would
>>>> prefer that ecosystem to be one where it's possible to field the box
>>>> without making prior arrangements with anyone about licenses.
>>>>
>>>> This argument hasn't changed one whit since last time we discussed it.
>>>> And I don't see much movement on the specifics of the proposals either.
>>>>
>>>> We'll have to interoperate well with the codecs we field. So using some
>>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't
>>>> finished either. There's one sentence that needs to be removed.)
>>>>
>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>> prefer to re-discuss based on the knowledge that some significant players
>>>> have actually changed their minds.
>>>>
>>>> At the moment, I don't see signs that any of the poll respondents have
>>>> said "I have changed my mind".
>>>>
>>>> Harald
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>> wrote:
>>>>>>> And that's because something substantive has changed, or simply
>>>>>>> because
>>>>>>> wasting the WG time on this again is more entertaining than actually
>>>>>>> finishing a specification that can be independently implemented by
>>>>>>> all
>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>> as far as
>>>>>>> I can tell)
>>>>>>>
>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>> our time productively.  If we can't find topics to occupy our meeting
>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>
>>>
>>>
>>>
>>>   --
>>> Alex. Gouaillard, PhD, PhD, MBA
>>>
>>> ------------------------------------------------------------------------------------
>>> CTO - Temasys Communications, S'pore / Mountain View
>>> President - CoSMo Software, Cambridge, MA
>>>
>>> ------------------------------------------------------------------------------------
>>> sg.linkedin.com/agouaillard
>>>
>>>    -
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>>
>>  --
>>  Jonathan Rosenberg, Ph.D.
>> jdrosen@jdrosen.net
>> http://www.jdrosen.net
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1134d9b2dcfedd0505cd4aba
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Harald said:=C2=A0<div><br></div><div>&quot;<span style=3D=
"font-family:arial,sans-serif;font-size:13px">Bernard,</span></div><br styl=
e=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-famil=
y:arial,sans-serif;font-size:13px">I think this is, to a large degree, code=
c independent.</span><br style=3D"font-family:arial,sans-serif;font-size:13=
px"><br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=
=3D"font-family:arial,sans-serif;font-size:13px">We have demonstrated inter=
operability on VP8 between Firefox and Chrome, and usage of various mechani=
sms for congestion control and repair of packet loss being applied in live =
services.</span><br style=3D"font-family:arial,sans-serif;font-size:13px"><=
br style=3D"font-family:arial,sans-serif;font-size:13px"><div><span style=
=3D"font-family:arial,sans-serif;font-size:13px">So it can&#39;t be all bad=
.....</span>&quot;</div><div><br></div><div>[BA] =C2=A0I agree that the lac=
k of interoperability is largely &quot;codec independent&quot;: =C2=A0 that=
 is, implementations based on different mechanisms for congestion control, =
repair, etc. will exhibit interoperability problems, regardless of the code=
c chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inte=
roperability of implementations sharing source code&quot; =C2=A0to &quot;in=
teroperability of independent implementations, based on open standards&quot=
;. =C2=A0=C2=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <span dir=3D"l=
tr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@al=
vestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
    <div>On 10/19/2014 05:43 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">&quot;And its one of the issues holding up wider
        adoption of the technology&quot;
        <div><br>
        </div>
        <div>[BA] Specifying an MTI encoder/decoder is not sufficient
          for interoperability.=C2=A0 It is also necessary to specify the
          transport in enough detail to allow independent
          implementations that interoperate well enough to be usable in
          a wide variety of scenarios, including wireless networks where
          loss is commonly experienced.=C2=A0 <br>
        </div>
      </div>
    </blockquote>
    <br></span>
    Bernard,<br>
    <br>
    I think this is, to a large degree, codec independent.<br>
    <br>
    We have demonstrated interoperability on VP8 between Firefox and
    Chrome, and usage of various mechanisms for congestion control and
    repair of packet loss being applied in live services.<br>
    <br>
    So it can&#39;t be all bad.....<div><div class=3D"h5"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div><br>
        </div>
        <div>We made the mistake of having an MTI discussion previously
          with not enough details on that subject, particularly with
          respect to H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4
          remain sketchy at best. =C2=A0</div>
        <div><br>
        </div>
        <div>So if we are to have the discussion again, it should occur
          in the context of detailed specifications and interoperability
          reports.</div>
        <div><br>
        </div>
        <div>=C2=A0</div>
        <div><br>
        </div>
        <div>=C2=A0</div>
      </div>
      <div class=3D"gmail_extra"><br>
        <div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 8:14 AM,
          Jonathan Rosenberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrose=
n@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a>&gt;</span> wrote:<=
br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
            <div dir=3D"ltr">I&#39;m in favor of taking another run at this=
.
              <div><br>
              </div>
              <div>The working group has repeatedly said that an MTI
                codec is something we need to produce. And its one of
                the issues holding up wider adoption of the technology
                (not the only one for sure).</div>
              <div><br>
              </div>
              <div>And things have changed since the last meeting, a
                year ago now (November in Vancouver). Cisco&#39;s open264
                plugin shipped and now just recently is integrated into
                Firefox. iOS8 shipped with APIs for H264. There are
                other things worth noting. Will this change the minds of
                everyone? Surely not. Will it sway enough for us to
                achieve rough consensus? Maybe. IMHO - worth finding
                out.</div>
            </div>
            <div class=3D"gmail_extra">
              <div>
                <div><br>
                  <div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:08
                    PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=
=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmail.com</=
a>&gt;</span>
                    wrote:<br>
                    <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir=3D"ltr">+1 to not having MTI codec
                        discussion unless some progress has been made on
                        VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing
                        harald&#39;s =C2=A0feeling that there was no change=
 on
                        the members position.=C2=A0</div>
                      <div class=3D"gmail_extra">
                        <div>
                          <div><br>
                            <div class=3D"gmail_quote">On Fri, Oct 17,
                              2014 at 9:22 PM, Harald Alvestrand <span dir=
=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">hara=
ld@alvestrand.no</a>&gt;</span>
                              wrote:<br>
                              <blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On
                                  10/17/2014 12:02 AM, Bernard Aboba
                                  wrote:<br>
                                  <blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                                    One thing we could do instead of
                                    wasting time on MTI is to actually
                                    make progress on Sections 4.2 - 4.4
                                    of draft-IETF-RTCWEB-video, so we
                                    could actually interoperate
                                    regardless of the codec.<br>
                                  </blockquote>
                                  <br>
                                </span>
                                The big argument for an MTI is actually
                                the one stated in -overview: It guards
                                against interoperability failure.<br>
                                <br>
                                I would like to have an ecosystem where
                                one can field a box, connect it to
                                everything else, and run well for *some*
                                level of &quot;well&quot; - and I would pre=
fer
                                that ecosystem to be one where it&#39;s
                                possible to field the box without making
                                prior arrangements with anyone about
                                licenses.<br>
                                <br>
                                This argument hasn&#39;t changed one whit
                                since last time we discussed it. And I
                                don&#39;t see much movement on the specific=
s
                                of the proposals either.<br>
                                <br>
                                We&#39;ll have to interoperate well with th=
e
                                codecs we field. So using some time to
                                discuss draft-ietf-rtcweb-video seems to
                                make sense. (And 4.1 isn&#39;t finished
                                either. There&#39;s one sentence that needs
                                to be removed.)<br>
                                <br>
                                I wouldn&#39;t say I&#39;d be happy to not
                                discuss this in Honolulu. But I&#39;d prefe=
r
                                to re-discuss based on the knowledge
                                that some significant players have
                                actually changed their minds.<br>
                                <br>
                                At the moment, I don&#39;t see signs that
                                any of the poll respondents have said &quot=
;I
                                have changed my mind&quot;.<span><font colo=
r=3D"#888888"><br>
                                    <br>
                                    Harald</font></span>
                                <div>
                                  <div><br>
                                    <br>
                                    <br>
                                    <blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                                      <br>
                                      <br>
                                      <br>
                                      <blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                                        On Oct 16, 2014, at 2:28 PM,
                                        Martin Thomson &lt;<a href=3D"mailt=
o:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&=
gt;
                                        wrote:<br>
                                        <br>
                                        <blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                                          On 16 October 2014 14:17,
                                          Matthew Kaufman &lt;<a href=3D"ma=
ilto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt;
                                          wrote:<br>
                                          And that&#39;s because something
                                          substantive has changed, or
                                          simply because<br>
                                          wasting the WG time on this
                                          again is more entertaining
                                          than actually<br>
                                          finishing a specification that
                                          can be independently
                                          implemented by all<br>
                                          browser vendors? (A
                                          specification that we are
                                          nowhere near having, as far as<br=
>
                                          I can tell)<br>
                                        </blockquote>
                                        Personally, I&#39;ve found the
                                        reprieve from this fight
                                        refreshing.=C2=A0 And<br>
                                        it would appear that we&#39;ve made
                                        some real progress as a result.=C2=
=A0
                                        I&#39;d<br>
                                        suggest that if we don&#39;t have
                                        new information, we continue to
                                        spend<br>
                                        our time productively.=C2=A0 If we
                                        can&#39;t find topics to occupy our
                                        meeting<br>
                                        agenda time, then maybe we can
                                        free an agenda slot.<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/mai=
lman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listin=
fo/rtcweb</a><br>
                                      </blockquote>
                                      _____________________________________=
__________<br>
                                      rtcweb mailing list<br>
                                      <a href=3D"mailto:rtcweb@ietf.org" ta=
rget=3D"_blank">rtcweb@ietf.org</a><br>
                                      <a href=3D"https://www.ietf.org/mailm=
an/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo=
/rtcweb</a><br>
                                    </blockquote>
                                    <br>
                                    _______________________________________=
________<br>
                                    rtcweb mailing list<br>
                                    <a href=3D"mailto:rtcweb@ietf.org" targ=
et=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/r=
tcweb</a><br>
                                  </div>
                                </div>
                              </blockquote>
                            </div>
                            <br>
                            <br clear=3D"all">
                            <div><br>
                            </div>
                          </div>
                        </div>
                        <span><font color=3D"#888888">-- <br>
                            <div dir=3D"ltr">Alex. Gouaillard, PhD, PhD,
                              MBA
                              <div>----------------------------------------=
--------------------------------------------</div>
                              <div>CTO - Temasys Communications, S&#39;pore
                                / Mountain View</div>
                              <div>President - CoSMo Software,
                                Cambridge, MA</div>
                              <div>----------------------------------------=
--------------------------------------------</div>
                              <div><a href=3D"http://sg.linkedin.com/agouai=
llard" target=3D"_blank">sg.linkedin.com/agouaillard</a></div>
                              <div>
                                <ul style=3D"margin:0px;padding:0px 0px 8px=
;border:0px;outline:0px;font-size:12px;font-family:Helvetica,Arial,sans-ser=
if;vertical-align:baseline;list-style:none;line-height:17px;display:table-c=
ell;width:504px;color:rgb(51,51,51)">
                                  <li style=3D"margin:0px;padding:8px 12px =
2px 0px;border:0px;outline:0px;font-style:inherit;font-size:11px;font-famil=
y:inherit;vertical-align:baseline;font-variant:inherit;line-height:1.2em">
                                    <dl style=3D"margin:0px;padding:0px;bor=
der:0px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:b=
aseline;font-variant:inherit;line-height:inherit;word-wrap:break-word">
                                      <br>
                                    </dl>
                                  </li>
                                </ul>
                              </div>
                            </div>
                          </font></span></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/rtcw=
eb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                      <br>
                    </blockquote>
                  </div>
                  <br>
                  <br clear=3D"all">
                  <div><br>
                  </div>
                  -- <br>
                </div>
              </div>
              <div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br>
                <a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jd=
rosen@jdrosen.net</a><br>
                <a href=3D"http://www.jdrosen.net" target=3D"_blank">http:/=
/www.jdrosen.net</a></div>
            </div>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@iet=
f.org</a><br>
            <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
            <br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    </div></div><span class=3D"HOEnZb"><font color=3D"#888888"><pre cols=3D=
"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
  </font></span></div>

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

--001a1134d9b2dcfedd0505cd4aba--


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On 10/16/14, 11:10 AM, Ted Hardie wrote:
> During this morning chairs call, we discussed the agenda for the
> upcoming meeting.  The two big items are JSEP and the video codec
> discussion, and our current thought is to have the JSEP discussion on
> Monday and the video codec discussion on Thursday.

As far as I can see, the temper of the list has been that it would be 
premature and a waste of energy to discuss the video codec topic at IETF 
91. Do the chairs see things differently?

Peter

-- 
Peter Saint-Andre
https://andyet.com/


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From: Stephan Wenger <stewe@stewe.org>
To: Watson Ladd <watsonbladd@gmail.com>, Alexandre GOUAILLARD <agouaillard@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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Subject: Re: [rtcweb] Plan for MTI video codec?
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Hi,

On 10/19/14, 9:15 AM, "Watson Ladd" <watsonbladd@gmail.com> wrote:

>On Sun, Oct 19, 2014 at 9:13 AM, Alexandre GOUAILLARD
><agouaillard@gmail.com> wrote:
>> @jonathan,
>>
>> while you are right and availability of 264 implementation or hardware
>> acceleration has improved, it has never been reported as a problem in
>>the
>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
>> AFAIR, the main reasons used by individuals to justify their positions.
>> Today, nothing has changed with respect to those two items (even though
>> providing open264 royalties and absorbing the license cost for some
>> platforms might have been a set in the right direction). This is why I
>>think
>> the conditions are not met for a consensus to be reached.
>
>But now VP8 is going through ISO,

... and is is DIS ballot.  Few projects in ISO get stopped at that stage.
To me, it=B9s pretty clear that VP8 will have an ISO/IEC blessing within a
year or two.  Without substantial technical changes.  Given the very
limited participation in the relevant subgroup in MPEG, it=B9s unclear to m=
e
what good that will do, though.

>and the people claiming patents on
>VP8 have had time to sue, and haven't.

That=B9s factually incorrect.  To the best of my knowledge, what would be
factually correct is this: in two cases, companies have been sued over
patents allegedly reading on VP8 in the context of the wider =B3smartphone
wars=B2 lawsuits, and the defendants have won non-infringement rulings in
the first instance (though, last I looked, appeals were pending in both
cases).  At least one other case was settled on undisclosed terms.  Some
of these cases were widely reported in the press, others are a little bit
harder to find without access to legal search tools.

>That's evidence that some
>concerns are overblown.

And that depends on your viewpoint.

Stephan

>
>>
>> Alex.
>>
>>
>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba
>><bernard.aboba@gmail.com>
>> wrote:
>>>
>>> "And its one of the issues holding up wider adoption of the technology"
>>>
>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>>> interoperability.  It is also necessary to specify the transport in
>>>enough
>>> detail to allow independent implementations that interoperate well
>>>enough to
>>> be usable in a wide variety of scenarios, including wireless networks
>>>where
>>> loss is commonly experienced.
>>>
>>> We made the mistake of having an MTI discussion previously with not
>>>enough
>>> details on that subject, particularly with respect to H.264.
>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>>
>>> So if we are to have the discussion again, it should occur in the
>>>context
>>> of detailed specifications and interoperability reports.
>>>
>>>
>>>
>>>
>>>
>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg
>>><jdrosen@jdrosen.net>
>>> wrote:
>>>>
>>>> I'm in favor of taking another run at this.
>>>>
>>>> The working group has repeatedly said that an MTI codec is something
>>>>we
>>>> need to produce. And its one of the issues holding up wider adoption
>>>>of the
>>>> technology (not the only one for sure).
>>>>
>>>> And things have changed since the last meeting, a year ago now
>>>>(November
>>>> in Vancouver). Cisco's open264 plugin shipped and now just recently is
>>>> integrated into Firefox. iOS8 shipped with APIs for H264. There are
>>>>other
>>>> things worth noting. Will this change the minds of everyone? Surely
>>>>not.
>>>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO -
>>>>worth
>>>> finding out.
>>>>
>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD
>>>> <agouaillard@gmail.com> wrote:
>>>>>
>>>>> +1 to not having MTI codec discussion unless some progress has been
>>>>>made
>>>>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>>>there
>>>>> was no change on the members position.
>>>>>
>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand
>>>>> <harald@alvestrand.no> wrote:
>>>>>>
>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>>
>>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so
>>>>>>>we could
>>>>>>> actually interoperate regardless of the codec.
>>>>>>
>>>>>>
>>>>>> The big argument for an MTI is actually the one stated in
>>>>>>-overview: It
>>>>>> guards against interoperability failure.
>>>>>>
>>>>>> I would like to have an ecosystem where one can field a box,
>>>>>>connect it
>>>>>> to everything else, and run well for *some* level of "well" - and I
>>>>>>would
>>>>>> prefer that ecosystem to be one where it's possible to field the
>>>>>>box without
>>>>>> making prior arrangements with anyone about licenses.
>>>>>>
>>>>>> This argument hasn't changed one whit since last time we discussed
>>>>>>it.
>>>>>> And I don't see much movement on the specifics of the proposals
>>>>>>either.
>>>>>>
>>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>>>some
>>>>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And
>>>>>>4.1 isn't
>>>>>> finished either. There's one sentence that needs to be removed.)
>>>>>>
>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>>> prefer to re-discuss based on the knowledge that some significant
>>>>>>players
>>>>>> have actually changed their minds.
>>>>>>
>>>>>> At the moment, I don't see signs that any of the poll respondents
>>>>>>have
>>>>>> said "I have changed my mind".
>>>>>>
>>>>>> Harald
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> On Oct 16, 2014, at 2:28 PM, Martin Thomson
>>>>>>>> <martin.thomson@gmail.com> wrote:
>>>>>>>>
>>>>>>>>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>>> wrote:
>>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>>> because
>>>>>>>>> wasting the WG time on this again is more entertaining than
>>>>>>>>>actually
>>>>>>>>> finishing a specification that can be independently implemented
>>>>>>>>>by
>>>>>>>>> all
>>>>>>>>> browser vendors? (A specification that we are nowhere near
>>>>>>>>>having,
>>>>>>>>> as far as
>>>>>>>>> I can tell)
>>>>>>>>
>>>>>>>> Personally, I've found the reprieve from this fight refreshing.
>>>>>>>>And
>>>>>>>> it would appear that we've made some real progress as a result.
>>>>>>>>I'd
>>>>>>>> suggest that if we don't have new information, we continue to
>>>>>>>>spend
>>>>>>>> our time productively.  If we can't find topics to occupy our
>>>>>>>>meeting
>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>
>>>>>=20
>>>>>----------------------------------------------------------------------
>>>>>--------------
>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>> President - CoSMo Software, Cambridge, MA
>>>>>
>>>>>=20
>>>>>----------------------------------------------------------------------
>>>>>--------------
>>>>> sg.linkedin.com/agouaillard
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Jonathan Rosenberg, Ph.D.
>>>> jdrosen@jdrosen.net
>>>> http://www.jdrosen.net
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>
>>
>>
>> --
>> Alex. Gouaillard, PhD, PhD, MBA
>>=20
>>-------------------------------------------------------------------------
>>-----------
>> CTO - Temasys Communications, S'pore / Mountain View
>> President - CoSMo Software, Cambridge, MA
>>=20
>>-------------------------------------------------------------------------
>>-----------
>> sg.linkedin.com/agouaillard
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>
>--=20
>"Those who would give up Essential Liberty to purchase a little
>Temporary Safety deserve neither  Liberty nor Safety."
>-- Benjamin Franklin
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: Harald Alvestrand <harald@alvestrand.no>, Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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Subject: [rtcweb] FEC -- was Re:  Plan for MTI video codec?
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On Sunday, October 19, 2014, Bernard Aboba <bernard.aboba@gmail.com> wrote:

> Alex said:
>
> "I don't think apple will make a statement, but I remember reading that
> microsoft would support 264 (at least for media element playback) if it
> is available on the system.  Bernard / shijun, any comment about which
> codec would be used for ORTC/WebRTC?"
>
> [BA] Microsoft, Apple, Cisco, Blackberry, Ericsson and others have
> previously co-authored an H.264 MTI proposal:
> https://tools.ietf.org/html/draft-burman-rtcweb-h264-proposal
>
> Even though that proposal expires next week, we still believe it is
> possible for an H.264 encoder/decoder to serve as a foundation for interoperability
> between independent implementations.   However, choosing an MTI codec and
> specifying aspects of the codec is not enough.  On wireless networks
> (bursty) packet loss is common, so it is important to specify transport
> aspects and robustness
>
> Bernard,

Regarding packet loss in wireless networks and FEC,  can you be more
specific?

It would be best if this problem was clearly articulated. Afaik, various
flavors of 802.11 as well as UMTS and LTE all support L2 FEC. And, in the
case of UMTS and LTE , all packets are retransmitted until the end point
acknowledges the packet has been received.

That said, it would be imprudent to put those functions again into L4 or
L7.

CB

> measures.  This would include an MTI FEC scheme and bandwidth estimation
> techniques, as well as use of codec-specific feedback messages.
>
> In terms of what needs to be done, there is work needed on the rtp-usage
> document (such as addressing the MTI FEC scheme),  draft-ietf-rtcweb-video,
> etc.  Focussing on another MTI debate at this time could make it difficult
> to make major progress before IETF 92.
>
>
> If instead of tackling the hard problems, we spend our precious time on
> yet another MTI debate, this will not satisfy customers who have burned
> before.  From past experience, they know that if interoperability issues
> are not addressed *early on*, patching things up after the fact can be
> difficult and time consuming.  Yet that is *exactly* where we are headed.
>
> On Sun, Oct 19, 2014 at 11:46 AM, Alexandre GOUAILLARD <
> agouaillard@gmail.com
> <javascript:_e(%7B%7D,'cvml','agouaillard@gmail.com');>> wrote:
>
>> @watson
>>
>> I would prefer the ISO process to see through before deciding. That's why
>> I keep bothering harald for news at every meeting ;-) If for any reason it
>> does not happen, we're back where we were when we did the last pool.
>> Nobody has sued .... yet, does not mean there is no IP risk. Actually,
>> there were at least 3 counts of nokia suing, but the three cases were
>> negotiated and the result is not available to the public. (here
>> <http://www.fosspatents.com/2013/05/nokia-files-third-patent-infringement.html>
>> )
>>
>> @jonathan
>>
>> Point taken, I hadn't thought about the implications of H264 hardware
>> acceleration. If indeed it leverages the need to get a license, that makes
>> the situation better. Would that be enough ...? We could map all the cases
>> and see what is left to be dealt with. In any case, deciding about 264
>> without looking at 265 seems shortsighted, and I am not aware of any
>> binary, or hardware acceleration support available for 265 yet (even though
>> iPhone 6 claims to supports 265 acc in its specs, I'm not aware of API that
>> would allow native app to leverage this capacity).
>>
>> Correct me if I'm wrong or if I'm missing anything (anyone):
>>
>> -- desktop browsers --
>> - firefox, let's assume the 264 binary is used. It also supports VP8.
>> - chrome on desktop supports vp8 but not 264 (as far as webrtc is
>> concerned, the media element supports 264 playback).
>> - what about IE / Safari ? I don't think apple will make a statement, but
>> I remember reading that microsoft would support 264 (at least for media
>> element playback) if it is available on the system. Bernard / shijun, any
>> comment about which codec would be used for ORTC/webRTC?
>>
>> -- mobile browsers --
>> - no firefox on iOS. What about the android version, what does it support?
>> - chrome supports only VP8, and can't support webRTC on iOS anyway (even
>> though justin is working on a audio-only, API-injection-in-webkit solution).
>> - opera?
>>
>> -- mobile native --
>> - iOS 8+ has system API for hw acc 264
>> - android has system API for both hw acc 264 and VP8 (is it true?)
>> - the webrtc.org stack does not support 264 even though kaiduan die from
>> blackberry provided a patch for this
>> - the ericson's openwebrtc stack supports 264 in multiple ways (does it
>> support hw acc?) but lack support for Windows and data channel.
>>
>> -- Others --
>> - lots of MCUs / SFUs / Gateways would need 264 in more cases than just
>> the interoperability between webrtc and SIP, and I'm not sure the cisco
>> binaries can be used there.
>>
>> On Mon, Oct 20, 2014 at 2:13 AM, Jonathan Rosenberg <jdrosen@jdrosen.net
>> <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>> wrote:
>>
>>> @Alexandre - you say "Today, nothing has changed with respect to those
>>> two items (even though providing open264 royalties and absorbing the
>>> license cost for some platforms might have been a set in the right
>>> direction). ". But, as you say, the availability of Firefox with H264 is a
>>> change (previously it was not yet available); the fact that Cisco has in
>>> fact fronted the cost is a change (at the last meeting some were skeptical
>>> this would happen, but it has). The other big news was IOS8, which now
>>> enables apps to access H264 and Apple pays the cost. Last time, the lack of
>>> a solution on IOS was a big deal. That is also now, no longer an issue. As
>>> such I think there are material changes.
>>>
>>>
>>> On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <
>>> agouaillard@gmail.com
>>> <javascript:_e(%7B%7D,'cvml','agouaillard@gmail.com');>> wrote:
>>>
>>>> @jonathan,
>>>>
>>>> while you are right and availability of 264 implementation or hardware
>>>> acceleration has improved, it has never been reported as a problem in the
>>>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
>>>> AFAIR, the main reasons used by individuals to justify their positions.
>>>> Today, nothing has changed with respect to those two items (even though
>>>> providing open264 royalties and absorbing the license cost for some
>>>> platforms might have been a set in the right direction). This is why I
>>>> think the conditions are not met for a consensus to be reached.
>>>>
>>>> Alex.
>>>>
>>>>
>>>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <
>>>> bernard.aboba@gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','bernard.aboba@gmail.com');>> wrote:
>>>>
>>>>> "And its one of the issues holding up wider adoption of the technology"
>>>>>
>>>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>>>>> interoperability.  It is also necessary to specify the transport in enough
>>>>> detail to allow independent implementations that interoperate well enough
>>>>> to be usable in a wide variety of scenarios, including wireless networks
>>>>> where loss is commonly experienced.
>>>>>
>>>>> We made the mistake of having an MTI discussion previously with not
>>>>> enough details on that subject, particularly with respect to H.264.
>>>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>>>>
>>>>> So if we are to have the discussion again, it should occur in the
>>>>> context of detailed specifications and interoperability reports.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <
>>>>> jdrosen@jdrosen.net
>>>>> <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>> wrote:
>>>>>
>>>>>> I'm in favor of taking another run at this.
>>>>>>
>>>>>> The working group has repeatedly said that an MTI codec is something
>>>>>> we need to produce. And its one of the issues holding up wider adoption of
>>>>>> the technology (not the only one for sure).
>>>>>>
>>>>>> And things have changed since the last meeting, a year ago now
>>>>>> (November in Vancouver). Cisco's open264 plugin shipped and now just
>>>>>> recently is integrated into Firefox. iOS8 shipped with APIs for H264. There
>>>>>> are other things worth noting. Will this change the minds of everyone?
>>>>>> Surely not. Will it sway enough for us to achieve rough consensus? Maybe.
>>>>>> IMHO - worth finding out.
>>>>>>
>>>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>>>>>> agouaillard@gmail.com
>>>>>> <javascript:_e(%7B%7D,'cvml','agouaillard@gmail.com');>> wrote:
>>>>>>
>>>>>>> +1 to not having MTI codec discussion unless some progress has been
>>>>>>> made on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>>>>> there was no change on the members position.
>>>>>>>
>>>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <
>>>>>>> harald@alvestrand.no
>>>>>>> <javascript:_e(%7B%7D,'cvml','harald@alvestrand.no');>> wrote:
>>>>>>>
>>>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>>>
>>>>>>>>> One thing we could do instead of wasting time on MTI is to
>>>>>>>>> actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so
>>>>>>>>> we could actually interoperate regardless of the codec.
>>>>>>>>>
>>>>>>>>
>>>>>>>> The big argument for an MTI is actually the one stated in
>>>>>>>> -overview: It guards against interoperability failure.
>>>>>>>>
>>>>>>>> I would like to have an ecosystem where one can field a box,
>>>>>>>> connect it to everything else, and run well for *some* level of "well" -
>>>>>>>> and I would prefer that ecosystem to be one where it's possible to field
>>>>>>>> the box without making prior arrangements with anyone about licenses.
>>>>>>>>
>>>>>>>> This argument hasn't changed one whit since last time we discussed
>>>>>>>> it. And I don't see much movement on the specifics of the proposals either.
>>>>>>>>
>>>>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>>>>> some time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1
>>>>>>>> isn't finished either. There's one sentence that needs to be removed.)
>>>>>>>>
>>>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But
>>>>>>>> I'd prefer to re-discuss based on the knowledge that some significant
>>>>>>>> players have actually changed their minds.
>>>>>>>>
>>>>>>>> At the moment, I don't see signs that any of the poll respondents
>>>>>>>> have said "I have changed my mind".
>>>>>>>>
>>>>>>>> Harald
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <
>>>>>>>>>> martin.thomson@gmail.com
>>>>>>>>>> <javascript:_e(%7B%7D,'cvml','martin.thomson@gmail.com');>>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at
>>>>>>>>>>> <javascript:_e(%7B%7D,'cvml','matthew@matthew.at');>> wrote:
>>>>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>>>>> because
>>>>>>>>>>> wasting the WG time on this again is more entertaining than
>>>>>>>>>>> actually
>>>>>>>>>>> finishing a specification that can be independently implemented
>>>>>>>>>>> by all
>>>>>>>>>>> browser vendors? (A specification that we are nowhere near
>>>>>>>>>>> having, as far as
>>>>>>>>>>> I can tell)
>>>>>>>>>>>
>>>>>>>>>> Personally, I've found the reprieve from this fight refreshing.
>>>>>>>>>> And
>>>>>>>>>> it would appear that we've made some real progress as a result.
>>>>>>>>>> I'd
>>>>>>>>>> suggest that if we don't have new information, we continue to
>>>>>>>>>> spend
>>>>>>>>>> our time productively.  If we can't find topics to occupy our
>>>>>>>>>> meeting
>>>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> rtcweb mailing list
>>>>>>>>>> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> rtcweb mailing list
>>>>>>>>> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>>>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> rtcweb mailing list
>>>>>>>> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>>>
>>>>>>> ------------------------------------------------------------------------------------
>>>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>>>> President - CoSMo Software, Cambridge, MA
>>>>>>>
>>>>>>> ------------------------------------------------------------------------------------
>>>>>>> sg.linkedin.com/agouaillard
>>>>>>>
>>>>>>>    -
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> rtcweb mailing list
>>>>>>> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Jonathan Rosenberg, Ph.D.
>>>>>> jdrosen@jdrosen.net
>>>>>> <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>
>>>>>> http://www.jdrosen.net
>>>>>>
>>>>>> _______________________________________________
>>>>>> rtcweb mailing list
>>>>>> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>> President - CoSMo Software, Cambridge, MA
>>>>
>>>> ------------------------------------------------------------------------------------
>>>> sg.linkedin.com/agouaillard
>>>>
>>>>    -
>>>>
>>>>
>>>
>>>
>>> --
>>> Jonathan Rosenberg, Ph.D.
>>> jdrosen@jdrosen.net
>>> <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>
>>> http://www.jdrosen.net
>>>
>>
>>
>>
>> --
>> Alex. Gouaillard, PhD, PhD, MBA
>>
>> ------------------------------------------------------------------------------------
>> CTO - Temasys Communications, S'pore / Mountain View
>> President - CoSMo Software, Cambridge, MA
>>
>> ------------------------------------------------------------------------------------
>> sg.linkedin.com/agouaillard
>>
>>    -
>>
>>
>

--047d7bea3de65e33ab0505d0bc69
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<br><br>On Sunday, October 19, 2014, Bernard Aboba &lt;<a href=3D"mailto:be=
rnard.aboba@gmail.com">bernard.aboba@gmail.com</a>&gt; wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr">







<p><span>Alex said:</span></p>
<p><span>&quot;I don&#39;t think apple will make a statement, but I remembe=
r reading that microsoft would support 264 (at least for media element play=
back)=C2=A0</span>if it is available on the system.=C2=A0 Bernard / shijun,=
 any comment about which codec would be used for ORTC/WebRTC?&quot;<br><spa=
n></span></p>
<p>[BA] Microsoft, Apple, Cisco, Blackberry, Ericsson and others have previ=
ously co-authored an H.264 MTI proposal: <a href=3D"https://tools.ietf.org/=
html/draft-burman-rtcweb-h264-proposal" target=3D"_blank">https://tools.iet=
f.org/html/draft-burman-rtcweb-h264-proposal</a></p>
<p><span>Even though that proposal expires next week, we still believe it i=
s possible for an H.264 encoder/decoder to serve as a foundation for </span=
>interoperability between independent implementations. =C2=A0 However, choo=
sing an MTI codec and specifying aspects of the codec is not enough.=C2=A0 =
On wireless networks (bursty) packet loss is common, so it is important to =
specify transport aspects and robustness=C2=A0</p><p></p></div></blockquote=
><div>Bernard,</div><div><br></div><div>Regarding packet loss in wireless n=
etworks=C2=A0and FEC,=C2=A0 can you be more specific?</div><div><br></div><=
div>It would be best if this problem was clearly articulated. Afaik, variou=
s flavors of=C2=A0802.11 as well as UMTS and LTE all support L2 FEC. And, i=
n the case of UMTS and LTE ,=C2=A0all packets are retransmitted until the e=
nd point acknowledges the packet has been received.=C2=A0</div><div><br></d=
iv><div>That said, it would be imprudent to put those functions again into =
L4 or L7.=C2=A0</div><div><br></div><div>CB</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex"><div dir=3D"ltr"><p>measures.=C2=A0 This would include an MTI FEC sche=
me and bandwidth estimation techniques, as well as use of codec-specific fe=
edback messages.=C2=A0</p>
<p>In terms of what needs to be done, there is work needed on the rtp-usage=
 document (such as addressing the MTI FEC scheme), =C2=A0draft-ietf-rtcweb-=
video, etc.=C2=A0 Focussing on another MTI debate at this time could make i=
t difficult to make major progress before IETF 92.=C2=A0=C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=C2=A0</p>
<p>If instead of tackling the hard problems, we spend our precious time on =
yet another MTI debate, this will not satisfy customers who have burned bef=
ore.=C2=A0 From past experience, they know that if interoperability issues =
are not addressed *early on*, patching things up after the fact can be diff=
icult and time consuming.=C2=A0 Yet that is *exactly* where we are headed.=
=C2=A0</p></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Sun, Oct 19, 2014 at 11:46 AM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt=
;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;agouaillard@gmail.com&=
#39;);" target=3D"_blank">agouaillard@gmail.com</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 dir=3D"ltr">@watson<div><br></div><div>I =
would prefer the ISO process to see through before deciding. That&#39;s why=
 I keep bothering harald for news at every meeting ;-) If for any reason it=
 does not happen, we&#39;re back where we were when we did the last pool.</=
div><div>Nobody has sued .... yet, does not mean there is no IP risk. Actua=
lly, there were at least 3 counts of nokia suing, but the three cases were =
negotiated and the result is not available to the public. (<a href=3D"http:=
//www.fosspatents.com/2013/05/nokia-files-third-patent-infringement.html" t=
arget=3D"_blank">here</a>)</div><div><br></div><div>@jonathan</div><div><br=
></div><div>Point taken, I hadn&#39;t thought about the implications of H26=
4 hardware acceleration. If indeed it leverages the need to get a license, =
that makes the situation better. Would that be enough ...? We could map all=
 the cases and see what is left to be dealt with. In any case, deciding abo=
ut 264 without looking at 265 seems shortsighted, and I am not aware of any=
 binary, or hardware acceleration support available for 265 yet (even thoug=
h iPhone 6 claims to supports 265 acc in its specs, I&#39;m not aware of AP=
I that would allow native app to leverage this capacity).=C2=A0</div><div><=
br></div><div>Correct me if I&#39;m wrong or if I&#39;m missing anything (a=
nyone):<br></div><div><br></div><div>-- desktop browsers --</div><div>- fir=
efox, let&#39;s assume the 264 binary is used. It also supports VP8.</div><=
div>- chrome on desktop supports vp8 but not 264 (as far as webrtc is conce=
rned, the media element supports 264 playback).</div><div>- what about IE /=
 Safari ? I don&#39;t think apple will make a statement, but I remember rea=
ding that microsoft would support 264 (at least for media element playback)=
 if it is available on the system. Bernard / shijun, any comment about whic=
h codec would be used for ORTC/webRTC?</div><div><br></div><div>-- mobile b=
rowsers --</div><div>- no firefox on iOS. What about the android version, w=
hat does it support?</div><div>- chrome supports only VP8, and can&#39;t su=
pport webRTC on iOS anyway (even though justin is working on a audio-only, =
API-injection-in-webkit solution).</div><div>- opera?</div><div><br></div><=
div>-- mobile native --</div><div>- iOS 8+ has system API for hw acc 264</d=
iv><div>- android has system API for both hw acc 264 and VP8 (is it true?)<=
/div><div>- the <a href=3D"http://webrtc.org" target=3D"_blank">webrtc.org<=
/a> stack does not support 264 even though kaiduan die from blackberry prov=
ided a patch for this</div><div>- the ericson&#39;s openwebrtc stack suppor=
ts 264 in multiple ways (does it support hw acc?) but lack support for Wind=
ows and data channel.</div><div><br></div><div>-- Others --</div><div>- lot=
s of MCUs / SFUs / Gateways would need 264 in more cases than just the inte=
roperability between webrtc and SIP, and I&#39;m not sure the cisco binarie=
s can be used there.=C2=A0</div></div><div><div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 2:13 AM, Jonathan Ro=
senberg <span dir=3D"ltr">&lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39=
;,&#39;jdrosen@jdrosen.net&#39;);" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">@Ale=
xandre - you say &quot;<span style=3D"font-family:arial,sans-serif;font-siz=
e:13px">Today, nothing has changed with respect to those two items (even th=
ough providing open264 royalties and absorbing the license cost for some pl=
atforms might have been a set in the right direction).=C2=A0&quot;. But, as=
 you say, the availability of Firefox with H264 is a change (previously it =
was not yet available); the fact that Cisco has in fact fronted the cost is=
 a change (at the last meeting some were skeptical this would happen, but i=
t has). The other big news was IOS8, which now enables apps to access H264 =
and Apple pays the cost. Last time, the lack of a solution on IOS was a big=
 deal. That is also now, no longer an issue. As such I think there are mate=
rial changes.</span><div><span style=3D"font-family:arial,sans-serif;font-s=
ize:13px"><br></span></div></div><div><div><div class=3D"gmail_extra"><br><=
div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAI=
LLARD <span dir=3D"ltr">&lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,=
&#39;agouaillard@gmail.com&#39;);" target=3D"_blank">agouaillard@gmail.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">@j=
onathan,<div><br></div><div>while you are right and availability of 264 imp=
lementation or hardware acceleration has improved, it has never been report=
ed as a problem in the previous pool by anyone. The 264 royalties, and the =
VP8 IP risks were, AFAIR, the main reasons used by individuals to justify t=
heir positions. Today, nothing has changed with respect to those two items =
(even though providing open264 royalties and absorbing the license cost for=
 some platforms might have been a set in the right direction). This is why =
I think the conditions are not met for a consensus to be reached.</div><div=
><br></div><div>Alex.</div><div><br></div></div><div><div><div class=3D"gma=
il_extra"><br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 11:43 PM, =
Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"javascript:_e(%7B%7D,&#39;cv=
ml&#39;,&#39;bernard.aboba@gmail.com&#39;);" target=3D"_blank">bernard.abob=
a@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div di=
r=3D"ltr"><span>&quot;And its one of the issues holding up wider adoption o=
f the technology&quot;<div><br></div></span><div>[BA] Specifying an MTI enc=
oder/decoder is not sufficient for interoperability.=C2=A0 It is also neces=
sary to specify the transport in enough detail to allow independent impleme=
ntations that interoperate well enough to be usable in a wide variety of sc=
enarios, including wireless networks where loss is commonly experienced. =
=C2=A0</div><div><br></div><div>We made the mistake of having an MTI discus=
sion previously with not enough details on that subject, particularly with =
respect to H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy=
 at best. =C2=A0</div><div><br></div><div>So if we are to have the discussi=
on again, it should occur in the context of detailed specifications and int=
eroperability reports.</div><div><br></div><div>=C2=A0</div><div><br></div>=
<div>=C2=A0</div></div><div><div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <span =
dir=3D"ltr">&lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;jdrosen=
@jdrosen.net&#39;);" target=3D"_blank">jdrosen@jdrosen.net</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">I&#39;m in favor o=
f taking another run at this.<div><br></div><div>The working group has repe=
atedly said that an MTI codec is something we need to produce. And its one =
of the issues holding up wider adoption of the technology (not the only one=
 for sure).</div><div><br></div><div>And things have changed since the last=
 meeting, a year ago now (November in Vancouver). Cisco&#39;s open264 plugi=
n shipped and now just recently is integrated into Firefox. iOS8 shipped wi=
th APIs for H264. There are other things worth noting. Will this change the=
 minds of everyone? Surely not. Will it sway enough for us to achieve rough=
 consensus? Maybe. IMHO - worth finding out.</div></div><div class=3D"gmail=
_extra"><div><div><br><div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:=
08 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"javascript:_e(=
%7B%7D,&#39;cvml&#39;,&#39;agouaillard@gmail.com&#39;);" target=3D"_blank">=
agouaillard@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div dir=3D"ltr">+1 to not having MTI codec discussion unless some progr=
ess has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing ha=
rald&#39;s =C2=A0feeling that there was no change on the members position.=
=C2=A0</div><div class=3D"gmail_extra"><div><div><br><div class=3D"gmail_qu=
ote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&=
lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;harald@alvestrand.no=
&#39;);" target=3D"_blank">harald@alvestrand.no</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"><span>On 10/17/2014 12:02 AM, Bernard Aboba wr=
ote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"javascript:_e(%7=
B%7D,&#39;cvml&#39;,&#39;martin.thomson@gmail.com&#39;);" target=3D"_blank"=
>martin.thomson@gmail.com</a>&gt; wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"javascript:_e(%7B%=
7D,&#39;cvml&#39;,&#39;matthew@matthew.at&#39;);" target=3D"_blank">matthew=
@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"javascript:_e(=
%7B%7D,&#39;cvml&#39;,&#39;jdrosen@jdrosen.net&#39;);" target=3D"_blank">jd=
rosen@jdrosen.net</a><br><a href=3D"http://www.jdrosen.net" target=3D"_blan=
k">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"javascript:_e(%7B%=
7D,&#39;cvml&#39;,&#39;jdrosen@jdrosen.net&#39;);" target=3D"_blank">jdrose=
n@jdrosen.net</a><br><a href=3D"http://www.jdrosen.net" target=3D"_blank">h=
ttp://www.jdrosen.net</a></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br></div>
</blockquote>

--047d7bea3de65e33ab0505d0bc69--


From nobody Sun Oct 19 19:18:53 2014
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Subject: Re: [rtcweb] Plan for MTI video codec?
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On Mon, Oct 20, 2014 at 01:20:04AM +0000, Cavigioli, Chris wrote:
> This is a complex topic with many network factors in play.  LTE
> operators in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS
> â€“ AMR transcoding in the network imposes an additional 211.5 ms on top
> of that, just for audio.  Optionally using AMR in WebRTC, those 211.5
> ms can be eliminated, delivering a superior end-user experience.

That seems like LTE's problem, not ours.

If you start with a codec like Opus where the maximum native frame
size is 60ms and manage to engineer a solution that blows that out
to near 400ms, we'd certainly have to applaud your, uh, innovation.


> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a
> good WebRTC â€“ LTE interop experience, emerging WebRTC systems must
> accommodate MTI codecs already deployed in the LTE domain and in
> mobile operating systems, namely AMR/AMR-WB and H.264.

You conclude that since LTE sucks, WebRTC ought to suck too?  That
seems like a strange direction to move in.  150ms is already ample
delay on its own to create a horrible user experience.  The aim of
WebRTC was to create a standard solution that doesn't suck.  If
LTE thinks it won't be able to compete with that, but considers
interoperating with it to be important, maybe it's time LTE thought
more about deploying a codec that's actually up to the job than
about protecting the royalty revenues of its voting group ...

If 150ms is your idea of a "superior end-user experience", then
maybe people who want a satisfactory user experience will simply
avoid using LTE at all, and then this won't be a problem for them,
or us, at all.

  Just sayin'



From nobody Sun Oct 19 21:24:41 2014
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From: Ca By <cb.list6@gmail.com>
To: "Cavigioli, Chris" <chris.cavigioli@intel.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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On Sunday, October 19, 2014, Cavigioli, Chris <chris.cavigioli@intel.com>
wrote:

>  Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
>
Chris,

My expectation is that something acceptable will be negotiated. And, these
LTE smartphones can and do support a variety of codecs, as you noted.

AMR and h264 are tied to walled garden mobile telco IMS mumbo jumbo, which
has little to do with webrtc, and webrtc has a lot to do with an open
internet. My guess is the IMS will change to interop with webrtc in an
effort to stay relevant.

CB

> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org
> <javascript:_e(%7B%7D,'cvml','rtcweb-bounces@ietf.org');>] *On Behalf Of =
*Bernard
> Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no
> <javascript:_e(%7B%7D,'cvml','harald@alvestrand.no');>> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net
> <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com
> <javascript:_e(%7B%7D,'cvml','agouaillard@gmail.com');>> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no
> <javascript:_e(%7B%7D,'cvml','harald@alvestrand.no');>> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com
> <javascript:_e(%7B%7D,'cvml','martin.thomson@gmail.com');>> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at
> <javascript:_e(%7B%7D,'cvml','matthew@matthew.at');>> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net <javascript:_e(%7B%7D,'cvml','jdrosen@jdrosen.net');>
> http://www.jdrosen.net
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> _______________________________________________
>
> rtcweb mailing list
>
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:_e(%7B%7D,'cvml','rtcweb@ietf.org');>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>

--047d7bfceb4c70846a0505d317f2
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<br><br>On Sunday, October 19, 2014, Cavigioli, Chris &lt;<a href=3D"mailto=
:chris.cavigioli@intel.com">chris.cavigioli@intel.com</a>&gt; wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-seri=
f&quot;;color:#1f497d"><u></u><u></u></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"><u></u>=C2=A0<u></u></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">We didn=E2=80=99t have quan=
titative data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=
=93 LTE interop is important.=C2=A0 A group of us in GSMA have been looking=
 at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">WebRTC =E2=80=93 LTE transc=
oding is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processin=
g Requirements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a significant degradation
 imposed on Web =E2=80=93 LTE links if only MTI codecs are used.=C2=A0 The =
only systems to avoid this are those which implement OPTIONAL audio codecs =
to be transcoder-free with LTE.=C2=A0 The same can apply for video codecs.<=
u></u><u></u></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"><u></u>=C2=A0<u></u></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">The GSMA whitepaper refers =
to:<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Quantitative voice-ove=
r-LTE (VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0=
 To paraphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">OPUS =E2=80=93 AMR tra=
nscoding adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub>=
 + T<sub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>3.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">To put these numbers i=
n perspective, it is in general desirable to minimize UE delays to ensure l=
ow enough end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">This is a complex topic wit=
h many network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays =
(T<sub>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS =E2=80=93
 AMR transcoding in the network imposes an additional 211.5 ms on top of th=
at, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 211.5 ms ca=
n be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">CONCLUSION: =C2=A0regardles=
s of MTI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE in=
terop experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">POSITION:=C2=A0 To be clear=
, several Intel SoCs launched at MWC this year for smartphones and tablets =
support both VP8 and H.264 hardware encode and decode =E2=80=93 so Intel
 is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has high-pe=
rformance solutions for transcoding.=C2=A0 However, we wish to inform the i=
ndustry, with quantitative data, about impact to end-user experience of man=
dating such transcoding so IETF can make a
 data-driven decision on video codecs as well as reflect on the impact of a=
lready having mandated transcoding on the audio side.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p><p class=3D"MsoNormal"></p></div></div></blockquote><div><br></div><div=
>Chris,</div><div><br></div><div>My expectation is that something acceptabl=
e will be negotiated. And, these LTE smartphones can and do support a varie=
ty of codecs, as you noted.=C2=A0</div><div><br></div><div>AMR and h264 are=
 tied to walled garden mobile telco IMS mumbo jumbo, which has little to do=
 with webrtc, and webrtc has a lot to do with an open internet.=C2=A0My gue=
ss is the IMS will change to interop with webrtc in an effort to stay relev=
ant.=C2=A0</div><div><br></div><div>CB=C2=A0</div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"M=
soNormal"></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Lucida Console&quot=
;;color:#1f497d">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">Intel Corp,
</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot=
;sans-serif&quot;;color:#0070c0">System Architecture and Planning</span><sp=
an style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans-seri=
f&quot;;color:#1f497d">, Video/Multimedia, Mobile and Communications Group =
(MCG)</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,=
&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">+1 (415) 254-4545 mobile</=
span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;s=
ans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">+1 (408) 653-8348 desk<u><=
/u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">+1 (408) 884-2400 fax</spa=
n><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans=
-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:gray">This e-mail may contain confi=
dential and privileged material for the sole use of the intended recipient(=
s).=C2=A0 Any review or distribution by others is strictly prohibited.
 If you are not the intended recipient, please contact the sender and delet=
e all copies.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
mailto:<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb-bounces@i=
etf.org&#39;);" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.=
org&#39;);" target=3D"_blank">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"font-size:10.0pt;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">So it can&#39;t be all bad.....</span>&qu=
ot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;harald@alvestrand.no=
&#39;);" target=3D"_blank">harald@alvestrand.no</a>&gt; wrote:<u></u><u></u=
></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u></u>
<u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;jdrosen@jdrosen.net=
&#39;);" target=3D"_blank">jdrosen@jdrosen.net</a>&gt; wrote:<u></u><u></u>=
</p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;agouaillard@gmail=
.com&#39;);" target=3D"_blank">agouaillard@gmail.com</a>&gt; wrote:<u></u><=
u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;harald@alvestrand.no=
&#39;);" target=3D"_blank">harald@alvestrand.no</a>&gt; wrote:<u></u><u></u=
></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"color:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On Oct 16, 2014, at 2=
:28 PM, Martin Thomson &lt;<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&=
#39;martin.thomson@gmail.com&#39;);" target=3D"_blank">martin.thomson@gmail=
.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;matthew@matthew.at&#39;);" ta=
rget=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA <u></u>
<u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-left:0in;line-height:14.4pt;vertical=
-align:baseline">
<u></u><span style=3D"font-size:10.0pt;font-family:Symbol;color:#333333"><s=
pan>=C2=B7<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:8.5pt;font-family:inhe=
rit;color:#333333"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;jdrosen@jdrosen.net&#39=
;);" target=3D"_blank">jdrosen@jdrosen.net</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#3=
9;);" target=3D"_blank">rtcweb@ietf.org</a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
</div>
</div>
<pre><span style=3D"color:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"color:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;rtcweb@ietf.org&#39;);"=
 target=3D"_blank">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>

</blockquote>

--047d7bfceb4c70846a0505d317f2--


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Date: Sun, 19 Oct 2014 22:52:14 -0700
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Subject: Re: [rtcweb] Plan for MTI video codec?
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--047d7b3a911c391d580505d4529a
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Looks like someone wants to have not just one, but two MTI codec
discussions...

On Sun, Oct 19, 2014 at 6:20 PM, Cavigioli, Chris <chris.cavigioli@intel.co=
m
> wrote:

>  Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Bernard
> Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> 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
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> 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
>
>

--047d7b3a911c391d580505d4529a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Looks like someone wants to have not just one, but two MTI=
 codec discussions...</div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Sun, Oct 19, 2014 at 6:20 PM, Cavigioli, Chris <span dir=3D"lt=
r">&lt;<a href=3D"mailto:chris.cavigioli@intel.com" target=3D"_blank">chris=
.cavigioli@intel.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-seri=
f&quot;;color:#1f497d"><u></u><u></u></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"><u></u>=C2=A0<u></u></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">We didn=E2=80=99t have quan=
titative data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=
=93 LTE interop is important.=C2=A0 A group of us in GSMA have been looking=
 at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">WebRTC =E2=80=93 LTE transc=
oding is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processin=
g Requirements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a significant degradation
 imposed on Web =E2=80=93 LTE links if only MTI codecs are used.=C2=A0 The =
only systems to avoid this are those which implement OPTIONAL audio codecs =
to be transcoder-free with LTE.=C2=A0 The same can apply for video codecs.<=
u></u><u></u></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"><u></u>=C2=A0<u></u></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">The GSMA whitepaper refers =
to:<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Quantitative voice-ove=
r-LTE (VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0=
 To paraphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">OPUS =E2=80=93 AMR tra=
nscoding adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub>=
 + T<sub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>3.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">To put these numbers i=
n perspective, it is in general desirable to minimize UE delays to ensure l=
ow enough end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">This is a complex topic wit=
h many network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays =
(T<sub>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS =E2=80=93
 AMR transcoding in the network imposes an additional 211.5 ms on top of th=
at, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 211.5 ms ca=
n be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">CONCLUSION: =C2=A0regardles=
s of MTI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE in=
terop experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">POSITION:=C2=A0 To be clear=
, several Intel SoCs launched at MWC this year for smartphones and tablets =
support both VP8 and H.264 hardware encode and decode =E2=80=93 so Intel
 is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has high-pe=
rformance solutions for transcoding.=C2=A0 However, we wish to inform the i=
ndustry, with quantitative data, about impact to end-user experience of man=
dating such transcoding so IETF can make a
 data-driven decision on video codecs as well as reflect on the impact of a=
lready having mandated transcoding on the audio side.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Lucida Console&quot=
;;color:#1f497d">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">Intel Corp,
</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot=
;sans-serif&quot;;color:#0070c0">System Architecture and Planning</span><sp=
an style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans-seri=
f&quot;;color:#1f497d">, Video/Multimedia, Mobile and Communications Group =
(MCG)</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,=
&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
15%29%20254-4545" value=3D"+14152544545" target=3D"_blank">+1 (415) 254-454=
5</a> mobile</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana=
&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20653-8348" value=3D"+14086538348" target=3D"_blank">+1 (408) 653-834=
8</a> desk<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20884-2400" value=3D"+14088842400" target=3D"_blank">+1 (408) 884-240=
0</a> fax</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&qu=
ot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:gray">This e-mail may contain confi=
dential and privileged material for the sole use of the intended recipient(=
s).=C2=A0 Any review or distribution by others is strictly prohibited.
 If you are not the intended recipient, please contact the sender and delet=
e all copies.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<span class=3D""><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</span></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p><div><div class=
=3D"h5">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"font-size:10.0pt;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">So it can&#39;t be all bad.....</span>&qu=
ot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div></div></div><div><div class=3D"h5">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u></u>
<u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"color:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On Oct 16, 2014, at 2=
:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA <u></u>
<u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-left:0in;line-height:14.4pt;vertical=
-align:baseline">
<u></u><span style=3D"font-size:10.0pt;font-family:Symbol;color:#333333"><s=
pan>=C2=B7<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:8.5pt;font-family:inhe=
rit;color:#333333"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
</div>
</div>
<pre><span style=3D"color:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"color:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>

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

--047d7b3a911c391d580505d4529a--


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From: Victor Pascual <victor.pascual.avila@gmail.com>
Date: Mon, 20 Oct 2014 08:00:21 +0200
To: Ca By <cb.list6@gmail.com>
Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/Awknzo7sxd1al1Xq5t5HVA_9cP4
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Subject: Re: [rtcweb] Plan for MTI video codec?
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And that's what the 3GPP is doing for release 12: http://webrtchacks.com/ims=
-approach-webrtc/

> On Oct 20, 2014, at 6:24 AM, Ca By <cb.list6@gmail.com> wrote:
>=20
> My guess is the IMS will change to interop with webrtc in an effort to sta=
y relevant.


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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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From: Barry Dingle <btdingle@gmail.com>
Date: Mon, 20 Oct 2014 19:52:38 +1100
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To: "Cavigioli, Chris" <chris.cavigioli@intel.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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Chris,

So you are saying that we should NOT have 'unqualified' MTI Audio and Video
codecs.

Rather, we should have 2 (or more) Network Scenarios and have Audio+Video
MTI's for Each Network Scenario.

Browsers that wish to support multiple Network Scenarios will support the
MTI's for those Scenarios. Special Browsers - or a sub-set of a Browser -
MAY support only specific Network Scenarios and their MTIs.

This seems to make a lot of sense.

Cheers,
Barry Dingle
Fixed - +61(0)3-9725-3937    Mob - +61(0)41-911-7578
Fellow of University of Melbourne, Australia

On Mon, Oct 20, 2014 at 6:09 PM, Cavigioli, Chris <chris.cavigioli@intel.co=
m
> wrote:

>  No, not 2 discussions =E2=80=A6 but 2 MTI codecs.
>
> Given:
>
> -       Industry is polarized c. 40% VP8 vs. c. 40% H.264
>
> -       Unlikely this will change
>
> -       Every single mobile device (100%) of the billions out there
> running a mobile OS (iOS, Android, Windows, BB OS, etc) will already
> support H.264, AMR-xx since those are MTI for 3GPP and for the OS
>
> Then:
>
> -       Why not support both:  VP8, Opus =3D web; H.264, AMR/AMR-WB =3D L=
TE
>
> -       Every single mobile device and OS already deals with H.264+AMR
> licensing for many years.
>
> -       Being royalty free, incrementally adding VP8 and Opus shouldn=E2=
=80=99t
> be a major additional impact
>
> -chris
>
>
>
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* Sunday, October 19, 2014 10:52 PM
> *To:* Cavigioli, Chris
> *Cc:* Harald Alvestrand; Bernard Aboba; rtcweb@ietf.org
>
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Looks like someone wants to have not just one, but two MTI codec
> discussions...
>
>
>
> On Sun, Oct 19, 2014 at 6:20 PM, Cavigioli, Chris <
> chris.cavigioli@intel.com> wrote:
>
> Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Bernard
> Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> 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
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> 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
>
>

--001a113369a66fddeb0505d6d745
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div>Chris, <br><br></div>So you are saying=
 that we should NOT have &#39;unqualified&#39; MTI Audio and Video codecs. =
<br><br></div>Rather, we should have 2 (or more) Network Scenarios and have=
 Audio+Video MTI&#39;s for Each Network Scenario. <br><br></div>Browsers th=
at wish to support multiple Network Scenarios will support the MTI&#39;s fo=
r those Scenarios. Special Browsers - or a sub-set of a Browser - MAY suppo=
rt only specific Network Scenarios and their MTIs. <br><br></div>This seems=
 to make a lot of sense. <br><div class=3D"gmail_extra"><br clear=3D"all"><=
div><div dir=3D"ltr">Cheers,<br>Barry Dingle<br>Fixed - <a href=3D"tel:%2B6=
1%280%293-9725-3937" value=3D"+61397253937" target=3D"_blank">+61(0)3-9725-=
3937</a>=C2=A0 =C2=A0 Mob - +61(0)41-911-7578=C2=A0=C2=A0 <br>Fellow of Uni=
versity of Melbourne, Australia <br></div></div>
<br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 6:09 PM, Cavigioli, =
Chris <span dir=3D"ltr">&lt;<a href=3D"mailto:chris.cavigioli@intel.com" ta=
rget=3D"_blank">chris.cavigioli@intel.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">





<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">No, not 2 discussions =E2=
=80=A6 but 2 MTI codecs.=C2=A0
<u></u><u></u></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">Given:<u></u><u></u></span>=
</p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Industry is polarized =
c. 40% VP8 vs. c. 40% H.264
<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Unlikely this will cha=
nge<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Every single mobile de=
vice (100%) of the billions out there running a mobile OS (iOS, Android, Wi=
ndows, BB OS, etc) will already support H.264, AMR-xx
 since those are MTI for 3GPP and for the OS<u></u><u></u></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">Then:<u></u><u></u></span><=
/p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Why not support both:=
=C2=A0 VP8, Opus =3D web; H.264, AMR/AMR-WB =3D LTE<u></u><u></u></span></p=
>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Every single mobile de=
vice and OS already deals with H.264+AMR licensing for many years.<u></u><u=
></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;T=
imes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Being royalty free, in=
crementally adding VP8 and Opus shouldn=E2=80=99t be a major additional imp=
act
<u></u><u></u></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">-chris<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Justin U=
berti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@google.com</a>]
<br>
<b>Sent:</b> Sunday, October 19, 2014 10:52 PM<br>
<b>To:</b> Cavigioli, Chris<br>
<b>Cc:</b> Harald Alvestrand; Bernard Aboba; <a href=3D"mailto:rtcweb@ietf.=
org" target=3D"_blank">rtcweb@ietf.org</a></span></p><div><div><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></div><=
/div><p></p><div><div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Looks like someone wants to have not just one, but t=
wo MTI codec discussions...<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 6:20 PM, Cavigioli, Chris &l=
t;<a href=3D"mailto:chris.cavigioli@intel.com" target=3D"_blank">chris.cavi=
gioli@intel.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">Harald said:=C2=A0
</span><u></u><u></u></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<u=
></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">We didn=E2=80=99t have quan=
titative data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=
=93 LTE interop is important.=C2=A0
 A group of us in GSMA have been looking at WebRTC interop with LTE (where =
3GPP / GSMA defined IMS-based voice/video/messaging), specifically looking =
at user experience impact of latency introduced by added in-network transco=
ding.=C2=A0 GSMA has approved the whitepaper
 and it is being prepared for public distribution.=C2=A0 </span><u></u><u><=
/u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">WebRTC =E2=80=93 LTE transc=
oding is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processin=
g Requirements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a
 significant degradation imposed on Web =E2=80=93 LTE links if only MTI cod=
ecs are used.=C2=A0 The only systems to avoid this are those which implemen=
t OPTIONAL audio codecs to be transcoder-free with LTE.=C2=A0 The same can =
apply for video codecs.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">The GSMA whitepaper refers =
to:</span><u></u><u></u></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans=
-serif&quot;;color:#1f497d">1.</span><span style=3D"font-size:7.0pt;color:#=
1f497d">=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;;color:#1f497d">Quantitative voice-over-LTE (VoLTE) delay l=
imits were agreed-to by 3GPP SA4 in August 2014.=C2=A0 To paraphrase, perfo=
rmance objectives for maximum VoLTE device delays in error
 and jitter free conditions and AMR speech codec operation, the sum of the =
UE delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub=
>) should be =E2=89=A4 150ms. If this performance objective cannot be met, =
the sum of the UE delays in sending and
 receiving directions (T<sub>S</sub> + T<sub>R</sub>) shall in any case be =
=E2=89=A4 190ms.</span><u></u><u></u></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans=
-serif&quot;;color:#1f497d">2.</span><span style=3D"font-size:7.0pt;color:#=
1f497d">=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;;color:#1f497d">OPUS =E2=80=93 AMR transcoding adds an addi=
tional 211.5 ms implementation-agnostic (T<sub>S</sub> + T<sub>R</sub>) del=
ay, including 40 ms for jitter buffer and 40 ms for discontinuous
 reception (DRX).</span><u></u><u></u></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans=
-serif&quot;;color:#1f497d">3.</span><span style=3D"font-size:7.0pt;color:#=
1f497d">=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;;color:#1f497d">To put these numbers in perspective, it is =
in general desirable to minimize UE delays to ensure low enough end-to-end =
delays and hence a good conversational experience.=C2=A0 Increasing
 delay causes people to double-talk making conversations increasingly frust=
rating.=C2=A0 Guidance to stay below 400 ms maximum one-way delay is found =
in ITU-T Recommendation G.114 and the computational E-model is in ITU-T Rec=
ommendation G.107.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">This is a complex topic wit=
h many network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays =
(T<sub>S</sub>
 + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS =E2=80=93 AMR transcodin=
g in the network imposes an additional 211.5 ms on top of that, just for au=
dio.=C2=A0 Optionally using AMR in WebRTC, those 211.5 ms can be eliminated=
, delivering a superior end-user experience.=C2=A0
</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">CONCLUSION: =C2=A0regardles=
s of MTI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE in=
terop experience,
 emerging WebRTC systems must accommodate MTI codecs already deployed in th=
e LTE domain and in mobile operating systems, namely AMR/AMR-WB and H.264.<=
/span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">POSITION:=C2=A0 To be clear=
, several Intel SoCs launched at MWC this year for smartphones and tablets =
support
 both VP8 and H.264 hardware encode and decode =E2=80=93 so Intel is codec-=
neutral in this MTI debate.=C2=A0 Additionally, Intel has high-performance =
solutions for transcoding.=C2=A0 However, we wish to inform the industry, w=
ith quantitative data, about impact to end-user experience
 of mandating such transcoding so IETF can make a data-driven decision on v=
ideo codecs as well as reflect on the impact of already having mandated tra=
nscoding on the audio side.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Lucida Console&quot=
;;color:#1f497d">Chris Cavigioli</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">Intel Corp,
</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot=
;sans-serif&quot;;color:#0070c0">System Architecture and Planning</span><sp=
an style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans-seri=
f&quot;;color:#1f497d">, Video/Multimedia, Mobile and Communications Group =
(MCG)</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
15%29%20254-4545" target=3D"_blank">+1 (415) 254-4545</a> mobile</span><u><=
/u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20653-8348" target=3D"_blank">+1 (408) 653-8348</a> desk</span><u></u=
><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20884-2400" target=3D"_blank">+1 (408) 884-2400</a> fax</span><u></u>=
<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:gray">This e-mail may contain confi=
dential and privileged material for the sole use of the intended recipient(=
s).=C2=A0
 Any review or distribution by others is strictly prohibited. If you are no=
t the intended recipient, please contact the sender and delete all copies.<=
/span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"font-size:10.0pt;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">So it can&#39;t be all bad.....</span>&qu=
ot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will
 exhibit interoperability problems, regardless of the codec chosen. =C2=A0 =
The real test for RTCWEB will be to move beyond &quot;interoperability of i=
mplementations sharing source code&quot; =C2=A0to &quot;interoperability of=
 independent implementations, based on open standards&quot;. =C2=A0=C2=A0<u=
></u><u></u></p>
</div>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations
 that interoperate well enough to be usable in a wide variety of scenarios,=
 including wireless networks where loss is commonly experienced.=C2=A0
<u></u><u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4
 remain sketchy at best. =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this.
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one
 for sure).<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped
 with APIs for H264. There are other things worth noting. Will this change =
the minds of everyone? Surely not. Will it sway enough for us to achieve ro=
ugh consensus? Maybe. IMHO - worth finding out.<u></u><u></u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members
 position.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless
 of the codec.<u></u><u></u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"color:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On Oct 16, 2014, at 2=
:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">--
</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA
</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------</span><u></u=
><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View</span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">President - CoSMo Soft=
ware, Cambridge, MA</span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------</span><u></u=
><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
/span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"line-height:14.4pt;vertical-align:baseline"=
>
<span style=3D"font-size:10.0pt;font-family:Symbol;color:#333333">=C2=B7</s=
pan><span style=3D"font-size:7.0pt;color:#333333">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:8.5pt;font-family:inherit;color:#333333">=
=C2=A0</span><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">--
<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">=C2=A0<u></u><u></u><=
/p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">=C2=A0<u></u><u></u><=
/p>
</div>
</div>
<pre><span style=3D"color:#888888">-- </span><u></u><u></u></pre>
<pre><span style=3D"color:#888888">Surveillance is pervasive. Go Dark.</spa=
n><u></u><u></u></pre>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>

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

--001a113369a66fddeb0505d6d745--


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Victor Pascual <victor.pascual.avila@gmail.com>, Ca By <cb.list6@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
Thread-Index: AQHP669ol2Qsv7sEAECBFDAOBE1Cspw3bfYAgABSSQCAAA5FgIAAQIoAgAAzigCAABrFgIAAWhNA
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Subject: Re: [rtcweb] Plan for MTI video codec?
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Given that codecs appear in the end device and the 3GPP activity is to prov=
ide a gateway, either the gateway provides transcoding or the two end devic=
es use SDP to agree on a mutually acceptable codec.=20

Our view is that in many of these cases, it is perfectly possible for a web=
RTC device and the 3GPP device to decide to use H.264 as the codec and will=
 do so, in order to avoid transcoding.=20

That will occur whatever is specified as an MTI video codec.

Keith

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
> Victor Pascual
> Sent: 20 October 2014 07:00
> To: Ca By
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Plan for MTI video codec?
>=20
> And that's what the 3GPP is doing for release 12:=20
> http://webrtchacks.com/ims-approach-webrtc/
>=20
> > On Oct 20, 2014, at 6:24 AM, Ca By <cb.list6@gmail.com> wrote:
> >=20
> > My guess is the IMS will change to interop with webrtc in=20
> an effort to stay relevant.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =


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Date: Mon, 20 Oct 2014 05:02:46 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: Barry Dingle <btdingle@gmail.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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On 20 October 2014 01:52, Barry Dingle <btdingle@gmail.com> wrote:
> Rather, we should have 2 (or more) Network Scenarios and have Audio+Video
> MTI's for Each Network Scenario.


Why should your network attachment conditions determine who you can talk to?


From nobody Mon Oct 20 06:52:39 2014
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From: Andrew Allen <aallen@blackberry.com>
To: "chris.cavigioli@intel.com" <chris.cavigioli@intel.com>, "harald@alvestrand.no" <harald@alvestrand.no>, "bernard.aboba@gmail.com" <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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Date: Mon, 20 Oct 2014 13:52:10 +0000
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Subject: Re: [rtcweb] Plan for MTI video codec?
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Andrew Allen <aallen@blackberry.com>, "chris.cavigioli@intel.com" <chris.cavigioli@intel.com>, "harald@alvestrand.no" <harald@alvestrand.no>, "bernard.aboba@gmail.com" <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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Date: Mon, 20 Oct 2014 14:34:58 +0000
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Subject: Re: [rtcweb] Plan for MTI video codec?
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The scenario where you might want to transcode the audio codec is where you=
 have wideband audio in both the OPUS and the AMR, and you believe by trans=
coding you can achieve a wideband quality, rather than falling back to G.71=
1. And for interworking between 3GPP devices that are not using webRTC, the=
 audio will still be AMR or AMR-WB or EVC, not OPUS.

But audio transcoding is not particularly a concern for me. It is trying wh=
ere possible to avoid video transcoding.

Keith

________________________________
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Andrew Allen
Sent: 20 October 2014 14:52
To: chris.cavigioli@intel.com; harald@alvestrand.no; bernard.aboba@gmail.co=
m
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Plan for MTI video codec?


Chris

What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?

All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.

My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.

For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices interoperating with circuit switched only=
 mobile devices which will be forced to interoperate via the PSTN.

As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with non-mobile RTCweb devices.

The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.

Andrew

From: Cavigioli, Chris [mailto:chris.cavigioli@intel.com]
Sent: Sunday, October 19, 2014 09:20 PM Eastern Standard Time
To: Harald Alvestrand <harald@alvestrand.no>; Bernard Aboba <bernard.aboba@=
gmail.com>
Cc: rtcweb@ietf.org <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?

Harald said:
"We made the mistake of having an MTI discussion previously with not enough=
 details on that subject..."

We didn't have quantitative data earlier for a data-driven decision.  Futur=
e Web - LTE interop is important.  A group of us in GSMA have been looking =
at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based voice/video=
/messaging), specifically looking at user experience impact of latency intr=
oduced by added in-network transcoding.  GSMA has approved the whitepaper a=
nd it is being prepared for public distribution.

WebRTC - LTE transcoding is now MANDATORY because of "WebRTC Audio Codec an=
d Processing Requirements". http://tools.ietf.org/html/draft-ietf-rtcweb-au=
dio-05, where MTI audio codecs in WebRTC and in LTE are dissimilar.  Thus, =
REGARDLESS of the video codecs, there is already a significant degradation =
imposed on Web - LTE links if only MTI codecs are used.  The only systems t=
o avoid this are those which implement OPTIONAL audio codecs to be transcod=
er-free with LTE.  The same can apply for video codecs.

The GSMA whitepaper refers to:

1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by 3=
GPP SA4 in August 2014.  To paraphrase, performance objectives for maximum =
VoLTE device delays in error and jitter free conditions and AMR speech code=
c operation, the sum of the UE delays in sending and receiving directions (=
TS + TR) should be =98 150ms. If this performance objective cannot be met, =
the sum of the UE delays in sending and receiving directions (TS + TR) shal=
l in any case be =98 190ms.

2.     OPUS - AMR transcoding adds an additional 211.5 ms implementation-ag=
nostic (TS + TR) delay, including 40 ms for jitter buffer and 40 ms for dis=
continuous reception (DRX).

3.     To put these numbers in perspective, it is in general desirable to m=
inimize UE delays to ensure low enough end-to-end delays and hence a good c=
onversational experience.  Increasing delay causes people to double-talk ma=
king conversations increasingly frustrating.  Guidance to stay below 400 ms=
 maximum one-way delay is found in ITU-T Recommendation G.114 and the compu=
tational E-model is in ITU-T Recommendation G.107.

This is a complex topic with many network factors in play.  LTE operators i=
n 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS - AMR transcodi=
ng in the network imposes an additional 211.5 ms on top of that, just for a=
udio.  Optionally using AMR in WebRTC, those 211.5 ms can be eliminated, de=
livering a superior end-user experience.

CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good W=
ebRTC - LTE interop experience, emerging WebRTC systems must accommodate MT=
I codecs already deployed in the LTE domain and in mobile operating systems=
, namely AMR/AMR-WB and H.264.

POSITION:  To be clear, several Intel SoCs launched at MWC this year for sm=
artphones and tablets support both VP8 and H.264 hardware encode and decode=
 - so Intel is codec-neutral in this MTI debate.  Additionally, Intel has h=
igh-performance solutions for transcoding.  However, we wish to inform the =
industry, with quantitative data, about impact to end-user experience of ma=
ndating such transcoding so IETF can make a data-driven decision on video c=
odecs as well as reflect on the impact of already having mandated transcodi=
ng on the audio side.

Chris Cavigioli
Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile and =
Communications Group (MCG)
+1 (415) 254-4545 mobile
+1 (408) 653-8348 desk
+1 (408) 884-2400 fax
This e-mail may contain confidential and privileged material for the sole u=
se of the intended recipient(s).  Any review or distribution by others is s=
trictly prohibited. If you are not the intended recipient, please contact t=
he sender and delete all copies.

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Bernard Aboba
Sent: Sunday, October 19, 2014 2:29 PM
To: Harald Alvestrand
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Plan for MTI video codec?

Harald said:

"Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.
So it can't be all bad....."

[BA]  I agree that the lack of interoperability is largely "codec independe=
nt":   that is, implementations based on different mechanisms for congestio=
n control, repair, etc. will exhibit interoperability problems, regardless =
of the codec chosen.   The real test for RTCWEB will be to move beyond "int=
eroperability of implementations sharing source code"  to "interoperability=
 of independent implementations, based on open standards".

On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
On 10/19/2014 05:43 PM, Bernard Aboba wrote:
"And its one of the issues holding up wider adoption of the technology"

[BA] Specifying an MTI encoder/decoder is not sufficient for interoperabili=
ty.  It is also necessary to specify the transport in enough detail to allo=
w independent implementations that interoperate well enough to be usable in=
 a wide variety of scenarios, including wireless networks where loss is com=
monly experienced.

Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.

So it can't be all bad.....




We made the mistake of having an MTI discussion previously with not enough =
details on that subject, particularly with respect to H.264. draft-ietf-rtc=
web-video sections 4.2 - 4.4 remain sketchy at best.

So if we are to have the discussion again, it should occur in the context o=
f detailed specifications and interoperability reports.





On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net<ma=
ilto:jdrosen@jdrosen.net>> wrote:
I'm in favor of taking another run at this.

The working group has repeatedly said that an MTI codec is something we nee=
d to produce. And its one of the issues holding up wider adoption of the te=
chnology (not the only one for sure).

And things have changed since the last meeting, a year ago now (November in=
 Vancouver). Cisco's open264 plugin shipped and now just recently is integr=
ated into Firefox. iOS8 shipped with APIs for H264. There are other things =
worth noting. Will this change the minds of everyone? Surely not. Will it s=
way enough for us to achieve rough consensus? Maybe. IMHO - worth finding o=
ut.

On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <agouaillard@gmail.co=
m<mailto:agouaillard@gmail.com>> wrote:
+1 to not having MTI codec discussion unless some progress has been made on=
 VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there wa=
s no change on the members position.

On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
On 10/17/2014 12:02 AM, Bernard Aboba wrote:
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.

The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.

I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of "well" - and I would pref=
er that ecosystem to be one where it's possible to field the box without ma=
king prior arrangements with anyone about licenses.

This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.

We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)

I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.

At the moment, I don't see signs that any of the poll respondents have said=
 "I have changed my mind".

Harald




On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com<mailt=
o:martin.thomson@gmail.com>> wrote:
On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at<mailto:matthe=
w@matthew.at>> wrote:
And that's because something substantive has changed, or simply because
wasting the WG time on this again is more entertaining than actually
finishing a specification that can be independently implemented by all
browser vendors? (A specification that we are nowhere near having, as far a=
s
I can tell)
Personally, I've found the reprieve from this fight refreshing.  And
it would appear that we've made some real progress as a result.  I'd
suggest that if we don't have new information, we continue to spend
our time productively.  If we can't find topics to occupy our meeting
agenda time, then maybe we can free an agenda slot.

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

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



--
Alex. Gouaillard, PhD, PhD, MBA
---------------------------------------------------------------------------=
---------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
---------------------------------------------------------------------------=
---------
sg.linkedin.com/agouaillard<http://sg.linkedin.com/agouaillard>
=9E

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



--
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net<mailto:jdrosen@jdrosen.net>
http://www.jdrosen.net

_______________________________________________
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


--

Surveillance is pervasive. Go Dark.

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


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<div dir=3D"ltr" align=3D"left"><span class=3D"152132914-20102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">The scenario where you might want=
 to transcode the audio codec is where you have wideband audio in both the =
OPUS and the AMR, and you believe by transcoding
 you can achieve a wideband quality, rather than falling back to G.711. And=
 for interworking between 3GPP devices that are not using webRTC, the audio=
 will still be AMR or AMR-WB or EVC, not OPUS.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"152132914-20102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"152132914-20102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">But audio transcoding is not part=
icularly a concern for me. It is trying where possible to avoid video trans=
coding.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"152132914-20102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"152132914-20102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDE=
R-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<div class=3D"OutlookMessageHeader" lang=3D"en-us" dir=3D"ltr" align=3D"lef=
t">
<hr tabindex=3D"-1">
<font face=3D"Tahoma" size=3D"2"><b>From:</b> rtcweb [mailto:rtcweb-bounces=
@ietf.org]
<b>On Behalf Of </b>Andrew Allen<br>
<b>Sent:</b> 20 October 2014 14:52<br>
<b>To:</b> chris.cavigioli@intel.com; harald@alvestrand.no; bernard.aboba@g=
mail.com<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</font><br>
</div>
<div></div>
<font style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','san=
s-serif'"><br>
Chris<br>
<br>
What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?
<br>
<br>
All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.
<br>
<br>
My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.<br>
<br>
For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices
 interoperating with circuit switched only mobile devices which will be for=
ced to interoperate via the PSTN.<br>
<br>
As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with
 non-mobile RTCweb devices. <br>
<br>
The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.<br>
<br>
Andrew</font><br>
&nbsp;<br>
<div style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b=
5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: mediu=
m none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<font style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"><b>From=
</b>: Cavigioli, Chris [mailto:chris.cavigioli@intel.com]
<br>
<b>Sent</b>: Sunday, October 19, 2014 09:20 PM Eastern Standard Time<br>
<b>To</b>: Harald Alvestrand &lt;harald@alvestrand.no&gt;; Bernard Aboba &l=
t;bernard.aboba@gmail.com&gt;
<br>
<b>Cc</b>: rtcweb@ietf.org &lt;rtcweb@ietf.org&gt; <br>
<b>Subject</b>: Re: [rtcweb] Plan for MTI video codec? <br>
</font>&nbsp;<br>
</div>
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">Harald said:&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal">&#8220;We made the mistake of having an MTI discussi=
on previously with not enough details on that subject&#8230;&#8221;<span st=
yle=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Arial','sans-serif'">=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">We didn&#8217;t have quantitative data earli=
er for a data-driven decision.&nbsp; Future Web &#8211; LTE interop is impo=
rtant.&nbsp; A group of us in GSMA have been looking at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.&nbsp; GSMA has approved the whitepaper and i=
t is being prepared for public distribution.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">WebRTC &#8211; LTE transcoding is now MANDAT=
ORY because of &#8220;WebRTC Audio Codec and Processing Requirements&#8221;=
.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05">http://to=
ols.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where MTI audio codecs in=
 WebRTC and in LTE are dissimilar.&nbsp; Thus, REGARDLESS of the video code=
cs, there is already a significant degradation
 imposed on Web &#8211; LTE links if only MTI codecs are used.&nbsp; The on=
ly systems to avoid this are those which implement OPTIONAL audio codecs to=
 be transcoder-free with LTE.&nbsp; The same can apply for video codecs.<o:=
p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">The GSMA whitepaper refers to:<o:p></o:p></s=
pan></p>
<p class=3D"MsoListParagraph" style=3D"TEXT-INDENT: -0.25in; mso-list: l1 l=
evel1 lfo2">
<![if !supportLists]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-F=
AMILY: 'Arial','sans-serif'"><span style=3D"mso-list: Ignore">1.<span style=
=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f49=
7d; FONT-FAMILY: 'Arial','sans-serif'">Quantitative voice-over-LTE (VoLTE) =
delay limits were agreed-to by 3GPP SA4 in August 2014.&nbsp; To paraphrase=
, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> &#43; T<sub>R</sub>) should be =98 150ms. If this performance obje=
ctive cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> &#43; T<sub>R</s=
ub>) shall in any case be =98 190ms.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"TEXT-INDENT: -0.25in; mso-list: l1 l=
evel1 lfo2">
<![if !supportLists]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-F=
AMILY: 'Arial','sans-serif'"><span style=3D"mso-list: Ignore">2.<span style=
=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f49=
7d; FONT-FAMILY: 'Arial','sans-serif'">OPUS &#8211; AMR transcoding adds an=
 additional 211.5 ms implementation-agnostic (T<sub>S</sub> &#43; T<sub>R</=
sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"TEXT-INDENT: -0.25in; mso-list: l1 l=
evel1 lfo2">
<![if !supportLists]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-F=
AMILY: 'Arial','sans-serif'"><span style=3D"mso-list: Ignore">3.<span style=
=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"FONT-SIZE: 10pt; COLOR: #1f49=
7d; FONT-FAMILY: 'Arial','sans-serif'">To put these numbers in perspective,=
 it is in general desirable to minimize UE delays to ensure low enough end-=
to-end delays and hence a good conversational
 experience.&nbsp; Increasing delay causes people to double-talk making con=
versations increasingly frustrating.&nbsp; Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">This is a complex topic with many network fa=
ctors in play.&nbsp; LTE operators in 3GPP seek UE delays (T<sub>S</sub> &#=
43; T<sub>R</sub>) around 150 ms.&nbsp; Adding OPUS
 &#8211; AMR transcoding in the network imposes an additional 211.5 ms on t=
op of that, just for audio.&nbsp; Optionally using AMR in WebRTC, those 211=
.5 ms can be eliminated, delivering a superior end-user experience.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">CONCLUSION: &nbsp;regardless of MTI codec ch=
oices in WebRTC, to provide a good WebRTC &#8211; LTE interop experience, e=
merging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">POSITION:&nbsp; To be clear, several Intel S=
oCs launched at MWC this year for smartphones and tablets support both VP8 =
and H.264 hardware encode and decode &#8211; so
 Intel is codec-neutral in this MTI debate.&nbsp; Additionally, Intel has h=
igh-performance solutions for transcoding.&nbsp; However, we wish to inform=
 the industry, with quantitative data, about impact to end-user experience =
of mandating such transcoding so IETF can
 make a data-driven decision on video codecs as well as reflect on the impa=
ct of already having mandated transcoding on the audio side.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"COLOR: #1f497d; FONT-FAMILY: 'Lucida =
Console'">Chris Cavigioli<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'">Intel Corp,
</span><span style=3D"FONT-SIZE: 8pt; COLOR: #0070c0; FONT-FAMILY: 'Verdana=
','sans-serif'">System Architecture and Planning</span><span style=3D"FONT-=
SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verdana','sans-serif'">, Video/Mul=
timedia, Mobile and Communications Group
 (MCG)</span><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'V=
erdana','sans-serif'"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'">&#43;1 (415) 254-4545 mobile</span><span st=
yle=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verdana','sans-serif'"=
><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'">&#43;1 (408) 653-8348 desk<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'">&#43;1 (408) 884-2400 fax</span><span style=
=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verdana','sans-serif'"><o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 7.5pt; COLOR: gray; FONT-F=
AMILY: 'Verdana','sans-serif'">This e-mail may contain confidential and pri=
vileged material for the sole use of the intended recipient(s).&nbsp; Any r=
eview or distribution by others is strictly
 prohibited. If you are not the intended recipient, please contact the send=
er and delete all copies.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tah=
oma','sans-serif'">From:</span></b><span style=3D"FONT-SIZE: 10pt; FONT-FAM=
ILY: 'Tahoma','sans-serif'"> rtcweb [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Harald said:&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: '=
Arial','sans-serif'">Bernard,</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><span style=3D"FONT-SI=
ZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial'=
,'sans-serif'">So it can't be all bad.....</span>&quot;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] &nbsp;I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: &nbsp; that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. &nbsp; The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; &nbsp;to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. &nbsp;&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<o:p></o=
:p></p>
</div>
<blockquote style=3D"MARGIN-TOP: 5pt; MARGIN-BOTTOM: 5pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.&nbsp; It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.&nbsp; <o:p>
</o:p></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can't be all bad.....<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 &nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">I'm in favor of taking another run at this. <o:p></o=
:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco's open264 plugin shipped and now =
just recently is integrated into Firefox. iOS8 shipped with APIs for H264. =
There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<o:p></o:p=
></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">&#43;1 to not having MTI codec discussion unless som=
e progress has been made on VP8 at MPEG.&nbsp;Any news on that? I'm sharing=
 harald's &nbsp;feeling that there was no change on the members position.&n=
bsp;<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<o:p></o=
:p></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<o:p></o:p><=
/p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it's possible to field the box =
without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.<br>
<br>
We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)<br>
<br>
I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.<br>
<br>
At the moment, I don't see signs that any of the poll respondents have said=
 &quot;I have changed my mind&quot;.<span style=3D"COLOR: #888888"><br>
<br>
Harald</span> <o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt">On Oct 16, 2014, at 2:=
28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targe=
t=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that's because something substantive has changed, or simply because<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<o:p></o:p></p>
<p class=3D"MsoNormal">Personally, I've found the reprieve from this fight =
refreshing.&nbsp; And<br>
it would appear that we've made some real progress as a result.&nbsp; I'd<b=
r>
suggest that if we don't have new information, we continue to spend<br>
our time productively.&nbsp; If we can't find topics to occupy our meeting<=
br>
agenda time, then maybe we can free an agenda slot.<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><o:p></o:p></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">-- <o:p></o:p></span>=
</p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">Alex. Gouaillard, PhD=
, PhD, MBA
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">---------------------=
---------------------------------------------------------------<o:p></o:p><=
/span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">CTO - Temasys Communi=
cations, S'pore / Mountain View<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">President - CoSMo Sof=
tware, Cambridge, MA<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">---------------------=
---------------------------------------------------------------<o:p></o:p><=
/span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888"><a href=3D"http://sg.=
linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a>=
<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-LEFT: 0in; VERTICAL-ALIGN: baseline;=
 TEXT-INDENT: -0.25in; LINE-HEIGHT: 14.4pt; mso-list: l0 level1 lfo1">
<![if !supportLists]><span style=3D"FONT-SIZE: 10pt; COLOR: #333333; FONT-F=
AMILY: Symbol"><span style=3D"mso-list: Ignore">=9E<span style=3D"FONT: 7pt=
 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"FONT-SIZE: 8.5pt; COLOR: #333=
333; FONT-FAMILY: inherit"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">-- <o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><o:p>&nbsp;</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" target=3D"_blank">rtcweb@ietf.org</=
a><o:p></o:p></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><o:p>&nbsp;</o:p></p>
</div>
</div>
<pre><span style=3D"COLOR: #888888">-- <o:p></o:p></span></pre>
<pre><span style=3D"COLOR: #888888">Surveillance is pervasive. Go Dark.<o:p=
></o:p></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</blockquote>
</body>
</html>

--_000_949EF20990823C4C85C18D59AA11AD8B2627F1FR712WXCHMBA11zeu_--


From nobody Mon Oct 20 11:55:42 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Peter Saint-Andre - &yet <peter@andyet.net>
Thread-Topic: [rtcweb] Agenda for the upcoming meeting
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So lets be clear here. Every time we checked, there has been consensus that=
 we need an MTI codec. So right now this work is sort of blocked on that an=
d we would like to get what people are calling webrtc 1.0 out before the en=
d of 2015.  I think that a pragmatic look at this would result there is at =
least some reasonable odds it might take more than one or two meetings to r=
esolve this if we end up in an alternative consensus process.=20

You said on the list you felt "the state of video codec development needs t=
o evolve" before this discussions. I did not know what you what you see nee=
ds to happen in this evolution and how long should we wait before on this. =
I was wondering if you meant we should wait for the IETF to standardize a v=
ideo codec (like the netvc BOF is proposing) but I really had no idea what =
you meant=20

Do you think there would be a better time to discuss this?

Cullen

So far much of conversation has been folks in the Microsoft camp - keep in =
mind some of that group has said in the past they would be=20

On Oct 19, 2014, at 3:38 PM, Peter Saint-Andre - &yet <peter@andyet.net> wr=
ote:

> On 10/16/14, 11:10 AM, Ted Hardie wrote:
>> During this morning chairs call, we discussed the agenda for the
>> upcoming meeting.  The two big items are JSEP and the video codec
>> discussion, and our current thought is to have the JSEP discussion on
>> Monday and the video codec discussion on Thursday.
>=20
> As far as I can see, the temper of the list has been that it would be pre=
mature and a waste of energy to discuss the video codec topic at IETF 91. D=
o the chairs see things differently?
>=20
> Peter
>=20
> --=20
> Peter Saint-Andre
> https://andyet.com/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 10/20/14, 12:55 PM, Cullen Jennings (fluffy) wrote:
>
> So lets be clear here. Every time we checked, there has been
> consensus that we need an MTI codec. So right now this work is sort
> of blocked on that and we would like to get what people are calling
> webrtc 1.0 out before the end of 2015.  I think that a pragmatic look
> at this would result there is at least some reasonable odds it might
> take more than one or two meetings to resolve this if we end up in an
> alternative consensus process.

Given the parameters being assumed, you're probably right that several 
IETF meeting cycles will likely be needed.

> You said on the list you felt "the state of video codec development
> needs to evolve" before this discussions. I did not know what you
> what you see needs to happen in this evolution and how long should we
> wait before on this. I was wondering if you meant we should wait for
> the IETF to standardize a video codec (like the netvc BOF is
> proposing) but I really had no idea what you meant

IMHO, if that working group had been formed after the video codec BoF 
two years ago, we might be further along toward a real solution.

> Do you think there would be a better time to discuss this?

When we can make a decision on a technical basis, not a political basis. 
The current situation involves "camps", as you say, and I just don't 
think that's a good foundation from which to make decisions.

As things stand today, the most likely outcome appears to be "two MTIs" 
(which we've done in draft-ietf-rtcweb-audio so I suppose it's not the 
end of the world).

Peter

-- 
Peter Saint-Andre
https://andyet.com/


From nobody Mon Oct 20 12:42:07 2014
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From: Emil Ivov <emcho@jitsi.org>
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On 20 Oct 2014 8:55 PM, "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote=
:
>
>
> So lets be clear here.

Good idea!

> Every time we checked, there has been consensus that we need an MTI codec=
.
> So right now this work is sort of blocked on that

So clearly, what work is blocked on that exactly?

> and we would like to get what people are calling webrtc 1.0 out before th=
e end of 2015.

I would love for that to happen as well. There are a bunch of pieces
still being worked on but I can't find any that depend on the MTI
debate happening. (Except for trickle ICE of course! The delay there
is totally not the fault of the authors and entirely caused by the
lack of an MTI video codec ;) ).

>  I think that a pragmatic look at this would result there is at least som=
e reasonable odds it might take more than one or two meetings to resolve th=
is if we end up in an alternative consensus process.

It's probably worth distinguishing between finishing the technical and
the administrative aspects of our work. I am sure that the latter
would be much less dramatic.

> You said on the list you felt "the state of video codec development needs=
 to evolve" before this discussions. I did not know what you what you see n=
eeds to happen in this evolution and how long should we wait before on this=
. I was wondering if you meant we should wait for the IETF to standardize a=
 video codec (like the netvc BOF is proposing) but I really had no idea wha=
t you meant
>
> Do you think there would be a better time to discuss this?

Well, for starters, it might be good to have some form of consensus
already appearing plausible on the mailing list! Right now it seems we
can't even agree to talk about it.

> So far much of conversation has been folks in the Microsoft camp - keep i=
n mind some of that group has said in the past they would be

Hm ... I personally saw strong thread participation from Cisco
employees ... but that might be because we attended different counting
schools! ;)

Emil

--
https://jitsi.org


From nobody Mon Oct 20 14:56:52 2014
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
To: Cullen Jennings <fluffy@cisco.com>
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On 21 Oct 2014 05:55, "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:
>
>
> So lets be clear here. Every time we checked, there has been consensus
that we need an MTI codec. So right now this work is sort of blocked on
that and we would like to get what people are calling webrtc 1.0 out before
the end of 2015.  I think that a pragmatic look at this would result there
is at least some reasonable odds it might take more than one or two
meetings to resolve this if we end up in an alternative consensus process.
>
> You said on the list you felt "the state of video codec development needs
to evolve" before this discussions. I did not know what you what you see
needs to happen in this evolution and how long should we wait before on
this. I was wondering if you meant we should wait for the IETF to
standardize a video codec (like the netvc BOF is proposing) but I really
had no idea what you meant
>
> Do you think there would be a better time to discuss this?
>
> Cullen
>
> So far much of conversation has been folks in the Microsoft camp - keep
in mind some of that group has said in the past they would be

What's the Microsoft camp? This far I can see a VP8/9 camp and a H.264/5
camp and both camps have their reasons for making the respective choice. I
think at this stage it's a religious discussion and not one likely to be
resolved until a completely new and clearly superior codec in all discussed
dimensions comes along. Is there really a point to spend more time
discussing now?

(BTW: I won't be at the meeting, so feel free to waste your time. Just
saying...)

Regards,
Silvia.

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<p dir=3D"ltr"><br>
On 21 Oct 2014 05:55, &quot;Cullen Jennings (fluffy)&quot; &lt;<a href=3D"m=
ailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; So lets be clear here. Every time we checked, there has been consensus=
 that we need an MTI codec. So right now this work is sort of blocked on th=
at and we would like to get what people are calling webrtc 1.0 out before t=
he end of 2015.=C2=A0 I think that a pragmatic look at this would result th=
ere is at least some reasonable odds it might take more than one or two mee=
tings to resolve this if we end up in an alternative consensus process.<br>
&gt;<br>
&gt; You said on the list you felt &quot;the state of video codec developme=
nt needs to evolve&quot; before this discussions. I did not know what you w=
hat you see needs to happen in this evolution and how long should we wait b=
efore on this. I was wondering if you meant we should wait for the IETF to =
standardize a video codec (like the netvc BOF is proposing) but I really ha=
d no idea what you meant<br>
&gt;<br>
&gt; Do you think there would be a better time to discuss this?<br>
&gt;<br>
&gt; Cullen<br>
&gt;<br>
&gt; So far much of conversation has been folks in the Microsoft camp - kee=
p in mind some of that group has said in the past they would be<br></p>
<p dir=3D"ltr">What&#39;s the Microsoft camp? This far I can see a VP8/9 ca=
mp and a H.264/5 camp and both camps have their reasons for making the resp=
ective choice. I think at this stage it&#39;s a religious discussion and no=
t one likely to be resolved until a completely new and clearly superior cod=
ec in all discussed dimensions comes along. Is there really a point to spen=
d more time discussing now?</p>
<p dir=3D"ltr">(BTW: I won&#39;t be at the meeting, so feel free to waste y=
our time. Just saying...)</p>
<p dir=3D"ltr">Regards,<br>
Silvia.<br>
</p>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 20 Oct 2014 15:29:15 -0700
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Subject: Re: [rtcweb] Plan for MTI video codec?
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>From the subject "Plan for MTI video codec", I thought this discussion was
about video codecs, not AMR interop.

An astute observer might take note of the fact that our inability to close
the door on H.264 is providing an opportunity to other royalty-bearing
codecs that want to join the party.

On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (Keith) <
keith.drage@alcatel-lucent.com> wrote:

>  The scenario where you might want to transcode the audio codec is where
> you have wideband audio in both the OPUS and the AMR, and you believe by
> transcoding you can achieve a wideband quality, rather than falling back =
to
> G.711. And for interworking between 3GPP devices that are not using webRT=
C,
> the audio will still be AMR or AMR-WB or EVC, not OPUS.
>
> But audio transcoding is not particularly a concern for me. It is trying
> where possible to avoid video transcoding.
>
> Keith
>
>  ------------------------------
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Andrew
> Allen
> *Sent:* 20 October 2014 14:52
> *To:* chris.cavigioli@intel.com; harald@alvestrand.no;
> bernard.aboba@gmail.com
>
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
> Chris
>
> What is the scenario that dictates the need for direct transcoding betwee=
n
> OPUS and AMR?
>
> All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE
> devices OPUS can be negotiated with non mobile RTCweb devices end to end.
>
> My expectation is that for mobile devices the RTCweb client will also hav=
e
> access to the embedded AMR codec so mobile RTCweb devices can interoperat=
e
> using AMR with non-RTCweb enabled mobile IMS devices that support AMR.
>
> For interoperation between non mobile RTCweb devices and mobile non RTCwe=
b
> IMS devices the non mobile RTCweb device can use G.711 to the transcoder
> for transcoding to AMR on the other side. While not ideal this will be no
> worse than non mobile RTCweb devices interoperating with circuit switched
> only mobile devices which will be forced to interoperate via the PSTN.
>
> As RTCweb becomes universal on mobile devices I expect that support for
> OPUS will become native and mobile IMS devices will also be able to use
> OPUS even when operating without the use of the browser (i.e. in non RTCw=
eb
> mode) for better interoperation with non-mobile RTCweb devices.
>
> The lifetime of most mobile devices is 2 years or less so this is likely
> to be mainly a short term transition issue only.
>
> Andrew
>
>  *From*: Cavigioli, Chris [mailto:chris.cavigioli@intel.com]
> *Sent*: Sunday, October 19, 2014 09:20 PM Eastern Standard Time
> *To*: Harald Alvestrand <harald@alvestrand.no>; Bernard Aboba <
> bernard.aboba@gmail.com>
> *Cc*: rtcweb@ietf.org <rtcweb@ietf.org>
> *Subject*: Re: [rtcweb] Plan for MTI video codec?
>
>
> Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Bernard
> Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> 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
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> 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
>
>

--047d7bacbbdcd846a20505e23fb5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">From the subject &quot;Plan for MTI video codec&quot;, I t=
hought this discussion was about video codecs, not AMR interop.=C2=A0<div><=
br></div><div>An astute observer might take note of the fact that our inabi=
lity to close the door on H.264 is providing an opportunity to other royalt=
y-bearing codecs that want to join the party.</div></div><div class=3D"gmai=
l_extra"><br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 7:34 AM, DR=
AGE, Keith (Keith) <span dir=3D"ltr">&lt;<a href=3D"mailto:keith.drage@alca=
tel-lucent.com" target=3D"_blank">keith.drage@alcatel-lucent.com</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><u></u>






<div lang=3D"EN-US" vlink=3D"purple" link=3D"blue">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">The scenario where you might want to transcode the audio codec is where y=
ou have wideband audio in both the OPUS and the AMR, and you believe by tra=
nscoding
 you can achieve a wideband quality, rather than falling back to G.711. And=
 for interworking between 3GPP devices that are not using webRTC, the audio=
 will still be AMR or AMR-WB or EVC, not OPUS.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">But audio transcoding is not particularly a concern for me. It is trying =
where possible to avoid video transcoding.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT:5px;MARGIN-LEFT:5px;BORDER-LE=
FT:#0000ff 2px solid;MARGIN-RIGHT:0px">
<div lang=3D"en-us" dir=3D"ltr" align=3D"left">
<hr>
<font face=3D"Tahoma"><b>From:</b> rtcweb [mailto:<a href=3D"mailto:rtcweb-=
bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Andrew Allen<br>
<b>Sent:</b> 20 October 2014 14:52<br>
<b>To:</b> <a href=3D"mailto:chris.cavigioli@intel.com" target=3D"_blank">c=
hris.cavigioli@intel.com</a>; <a href=3D"mailto:harald@alvestrand.no" targe=
t=3D"_blank">harald@alvestrand.no</a>; <a href=3D"mailto:bernard.aboba@gmai=
l.com" target=3D"_blank">bernard.aboba@gmail.com</a><div><div class=3D"h5">=
<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</div></div></font><br>
</div><div><div class=3D"h5">
<div></div>
<font style=3D"FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:&#39;Calibri&#39;,&=
#39;sans-serif&#39;"><br>
Chris<br>
<br>
What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?
<br>
<br>
All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.
<br>
<br>
My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.<br>
<br>
For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices
 interoperating with circuit switched only mobile devices which will be for=
ced to interoperate via the PSTN.<br>
<br>
As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with
 non-mobile RTCweb devices. <br>
<br>
The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.<br>
<br>
Andrew</font><br>
=C2=A0<br>
<div style=3D"BORDER-RIGHT:medium none;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df=
 1pt solid;PADDING-LEFT:0in;PADDING-BOTTOM:0in;BORDER-LEFT:medium none;PADD=
ING-TOP:3pt;BORDER-BOTTOM:medium none">
<font style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Tahoma&#39;,&#39;sans-serif&=
#39;"><b>From</b>: Cavigioli, Chris [mailto:<a href=3D"mailto:chris.cavigio=
li@intel.com" target=3D"_blank">chris.cavigioli@intel.com</a>]
<br>
<b>Sent</b>: Sunday, October 19, 2014 09:20 PM Eastern Standard Time<br>
<b>To</b>: Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;; Bernard Aboba &lt;<a href=3D"=
mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</=
a>&gt;
<br>
<b>Cc</b>: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a> &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt; <br>
<b>Subject</b>: Re: [rtcweb] Plan for MTI video codec? <br>
</font>=C2=A0<br>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Arial&#39;,&#39;=
sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">We didn=E2=80=99t have quantitati=
ve data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=93 LTE =
interop is important.=C2=A0 A group of us in GSMA have been looking at WebR=
TC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">WebRTC =E2=80=93 LTE transcoding =
is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processing Requ=
irements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a significant degradation
 imposed on Web =E2=80=93 LTE links if only MTI codecs are used.=C2=A0 The =
only systems to avoid this are those which implement OPTIONAL audio codecs =
to be transcoder-free with LTE.=C2=A0 The same can apply for video codecs.<=
u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">The GSMA whitepaper refers to:<u>=
</u><u></u></span></p>
<p>
<u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Arial&#=
39;,&#39;sans-serif&#39;"><span>1.<span style=3D"FONT:7pt &#39;Times New Ro=
man&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">Quantitative voice-over-LTE =
(VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0 To pa=
raphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p>
<u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Arial&#=
39;,&#39;sans-serif&#39;"><span>2.<span style=3D"FONT:7pt &#39;Times New Ro=
man&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">OPUS =E2=80=93 AMR transcodi=
ng adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub> + T<s=
ub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p>
<u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Arial&#=
39;,&#39;sans-serif&#39;"><span>3.<span style=3D"FONT:7pt &#39;Times New Ro=
man&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">To put these numbers in pers=
pective, it is in general desirable to minimize UE delays to ensure low eno=
ugh end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">This is a complex topic with many=
 network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays (T<sub=
>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS
 =E2=80=93 AMR transcoding in the network imposes an additional 211.5 ms on=
 top of that, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 2=
11.5 ms can be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">CONCLUSION: =C2=A0regardless of M=
TI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE interop =
experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">POSITION:=C2=A0 To be clear, seve=
ral Intel SoCs launched at MWC this year for smartphones and tablets suppor=
t both VP8 and H.264 hardware encode and decode =E2=80=93 so
 Intel is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has h=
igh-performance solutions for transcoding.=C2=A0 However, we wish to inform=
 the industry, with quantitative data, about impact to end-user experience =
of mandating such transcoding so IETF can
 make a data-driven decision on video codecs as well as reflect on the impa=
ct of already having mandated transcoding on the audio side.<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"COLOR:#1f497d;FONT-FAMILY:&#39;Lucida=
 Console&#39;">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;">Intel Corp,
</span><span style=3D"FONT-SIZE:8pt;COLOR:#0070c0;FONT-FAMILY:&#39;Verdana&=
#39;,&#39;sans-serif&#39;">System Architecture and Planning</span><span sty=
le=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Verdana&#39;,&#39;sans-s=
erif&#39;">, Video/Multimedia, Mobile and Communications Group
 (MCG)</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Ve=
rdana&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28415%29%=
20254-4545" value=3D"+14152544545" target=3D"_blank">+1 (415) 254-4545</a> =
mobile</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Ve=
rdana&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28408%29%=
20653-8348" value=3D"+14086538348" target=3D"_blank">+1 (408) 653-8348</a> =
desk<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28408%29%=
20884-2400" value=3D"+14088842400" target=3D"_blank">+1 (408) 884-2400</a> =
fax</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Verda=
na&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:7.5pt;COLOR:gray;FONT-FAMIL=
Y:&#39;Verdana&#39;,&#39;sans-serif&#39;">This e-mail may contain confident=
ial and privileged material for the sole use of the intended recipient(s).=
=C2=A0 Any review or distribution by others is strictly
 prohibited. If you are not the intended recipient, please contact the send=
er and delete all copies.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Ta=
homa&#39;,&#39;sans-serif&#39;">From:</span></b><span style=3D"FONT-SIZE:10=
pt;FONT-FAMILY:&#39;Tahoma&#39;,&#39;sans-serif&#39;"> rtcweb [mailto:<a hr=
ef=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf=
.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39=
;Arial&#39;,&#39;sans-serif&#39;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><span style=3D"FONT-SIZ=
E:10pt;FONT-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Arial=
&#39;,&#39;sans-serif&#39;">So it can&#39;t be all bad.....</span>&quot;<u>=
</u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"MARGIN-TOP:5pt;MARGIN-BOTTOM:5pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u></u>
<u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"COLOR:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt">On Oct 16, 2014, at 2:2=
8 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" target=
=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA
<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-LEFT:0in;VERTICAL-ALIGN:baseline;LIN=
E-HEIGHT:14.4pt">
<u></u><span style=3D"FONT-SIZE:10pt;COLOR:#333333;FONT-FAMILY:Symbol"><spa=
n>=C2=B7<span style=3D"FONT:7pt &#39;Times New Roman&#39;">=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:8.5pt;COLOR:#333333;FO=
NT-FAMILY:inherit"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><u></u>=C2=A0<u></u></p=
>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><u></u>=C2=A0<u></u></p=
>
</div>
</div>
<pre><span style=3D"COLOR:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"COLOR:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div></div></blockquote>
</div>

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

--047d7bacbbdcd846a20505e23fb5--


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Subject: Re: [rtcweb] Plan for MTI video codec?
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Content-Type: text/plain; charset=UTF-8

Unless H.264 became royalty free or royalty free profile for H.264 is
developed (and I am talking about ability to include any implementation of
this H.264 profile into any project without paying royalties, not linking
to something the end user must  download), H.264 would not be acceptable to
me as an MTI. All the other news updates regarding H.264 are irrelevant. I
would assume most open source developer or smaller software developers
would share my opinion.

Settled law suits regarding VP8 also do not provide a significant change
regarding video MTI.

There is nothing here to discuss except that we all want video MTI, but we
cannot agree on one. Why waste time on this?

_____________
Roman Shpount

On Sun, Oct 19, 2014 at 2:13 PM, Jonathan Rosenberg <jdrosen@jdrosen.net>
wrote:

> @Alexandre - you say "Today, nothing has changed with respect to those
> two items (even though providing open264 royalties and absorbing the
> license cost for some platforms might have been a set in the right
> direction). ". But, as you say, the availability of Firefox with H264 is a
> change (previously it was not yet available); the fact that Cisco has in
> fact fronted the cost is a change (at the last meeting some were skeptical
> this would happen, but it has). The other big news was IOS8, which now
> enables apps to access H264 and Apple pays the cost. Last time, the lack of
> a solution on IOS was a big deal. That is also now, no longer an issue. As
> such I think there are material changes.
>
>
> On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
>> @jonathan,
>>
>> while you are right and availability of 264 implementation or hardware
>> acceleration has improved, it has never been reported as a problem in the
>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were,
>> AFAIR, the main reasons used by individuals to justify their positions.
>> Today, nothing has changed with respect to those two items (even though
>> providing open264 royalties and absorbing the license cost for some
>> platforms might have been a set in the right direction). This is why I
>> think the conditions are not met for a consensus to be reached.
>>
>> Alex.
>>
>>
>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.com>
>> wrote:
>>
>>> "And its one of the issues holding up wider adoption of the technology"
>>>
>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>>> interoperability.  It is also necessary to specify the transport in enough
>>> detail to allow independent implementations that interoperate well enough
>>> to be usable in a wide variety of scenarios, including wireless networks
>>> where loss is commonly experienced.
>>>
>>> We made the mistake of having an MTI discussion previously with not
>>> enough details on that subject, particularly with respect to H.264.
>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>>
>>> So if we are to have the discussion again, it should occur in the
>>> context of detailed specifications and interoperability reports.
>>>
>>>
>>>
>>>
>>>
>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net
>>> > wrote:
>>>
>>>> I'm in favor of taking another run at this.
>>>>
>>>> The working group has repeatedly said that an MTI codec is something we
>>>> need to produce. And its one of the issues holding up wider adoption of the
>>>> technology (not the only one for sure).
>>>>
>>>> And things have changed since the last meeting, a year ago now
>>>> (November in Vancouver). Cisco's open264 plugin shipped and now just
>>>> recently is integrated into Firefox. iOS8 shipped with APIs for H264. There
>>>> are other things worth noting. Will this change the minds of everyone?
>>>> Surely not. Will it sway enough for us to achieve rough consensus? Maybe.
>>>> IMHO - worth finding out.
>>>>
>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>>>> agouaillard@gmail.com> wrote:
>>>>
>>>>> +1 to not having MTI codec discussion unless some progress has been
>>>>> made on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that
>>>>> there was no change on the members position.
>>>>>
>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <
>>>>> harald@alvestrand.no> wrote:
>>>>>
>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>
>>>>>>> One thing we could do instead of wasting time on MTI is to actually
>>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>>>>>>> actually interoperate regardless of the codec.
>>>>>>>
>>>>>>
>>>>>> The big argument for an MTI is actually the one stated in -overview:
>>>>>> It guards against interoperability failure.
>>>>>>
>>>>>> I would like to have an ecosystem where one can field a box, connect
>>>>>> it to everything else, and run well for *some* level of "well" - and I
>>>>>> would prefer that ecosystem to be one where it's possible to field the box
>>>>>> without making prior arrangements with anyone about licenses.
>>>>>>
>>>>>> This argument hasn't changed one whit since last time we discussed
>>>>>> it. And I don't see much movement on the specifics of the proposals either.
>>>>>>
>>>>>> We'll have to interoperate well with the codecs we field. So using
>>>>>> some time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1
>>>>>> isn't finished either. There's one sentence that needs to be removed.)
>>>>>>
>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>>>>>> prefer to re-discuss based on the knowledge that some significant players
>>>>>> have actually changed their minds.
>>>>>>
>>>>>> At the moment, I don't see signs that any of the poll respondents
>>>>>> have said "I have changed my mind".
>>>>>>
>>>>>> Harald
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <
>>>>>>>> martin.thomson@gmail.com> wrote:
>>>>>>>>
>>>>>>>>  On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
>>>>>>>>> wrote:
>>>>>>>>> And that's because something substantive has changed, or simply
>>>>>>>>> because
>>>>>>>>> wasting the WG time on this again is more entertaining than
>>>>>>>>> actually
>>>>>>>>> finishing a specification that can be independently implemented by
>>>>>>>>> all
>>>>>>>>> browser vendors? (A specification that we are nowhere near having,
>>>>>>>>> as far as
>>>>>>>>> I can tell)
>>>>>>>>>
>>>>>>>> Personally, I've found the reprieve from this fight refreshing.  And
>>>>>>>> it would appear that we've made some real progress as a result.  I'd
>>>>>>>> suggest that if we don't have new information, we continue to spend
>>>>>>>> our time productively.  If we can't find topics to occupy our
>>>>>>>> meeting
>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>
>>>>> ------------------------------------------------------------------------------------
>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>> President - CoSMo Software, Cambridge, MA
>>>>>
>>>>> ------------------------------------------------------------------------------------
>>>>> sg.linkedin.com/agouaillard
>>>>>
>>>>>    -
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Jonathan Rosenberg, Ph.D.
>>>> jdrosen@jdrosen.net
>>>> http://www.jdrosen.net
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>
>>
>>
>> --
>> Alex. Gouaillard, PhD, PhD, MBA
>>
>> ------------------------------------------------------------------------------------
>> CTO - Temasys Communications, S'pore / Mountain View
>> President - CoSMo Software, Cambridge, MA
>>
>> ------------------------------------------------------------------------------------
>> sg.linkedin.com/agouaillard
>>
>>    -
>>
>>
>
>
> --
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> http://www.jdrosen.net
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--e89a8f839d293048050505e27c75
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Unless H.264 became royalty free or royalty free profile f=
or H.264 is developed (and I am talking about ability to include any implem=
entation of this H.264 profile into any project without paying royalties, n=
ot linking to something the end user must =C2=A0download), H.264 would not =
be acceptable to me as an MTI. All the other news updates regarding H.264 a=
re irrelevant. I would assume most open source developer or smaller softwar=
e developers would share my opinion.<div><br></div><div>Settled law suits r=
egarding VP8 also do not provide a significant change regarding video MTI.<=
/div><div><br></div><div>There is nothing here to discuss except that we al=
l want video MTI, but we cannot agree on one. Why waste time on this?</div>=
</div><div class=3D"gmail_extra"><br clear=3D"all"><div>_____________<br>Ro=
man Shpount</div>
<br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 2:13 PM, Jonathan Ro=
senberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=
=3D"_blank">jdrosen@jdrosen.net</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 dir=3D"ltr">@Alexandre - you say &quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">Today, nothing has changed with =
respect to those two items (even though providing open264 royalties and abs=
orbing the license cost for some platforms might have been a set in the rig=
ht direction).=C2=A0&quot;. But, as you say, the availability of Firefox wi=
th H264 is a change (previously it was not yet available); the fact that Ci=
sco has in fact fronted the cost is a change (at the last meeting some were=
 skeptical this would happen, but it has). The other big news was IOS8, whi=
ch now enables apps to access H264 and Apple pays the cost. Last time, the =
lack of a solution on IOS was a big deal. That is also now, no longer an is=
sue. As such I think there are material changes.</span><div><span style=3D"=
font-family:arial,sans-serif;font-size:13px"><br></span></div></div><div cl=
ass=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blan=
k">agouaillard@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div dir=3D"ltr">@jonathan,<div><br></div><div>while you are right an=
d availability of 264 implementation or hardware acceleration has improved,=
 it has never been reported as a problem in the previous pool by anyone. Th=
e 264 royalties, and the VP8 IP risks were, AFAIR, the main reasons used by=
 individuals to justify their positions. Today, nothing has changed with re=
spect to those two items (even though providing open264 royalties and absor=
bing the license cost for some platforms might have been a set in the right=
 direction). This is why I think the conditions are not met for a consensus=
 to be reached.</div><div><br></div><div>Alex.</div><div><br></div></div><d=
iv><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, O=
ct 19, 2014 at 11:43 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span>=
&quot;And its one of the issues holding up wider adoption of the technology=
&quot;<div><br></div></span><div>[BA] Specifying an MTI encoder/decoder is =
not sufficient for interoperability.=C2=A0 It is also necessary to specify =
the transport in enough detail to allow independent implementations that in=
teroperate well enough to be usable in a wide variety of scenarios, includi=
ng wireless networks where loss is commonly experienced. =C2=A0</div><div><=
br></div><div>We made the mistake of having an MTI discussion previously wi=
th not enough details on that subject, particularly with respect to H.264. =
draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best. =C2=A0</=
div><div><br></div><div>So if we are to have the discussion again, it shoul=
d occur in the context of detailed specifications and interoperability repo=
rts.</div><div><br></div><div>=C2=A0</div><div><br></div><div>=C2=A0</div><=
/div><div><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">I&#=
39;m in favor of taking another run at this.<div><br></div><div>The working=
 group has repeatedly said that an MTI codec is something we need to produc=
e. And its one of the issues holding up wider adoption of the technology (n=
ot the only one for sure).</div><div><br></div><div>And things have changed=
 since the last meeting, a year ago now (November in Vancouver). Cisco&#39;=
s open264 plugin shipped and now just recently is integrated into Firefox. =
iOS8 shipped with APIs for H264. There are other things worth noting. Will =
this change the minds of everyone? Surely not. Will it sway enough for us t=
o achieve rough consensus? Maybe. IMHO - worth finding out.</div></div><div=
 class=3D"gmail_extra"><div><div><br><div class=3D"gmail_quote">On Sat, Oct=
 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D=
"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmail.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">+1 to =
not having MTI codec discussion unless some progress has been made on VP8 a=
t MPEG.=C2=A0Any news on that? I&#39;m sharing harald&#39;s =C2=A0feeling t=
hat there was no change on the members position.=C2=A0</div><div class=3D"g=
mail_extra"><div><div><br><div class=3D"gmail_quote">On Fri, Oct 17, 2014 a=
t 9:22 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald=
@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><span>On 10/17/2014 12:02 AM, Bernard A=
boba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.th=
omson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<=
br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@mat=
thew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<br>
</blockquote>
Personally, I&#39;ve found the reprieve from this fight refreshing.=C2=A0 A=
nd<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div><=
/div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard=
, PhD, PhD, MBA<div>-------------------------------------------------------=
-----------------------------</div><div>CTO - Temasys Communications, S&#39=
;pore / Mountain View</div><div>President - CoSMo Software, Cambridge, MA</=
div><div>------------------------------------------------------------------=
------------------</div><div><a href=3D"http://sg.linkedin.com/agouaillard"=
 target=3D"_blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"m=
argin:0px;padding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-fa=
mily:Helvetica,Arial,sans-serif;vertical-align:baseline;list-style:none;lin=
e-height:17px;display:table-cell;width:504px;color:rgb(51,51,51)"><li style=
=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline:0px;font-style:i=
nherit;font-size:11px;font-family:inherit;vertical-align:baseline;font-vari=
ant:inherit;line-height:1.2em"><dl style=3D"margin:0px;padding:0px;border:0=
px;outline:0px;font-style:inherit;font-family:inherit;vertical-align:baseli=
ne;font-variant:inherit;line-height:inherit;word-wrap:break-word"><br></dl>=
</li></ul></div></div>
</font></span></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"><div><br></div>-- <br></div></=
div><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen=
@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http:=
//www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>----------------------=
--------------------------------------------------------------</div><div>CT=
O - Temasys Communications, S&#39;pore / Mountain View</div><div>President =
- CoSMo Software, Cambridge, MA</div><div>---------------------------------=
---------------------------------------------------</div><div><a href=3D"ht=
tp://sg.linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouail=
lard</a></div><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;o=
utline:0px;font-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-a=
lign:baseline;list-style:none;line-height:17px;display:table-cell;width:504=
px;color:rgb(51,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;bo=
rder:0px;outline:0px;font-style:inherit;font-size:11px;font-family:inherit;=
vertical-align:baseline;font-variant:inherit;line-height:1.2em"><dl style=
=3D"margin:0px;padding:0px;border:0px;outline:0px;font-style:inherit;font-f=
amily:inherit;vertical-align:baseline;font-variant:inherit;line-height:inhe=
rit;word-wrap:break-word"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdr=
osen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://ww=
w.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--e89a8f839d293048050505e27c75--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 21 Oct 2014 00:50:59 +0200
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To: Roman Shpount <roman@telurix.com>
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2014-10-21 0:46 GMT+02:00 Roman Shpount <roman@telurix.com>:
> Unless H.264 became royalty free or royalty free profile for H.264 is
> developed (and I am talking about ability to include any implementation o=
f
> this H.264 profile into any project without paying royalties, not linking=
 to
> something the end user must  download), H.264 would not be acceptable to =
me
> as an MTI. All the other news updates regarding H.264 are irrelevant. I
> would assume most open source developer or smaller software developers wo=
uld
> share my opinion.

+1


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


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From: Barry Dingle <btdingle@gmail.com>
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--001a11349882e979090505e30d13
Content-Type: text/plain; charset=UTF-8

Martin,

Sorry for the confusion. I was trying to use 'network scenario' as a
general term to encompass different end-to-end call situations that involve
my WebRTC network and the destination network e.g. mobile, PSTN, VoIP,
existing Video Services etc. (Opposite call directions also apply of
course.)

I am suggesting that there should be MTI's for several major end-to-end
network scenarios as that is something that the IETF can specify. HOW the
small range of MTI's are Implemented is up to Browsers and the Browsers can
be categorised according to their support of the different network
scenarios.

The Browsers are the key to what codecs are supported. Trying to force
people by Specification to implement something they do not want to support
(for whatever reason) is what is causing us this video codec MTI debate to
be unsolvable.

The IETF Specifications should describe what protocols etc are needed for
interoperability. A separate document can describe which options (in this
case codecs) to use in different end-to-end network scenarios. That is, the
'case-specific' MTIs.

Barry Dingle
Fixed - +61(0)3-9725-3937    Mob - +61(0)41-911-7578
Fellow of University of Melbourne, Australia

On Mon, Oct 20, 2014 at 11:02 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 20 October 2014 01:52, Barry Dingle <btdingle@gmail.com> wrote:
> > Rather, we should have 2 (or more) Network Scenarios and have Audio+Video
> > MTI's for Each Network Scenario.
>
>
> Why should your network attachment conditions determine who you can talk
> to?
>

--001a11349882e979090505e30d13
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div>Martin, <br><br></div>Sorry for the confusion. I=
 was trying to use &#39;network scenario&#39; as a general term to encompas=
s different end-to-end call situations that involve my WebRTC network and t=
he destination network e.g. mobile, PSTN, VoIP, existing Video Services etc=
. (Opposite call directions also apply of course.) <br><br></div>I am sugge=
sting that there should be MTI&#39;s for several major end-to-end network s=
cenarios as that is something that the IETF can specify. HOW the small rang=
e of MTI&#39;s are Implemented is up to Browsers and the Browsers can be ca=
tegorised according to their support of the different network scenarios.=C2=
=A0 <br><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">The=
 Browsers are the key to what codecs are supported. Trying to force people =
by Specification to implement something they do not want to support (for wh=
atever reason) is what is causing us this video codec MTI debate to be unso=
lvable. <br><br></div><div class=3D"gmail_extra">The IETF Specifications sh=
ould describe what protocols etc are needed for interoperability. A separat=
e document can describe which options (in this case codecs) to use in diffe=
rent end-to-end network scenarios. That is, the &#39;case-specific&#39; MTI=
s. <br></div><div class=3D"gmail_extra"><div><div dir=3D"ltr"><br>Barry Din=
gle<br>Fixed - +61(0)3-9725-3937=C2=A0 =C2=A0 Mob - +61(0)41-911-7578=C2=A0=
=C2=A0 <br>Fellow of University of Melbourne, Australia <br></div></div>
<br><div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 11:02 PM, Martin Tho=
mson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px so=
lid rgb(204,204,204);padding-left:1ex"><span class=3D"">On 20 October 2014 =
01:52, Barry Dingle &lt;<a href=3D"mailto:btdingle@gmail.com">btdingle@gmai=
l.com</a>&gt; wrote:<br>
&gt; Rather, we should have 2 (or more) Network Scenarios and have Audio+Vi=
deo<br>
&gt; MTI&#39;s for Each Network Scenario.<br>
<br>
<br>
</span>Why should your network attachment conditions determine who you can =
talk to?<br>
</blockquote></div><br></div></div>

--001a11349882e979090505e30d13--


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On Oct 20, 2014, at 3:29 PM, Justin Uberti <juberti@google.com> wrote:
>=20
> An astute observer might take note of the fact that our inability to close=
 the door on H.264 is providing an opportunity to other royalty-bearing code=
cs that want to join the party.

[BA] IETF WG Chairs have the ability to avoid revisiting previously settled i=
ssues (e.g. Audio codecs) or unsettled ones (e.g. Video codecs). =20

BTW, I've had difficulty verifying the assertion that "AMR codecs are access=
ible via the hardware without additional fees". There may be an SDK availabl=
e, but my understanding is that this requires a licensing fee above and beyo=
nd the fee required to license AMR codecs in Voice over LTE. In other words,=
 not only is the codec royalty bearing, but access to it for a purpose other=
 than VoLTE requires additional licensing fees.  If this is wrong, please po=
int me to the web page for the appropriate license-free SDKs.=


From nobody Mon Oct 20 21:38:15 2014
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From: Victor Pascual <victor.pascual.avila@gmail.com>
Date: Tue, 21 Oct 2014 06:37:45 +0200
To: Roman Shpount <roman@telurix.com>
Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/ztN8eiih5ftW8QwefL1FGIED2LQ
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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Sounds like only consensus so far is we agree to disagree

> On Oct 21, 2014, at 12:46 AM, Roman Shpount <roman@telurix.com> wrote:
>=20
> Unless H.264 became royalty free or royalty free profile for H.264 is deve=
loped (and I am talking about ability to include any implementation of this H=
.264 profile into any project without paying royalties, not linking to somet=
hing the end user must  download), H.264 would not be acceptable to me as an=
 MTI. All the other news updates regarding H.264 are irrelevant. I would ass=
ume most open source developer or smaller software developers would share my=
 opinion.
>=20
> Settled law suits regarding VP8 also do not provide a significant change r=
egarding video MTI.
>=20
> There is nothing here to discuss except that we all want video MTI, but we=
 cannot agree on one. Why waste time on this?
>=20
> _____________
> Roman Shpount
>=20
>> On Sun, Oct 19, 2014 at 2:13 PM, Jonathan Rosenberg <jdrosen@jdrosen.net>=
 wrote:
>> @Alexandre - you say "Today, nothing has changed with respect to those tw=
o items (even though providing open264 royalties and absorbing the license c=
ost for some platforms might have been a set in the right direction). ". But=
, as you say, the availability of Firefox with H264 is a change (previously i=
t was not yet available); the fact that Cisco has in fact fronted the cost i=
s a change (at the last meeting some were skeptical this would happen, but i=
t has). The other big news was IOS8, which now enables apps to access H264 a=
nd Apple pays the cost. Last time, the lack of a solution on IOS was a big d=
eal. That is also now, no longer an issue. As such I think there are materia=
l changes.
>>=20
>>=20
>>> On Sun, Oct 19, 2014 at 12:13 PM, Alexandre GOUAILLARD <agouaillard@gmai=
l.com> wrote:
>>> @jonathan,
>>>=20
>>> while you are right and availability of 264 implementation or hardware a=
cceleration has improved, it has never been reported as a problem in the pre=
vious pool by anyone. The 264 royalties, and the VP8 IP risks were, AFAIR, t=
he main reasons used by individuals to justify their positions. Today, nothi=
ng has changed with respect to those two items (even though providing open26=
4 royalties and absorbing the license cost for some platforms might have bee=
n a set in the right direction). This is why I think the conditions are not m=
et for a consensus to be reached.
>>>=20
>>> Alex.
>>>=20
>>>=20
>>>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <bernard.aboba@gmail.co=
m> wrote:
>>>> "And its one of the issues holding up wider adoption of the technology"=

>>>>=20
>>>> [BA] Specifying an MTI encoder/decoder is not sufficient for interopera=
bility.  It is also necessary to specify the transport in enough detail to a=
llow independent implementations that interoperate well enough to be usable i=
n a wide variety of scenarios, including wireless networks where loss is com=
monly experienced. =20
>>>>=20
>>>> We made the mistake of having an MTI discussion previously with not eno=
ugh details on that subject, particularly with respect to H.264. draft-ietf-=
rtcweb-video sections 4.2 - 4.4 remain sketchy at best. =20
>>>>=20
>>>> So if we are to have the discussion again, it should occur in the conte=
xt of detailed specifications and interoperability reports.
>>>>=20
>>>> =20
>>>>=20
>>>> =20
>>>>=20
>>>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.n=
et> wrote:
>>>>> I'm in favor of taking another run at this.
>>>>>=20
>>>>> The working group has repeatedly said that an MTI codec is something w=
e need to produce. And its one of the issues holding up wider adoption of th=
e technology (not the only one for sure).
>>>>>=20
>>>>> And things have changed since the last meeting, a year ago now (Novemb=
er in Vancouver). Cisco's open264 plugin shipped and now just recently is in=
tegrated into Firefox. iOS8 shipped with APIs for H264. There are other thin=
gs worth noting. Will this change the minds of everyone? Surely not. Will it=
 sway enough for us to achieve rough consensus? Maybe. IMHO - worth finding o=
ut.
>>>>>=20
>>>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <agouaillard@gm=
ail.com> wrote:
>>>>>> +1 to not having MTI codec discussion unless some progress has been m=
ade on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that the=
re was no change on the members position.=20
>>>>>>=20
>>>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestran=
d.no> wrote:
>>>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>>>>>>> One thing we could do instead of wasting time on MTI is to actually=
 make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could=
 actually interoperate regardless of the codec.
>>>>>>>=20
>>>>>>> The big argument for an MTI is actually the one stated in -overview:=
 It guards against interoperability failure.
>>>>>>>=20
>>>>>>> I would like to have an ecosystem where one can field a box, connect=
 it to everything else, and run well for *some* level of "well" - and I woul=
d prefer that ecosystem to be one where it's possible to field the box witho=
ut making prior arrangements with anyone about licenses.
>>>>>>>=20
>>>>>>> This argument hasn't changed one whit since last time we discussed i=
t. And I don't see much movement on the specifics of the proposals either.
>>>>>>>=20
>>>>>>> We'll have to interoperate well with the codecs we field. So using s=
ome time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 is=
n't finished either. There's one sentence that needs to be removed.)
>>>>>>>=20
>>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd=
 prefer to re-discuss based on the knowledge that some significant players h=
ave actually changed their minds.
>>>>>>>=20
>>>>>>> At the moment, I don't see signs that any of the poll respondents ha=
ve said "I have changed my mind".
>>>>>>>=20
>>>>>>> Harald
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.=
com> wrote:
>>>>>>>>>=20
>>>>>>>>>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wr=
ote:
>>>>>>>>>> And that's because something substantive has changed, or simply b=
ecause
>>>>>>>>>> wasting the WG time on this again is more entertaining than actua=
lly
>>>>>>>>>> finishing a specification that can be independently implemented b=
y all
>>>>>>>>>> browser vendors? (A specification that we are nowhere near having=
, as far as
>>>>>>>>>> I can tell)
>>>>>>>>> Personally, I've found the reprieve from this fight refreshing.  A=
nd
>>>>>>>>> it would appear that we've made some real progress as a result.  I=
'd
>>>>>>>>> suggest that if we don't have new information, we continue to spen=
d
>>>>>>>>> our time productively.  If we can't find topics to occupy our meet=
ing
>>>>>>>>> agenda time, then maybe we can free an agenda slot.
>>>>>>>>>=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
>>>>>>>=20
>>>>>>> _______________________________________________
>>>>>>> rtcweb mailing list
>>>>>>> rtcweb@ietf.org
>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> --=20
>>>>>> Alex. Gouaillard, PhD, PhD, MBA
>>>>>> ---------------------------------------------------------------------=
---------------
>>>>>> CTO - Temasys Communications, S'pore / Mountain View
>>>>>> President - CoSMo Software, Cambridge, MA
>>>>>> ---------------------------------------------------------------------=
---------------
>>>>>> sg.linkedin.com/agouaillard
>>>>>>=20
>>>>>>=20
>>>>>> _______________________________________________
>>>>>> rtcweb mailing list
>>>>>> rtcweb@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> --=20
>>>>> Jonathan Rosenberg, Ph.D.
>>>>> jdrosen@jdrosen.net
>>>>> http://www.jdrosen.net
>>>>>=20
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>>=20
>>>=20
>>> --=20
>>> Alex. Gouaillard, PhD, PhD, MBA
>>> ------------------------------------------------------------------------=
------------
>>> CTO - Temasys Communications, S'pore / Mountain View
>>> President - CoSMo Software, Cambridge, MA
>>> ------------------------------------------------------------------------=
------------
>>> sg.linkedin.com/agouaillard
>>=20
>>=20
>>=20
>> --=20
>> Jonathan Rosenberg, Ph.D.
>> jdrosen@jdrosen.net
>> http://www.jdrosen.net
>>=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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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>Sounds like only consensus so far is w=
e agree to disagree<br></div><div><br>On Oct 21, 2014, at 12:46 AM, Roman Sh=
pount &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wro=
te:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr">Unless H.26=
4 became royalty free or royalty free profile for H.264 is developed (and I a=
m talking about ability to include any implementation of this H.264 profile i=
nto any project without paying royalties, not linking to something the end u=
ser must &nbsp;download), H.264 would not be acceptable to me as an MTI. All=
 the other news updates regarding H.264 are irrelevant. I would assume most o=
pen source developer or smaller software developers would share my opinion.<=
div><br></div><div>Settled law suits regarding VP8 also do not provide a sig=
nificant change regarding video MTI.</div><div><br></div><div>There is nothi=
ng here to discuss except that we all want video MTI, but we cannot agree on=
 one. Why waste time on this?</div></div><div class=3D"gmail_extra"><br clea=
r=3D"all"><div>_____________<br>Roman Shpount</div>
<br><div class=3D"gmail_quote">On Sun, Oct 19, 2014 at 2:13 PM, Jonathan Ros=
enberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D=
"_blank">jdrosen@jdrosen.net</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><div dir=3D"ltr">@Alexandre - you say "<span style=3D"font-family:ar=
ial,sans-serif;font-size:13px">Today, nothing has changed with respect to th=
ose two items (even though providing open264 royalties and absorbing the lic=
ense cost for some platforms might have been a set in the right direction).&=
nbsp;". But, as you say, the availability of Firefox with H264 is a change (=
previously it was not yet available); the fact that Cisco has in fact fronte=
d the cost is a change (at the last meeting some were skeptical this would h=
appen, but it has). The other big news was IOS8, which now enables apps to a=
ccess H264 and Apple pays the cost. Last time, the lack of a solution on IOS=
 was a big deal. That is also now, no longer an issue. As such I think there=
 are material changes.</span><div><span style=3D"font-family:arial,sans-seri=
f;font-size:13px"><br></span></div></div><div class=3D"HOEnZb"><div class=3D=
"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Oct 1=
9, 2014 at 12:13 PM, Alexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:agouaillard@gmail.com" target=3D"_blank">agouaillard@gmail.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">@jonathan,<=
div><br></div><div>while you are right and availability of 264 implementatio=
n or hardware acceleration has improved, it has never been reported as a pro=
blem in the previous pool by anyone. The 264 royalties, and the VP8 IP risks=
 were, AFAIR, the main reasons used by individuals to justify their position=
s. Today, nothing has changed with respect to those two items (even though p=
roviding open264 royalties and absorbing the license cost for some platforms=
 might have been a set in the right direction). This is why I think the cond=
itions are not met for a consensus to be reached.</div><div><br></div><div>A=
lex.</div><div><br></div></div><div><div><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote">On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank=
">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex"><div dir=3D"ltr"><span>"And its one of the issues holding up wider adop=
tion of the technology"<div><br></div></span><div>[BA] Specifying an MTI enc=
oder/decoder is not sufficient for interoperability.&nbsp; It is also necess=
ary to specify the transport in enough detail to allow independent implement=
ations that interoperate well enough to be usable in a wide variety of scena=
rios, including wireless networks where loss is commonly experienced. &nbsp;=
</div><div><br></div><div>We made the mistake of having an MTI discussion pr=
eviously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best. &=
nbsp;</div><div><br></div><div>So if we are to have the discussion again, it=
 should occur in the context of detailed specifications and interoperability=
 reports.</div><div><br></div><div>&nbsp;</div><div><br></div><div>&nbsp;</d=
iv></div><div><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <span dir=3D"ltr">&lt;<=
a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.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"><div dir=3D"ltr">I'm i=
n favor of taking another run at this.<div><br></div><div>The working group h=
as repeatedly said that an MTI codec is something we need to produce. And it=
s one of the issues holding up wider adoption of the technology (not the onl=
y one for sure).</div><div><br></div><div>And things have changed since the l=
ast meeting, a year ago now (November in Vancouver). Cisco's open264 plugin s=
hipped and now just recently is integrated into Firefox. iOS8 shipped with A=
PIs for H264. There are other things worth noting. Will this change the mind=
s of everyone? Surely not. Will it sway enough for us to achieve rough conse=
nsus? Maybe. IMHO - worth finding out.</div></div><div class=3D"gmail_extra"=
><div><div><br><div class=3D"gmail_quote">On Sat, Oct 18, 2014 at 5:08 PM, A=
lexandre GOUAILLARD <span dir=3D"ltr">&lt;<a href=3D"mailto:agouaillard@gmai=
l.com" target=3D"_blank">agouaillard@gmail.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"ltr">+1 to not having MTI codec discus=
sion unless some progress has been made on VP8 at MPEG.&nbsp;Any news on tha=
t? I'm sharing harald's &nbsp;feeling that there was no change on the member=
s position.&nbsp;</div><div class=3D"gmail_extra"><div><div><br><div class=3D=
"gmail_quote">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <span dir=3D=
"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@a=
lvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On=
 10/17/2014 12:02 AM, Bernard Aboba wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
One thing we could do instead of wasting time on MTI is to actually make pro=
gress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actually=
 interoperate regardless of the codec.<br>
</blockquote>
<br></span>
The big argument for an MTI is actually the one stated in -overview: It guar=
ds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to e=
verything else, and run well for *some* level of "well" - and I would prefer=
 that ecosystem to be one where it's possible to field the box without makin=
g prior arrangements with anyone about licenses.<br>
<br>
This argument hasn't changed one whit since last time we discussed it. And I=
 don't see much movement on the specifics of the proposals either.<br>
<br>
We'll have to interoperate well with the codecs we field. So using some time=
 to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fini=
shed either. There's one sentence that needs to be removed.)<br>
<br>
I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer t=
o re-discuss based on the knowledge that some significant players have actua=
lly changed their minds.<br>
<br>
At the moment, I don't see signs that any of the poll respondents have said "=
I have changed my mind".<span><font color=3D"#888888"><br>
<br>
Harald</font></span><div><div><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
On Oct 16, 2014, at 2:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.tho=
mson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<br=
>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
On 16 October 2014 14:17, Matthew Kaufman &lt;<a href=3D"mailto:matthew@matt=
hew.at" target=3D"_blank">matthew@matthew.at</a>&gt; wrote:<br>
And that's because something substantive has changed, or simply because<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far as=
<br>
I can tell)<br>
</blockquote>
Personally, I've found the reprieve from this fight refreshing.&nbsp; And<br=
>
it would appear that we've made some real progress as a result.&nbsp; I'd<br=
>
suggest that if we don't have new information, we continue to spend<br>
our time productively.&nbsp; If we can't find topics to occupy our meeting<b=
r>
agenda time, then maybe we can free an agenda slot.<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=

<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=

<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=

<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div></div></=
div><span><font color=3D"#888888">-- <br><div dir=3D"ltr">Alex. Gouaillard, P=
hD, PhD, MBA<div>-----------------------------------------------------------=
-------------------------</div><div>CTO - Temasys Communications, S'pore / M=
ountain View</div><div>President - CoSMo Software, Cambridge, MA</div><div>-=
----------------------------------------------------------------------------=
-------</div><div><a href=3D"http://sg.linkedin.com/agouaillard" target=3D"_=
blank">sg.linkedin.com/agouaillard</a></div><div><ul style=3D"margin:0px;pad=
ding:0px 0px 8px;border:0px;outline:0px;font-size:12px;font-family:Helvetica=
,Arial,sans-serif;vertical-align:baseline;list-style:none;line-height:17px;d=
isplay:table-cell;width:504px;color:rgb(51,51,51)"><li style=3D"margin:0px;p=
adding:8px 12px 2px 0px;border:0px;outline:0px;font-style:inherit;font-size:=
11px;font-family:inherit;vertical-align:baseline;font-variant:inherit;line-h=
eight:1.2em"><dl style=3D"margin:0px;padding:0px;border:0px;outline:0px;font=
-style:inherit;font-family:inherit;vertical-align:baseline;font-variant:inhe=
rit;line-height:inherit;word-wrap:break-word"><br></dl></li></ul></div></div=
>
</font></span></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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br></div></d=
iv><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@j=
drosen.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://w=
ww.jdrosen.net" target=3D"_blank">http://www.jdrosen.net</a></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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><=
div dir=3D"ltr">Alex. Gouaillard, PhD, PhD, MBA<div>------------------------=
------------------------------------------------------------</div><div>CTO -=
 Temasys Communications, S'pore / Mountain View</div><div>President - CoSMo S=
oftware, Cambridge, MA</div><div>-------------------------------------------=
-----------------------------------------</div><div><a href=3D"http://sg.lin=
kedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a></di=
v><div><ul style=3D"margin:0px;padding:0px 0px 8px;border:0px;outline:0px;fo=
nt-size:12px;font-family:Helvetica,Arial,sans-serif;vertical-align:baseline;=
list-style:none;line-height:17px;display:table-cell;width:504px;color:rgb(51=
,51,51)"><li style=3D"margin:0px;padding:8px 12px 2px 0px;border:0px;outline=
:0px;font-style:inherit;font-size:11px;font-family:inherit;vertical-align:ba=
seline;font-variant:inherit;line-height:1.2em"><dl style=3D"margin:0px;paddi=
ng:0px;border:0px;outline:0px;font-style:inherit;font-family:inherit;vertica=
l-align:baseline;font-variant:inherit;line-height:inherit;word-wrap:break-wo=
rd"><br></dl></li></ul></div></div>
</div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><=
div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a href=3D"mailto:jdrosen@jdros=
en.net" target=3D"_blank">jdrosen@jdrosen.net</a><br><a href=3D"http://www.j=
drosen.net" target=3D"_blank">http://www.jdrosen.net</a></div>
</div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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Subject: Re: [rtcweb] Plan for MTI video codec?
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On 10/21/2014 12:29 AM, Justin Uberti wrote:
> From the subject "Plan for MTI video codec", I thought this discussion
> was about video codecs, not AMR interop.

All discussions branch, but I strongly recommend changing the subject
line when changing the subject.

The discussion is supposed to be managed by the WG chairs; since they
have not spoken up on this thread at all, I must assume that they do not
wish to constrain the discussion at this time.

>
> An astute observer might take note of the fact that our inability to
> close the door on H.264 is providing an opportunity to other
> royalty-bearing codecs that want to join the party.
>
> On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (Keith)
> <keith.drage@alcatel-lucent.com
> <mailto:keith.drage@alcatel-lucent.com>> wrote:
>
>     The scenario where you might want to transcode the audio codec is
>     where you have wideband audio in both the OPUS and the AMR, and
>     you believe by transcoding you can achieve a wideband quality,
>     rather than falling back to G.711. And for interworking between
>     3GPP devices that are not using webRTC, the audio will still be
>     AMR or AMR-WB or EVC, not OPUS.
>      
>     But audio transcoding is not particularly a concern for me. It is
>     trying where possible to avoid video transcoding.
>      
>     Keith
>
>         ------------------------------------------------------------------------
>         *From:* rtcweb [mailto:rtcweb-bounces@ietf.org
>         <mailto:rtcweb-bounces@ietf.org>] *On Behalf Of *Andrew Allen
>         *Sent:* 20 October 2014 14:52
>         *To:* chris.cavigioli@intel.com
>         <mailto:chris.cavigioli@intel.com>; harald@alvestrand.no
>         <mailto:harald@alvestrand.no>; bernard.aboba@gmail.com
>         <mailto:bernard.aboba@gmail.com>
>
>         *Cc:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>         Chris
>
>         What is the scenario that dictates the need for direct
>         transcoding between OPUS and AMR?
>
>         All RTCweb clients MUST support OPUS so for Web browsers on
>         mobile LTE devices OPUS can be negotiated with non mobile
>         RTCweb devices end to end.
>
>         My expectation is that for mobile devices the RTCweb client
>         will also have access to the embedded AMR codec so mobile
>         RTCweb devices can interoperate using AMR with non-RTCweb
>         enabled mobile IMS devices that support AMR.
>
>         For interoperation between non mobile RTCweb devices and
>         mobile non RTCweb IMS devices the non mobile RTCweb device can
>         use G.711 to the transcoder for transcoding to AMR on the
>         other side. While not ideal this will be no worse than non
>         mobile RTCweb devices interoperating with circuit switched
>         only mobile devices which will be forced to interoperate via
>         the PSTN.
>
>         As RTCweb becomes universal on mobile devices I expect that
>         support for OPUS will become native and mobile IMS devices
>         will also be able to use OPUS even when operating without the
>         use of the browser (i.e. in non RTCweb mode) for better
>         interoperation with non-mobile RTCweb devices.
>
>         The lifetime of most mobile devices is 2 years or less so this
>         is likely to be mainly a short term transition issue only.
>
>         Andrew
>          
>         *From*: Cavigioli, Chris [mailto:chris.cavigioli@intel.com
>         <mailto:chris.cavigioli@intel.com>]
>         *Sent*: Sunday, October 19, 2014 09:20 PM Eastern Standard Time
>         *To*: Harald Alvestrand <harald@alvestrand.no
>         <mailto:harald@alvestrand.no>>; Bernard Aboba
>         <bernard.aboba@gmail.com <mailto:bernard.aboba@gmail.com>>
>         *Cc*: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>         *Subject*: Re: [rtcweb] Plan for MTI video codec?
>          
>
>         Harald said: 
>
>         â€œWe made the mistake of having an MTI discussion previously
>         with not enough details on that subjectâ€¦â€
>
>          
>
>         We didnâ€™t have quantitative data earlier for a data-driven
>         decision.  Future Web â€“ LTE interop is important.  A group of
>         us in GSMA have been looking at WebRTC interop with LTE (where
>         3GPP / GSMA defined IMS-based voice/video/messaging),
>         specifically looking at user experience impact of latency
>         introduced by added in-network transcoding.  GSMA has approved
>         the whitepaper and it is being prepared for public distribution. 
>
>          
>
>         WebRTC â€“ LTE transcoding is now MANDATORY because of â€œWebRTC
>         Audio Codec and Processing Requirementsâ€.
>         http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where
>         MTI audio codecs in WebRTC and in LTE are dissimilar.  Thus,
>         REGARDLESS of the video codecs, there is already a significant
>         degradation imposed on Web â€“ LTE links if only MTI codecs are
>         used.  The only systems to avoid this are those which
>         implement OPTIONAL audio codecs to be transcoder-free with
>         LTE.  The same can apply for video codecs.
>
>          
>
>         The GSMA whitepaper refers to:
>
>         1.     Quantitative voice-over-LTE (VoLTE) delay limits were
>         agreed-to by 3GPP SA4 in August 2014.  To paraphrase,
>         performance objectives for maximum VoLTE device delays in
>         error and jitter free conditions and AMR speech codec
>         operation, the sum of the UE delays in sending and receiving
>         directions (T_S + T_R ) should be â‰¤ 150ms. If this performance
>         objective cannot be met, the sum of the UE delays in sending
>         and receiving directions (T_S + T_R ) shall in any case be â‰¤
>         190ms.
>
>         2.     OPUS â€“ AMR transcoding adds an additional 211.5 ms
>         implementation-agnostic (T_S + T_R ) delay, including 40 ms
>         for jitter buffer and 40 ms for discontinuous reception (DRX).
>
>         3.     To put these numbers in perspective, it is in general
>         desirable to minimize UE delays to ensure low enough
>         end-to-end delays and hence a good conversational experience. 
>         Increasing delay causes people to double-talk making
>         conversations increasingly frustrating.  Guidance to stay
>         below 400 ms maximum one-way delay is found in ITU-T
>         Recommendation G.114 and the computational E-model is in ITU-T
>         Recommendation G.107.
>
>          
>
>         This is a complex topic with many network factors in play. 
>         LTE operators in 3GPP seek UE delays (T_S + T_R ) around 150
>         ms.  Adding OPUS â€“ AMR transcoding in the network imposes an
>         additional 211.5 ms on top of that, just for audio. 
>         Optionally using AMR in WebRTC, those 211.5 ms can be
>         eliminated, delivering a superior end-user experience. 
>
>          
>
>         CONCLUSION:  regardless of MTI codec choices in WebRTC, to
>         provide a good WebRTC â€“ LTE interop experience, emerging
>         WebRTC systems must accommodate MTI codecs already deployed in
>         the LTE domain and in mobile operating systems, namely
>         AMR/AMR-WB and H.264.
>
>          
>
>         POSITION:  To be clear, several Intel SoCs launched at MWC
>         this year for smartphones and tablets support both VP8 and
>         H.264 hardware encode and decode â€“ so Intel is codec-neutral
>         in this MTI debate.  Additionally, Intel has high-performance
>         solutions for transcoding.  However, we wish to inform the
>         industry, with quantitative data, about impact to end-user
>         experience of mandating such transcoding so IETF can make a
>         data-driven decision on video codecs as well as reflect on the
>         impact of already having mandated transcoding on the audio side.
>
>          
>
>         Chris Cavigioli
>
>         Intel Corp, System Architecture and Planning,
>         Video/Multimedia, Mobile and Communications Group (MCG)
>
>         +1 (415) 254-4545 <tel:%2B1%20%28415%29%20254-4545> mobile
>
>         +1 (408) 653-8348 <tel:%2B1%20%28408%29%20653-8348> desk
>
>         +1 (408) 884-2400 <tel:%2B1%20%28408%29%20884-2400> fax
>
>         This e-mail may contain confidential and privileged material
>         for the sole use of the intended recipient(s).  Any review or
>         distribution by others is strictly prohibited. If you are not
>         the intended recipient, please contact the sender and delete
>         all copies.
>
>          
>
>         *From:*rtcweb [mailto:rtcweb-bounces@ietf.org
>         <mailto:rtcweb-bounces@ietf.org>] *On Behalf Of *Bernard Aboba
>         *Sent:* Sunday, October 19, 2014 2:29 PM
>         *To:* Harald Alvestrand
>         *Cc:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>          
>
>         Harald said: 
>
>          
>
>         "Bernard,
>
>
>         I think this is, to a large degree, codec independent.
>
>         We have demonstrated interoperability on VP8 between Firefox
>         and Chrome, and usage of various mechanisms for congestion
>         control and repair of packet loss being applied in live services.
>
>         So it can't be all bad....."
>
>          
>
>         [BA]  I agree that the lack of interoperability is largely
>         "codec independent":   that is, implementations based on
>         different mechanisms for congestion control, repair, etc. will
>         exhibit interoperability problems, regardless of the codec
>         chosen.   The real test for RTCWEB will be to move beyond
>         "interoperability of implementations sharing source code"  to
>         "interoperability of independent implementations, based on
>         open standards".   
>
>          
>
>         On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand
>         <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>         On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>             "And its one of the issues holding up wider adoption of
>             the technology"
>
>              
>
>             [BA] Specifying an MTI encoder/decoder is not sufficient
>             for interoperability.  It is also necessary to specify the
>             transport in enough detail to allow independent
>             implementations that interoperate well enough to be usable
>             in a wide variety of scenarios, including wireless
>             networks where loss is commonly experienced. 
>
>
>         Bernard,
>
>         I think this is, to a large degree, codec independent.
>
>         We have demonstrated interoperability on VP8 between Firefox
>         and Chrome, and usage of various mechanisms for congestion
>         control and repair of packet loss being applied in live services.
>
>         So it can't be all bad.....
>
>
>
>
>          
>
>         We made the mistake of having an MTI discussion previously
>         with not enough details on that subject, particularly with
>         respect to H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4
>         remain sketchy at best.  
>
>          
>
>         So if we are to have the discussion again, it should occur in
>         the context of detailed specifications and interoperability
>         reports.
>
>          
>
>          
>
>          
>
>          
>
>          
>
>         On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg
>         <jdrosen@jdrosen.net <mailto:jdrosen@jdrosen.net>> wrote:
>
>         I'm in favor of taking another run at this.
>
>          
>
>         The working group has repeatedly said that an MTI codec is
>         something we need to produce. And its one of the issues
>         holding up wider adoption of the technology (not the only one
>         for sure).
>
>          
>
>         And things have changed since the last meeting, a year ago now
>         (November in Vancouver). Cisco's open264 plugin shipped and
>         now just recently is integrated into Firefox. iOS8 shipped
>         with APIs for H264. There are other things worth noting. Will
>         this change the minds of everyone? Surely not. Will it sway
>         enough for us to achieve rough consensus? Maybe. IMHO - worth
>         finding out.
>
>          
>
>         On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD
>         <agouaillard@gmail.com <mailto:agouaillard@gmail.com>> wrote:
>
>         +1 to not having MTI codec discussion unless some progress has
>         been made on VP8 at MPEG. Any news on that? I'm sharing
>         harald's  feeling that there was no change on the members
>         position. 
>
>          
>
>         On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand
>         <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>         On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
>         One thing we could do instead of wasting time on MTI is to
>         actually make progress on Sections 4.2 - 4.4 of
>         draft-IETF-RTCWEB-video, so we could actually interoperate
>         regardless of the codec.
>
>
>         The big argument for an MTI is actually the one stated in
>         -overview: It guards against interoperability failure.
>
>         I would like to have an ecosystem where one can field a box,
>         connect it to everything else, and run well for *some* level
>         of "well" - and I would prefer that ecosystem to be one where
>         it's possible to field the box without making prior
>         arrangements with anyone about licenses.
>
>         This argument hasn't changed one whit since last time we
>         discussed it. And I don't see much movement on the specifics
>         of the proposals either.
>
>         We'll have to interoperate well with the codecs we field. So
>         using some time to discuss draft-ietf-rtcweb-video seems to
>         make sense. (And 4.1 isn't finished either. There's one
>         sentence that needs to be removed.)
>
>         I wouldn't say I'd be happy to not discuss this in Honolulu.
>         But I'd prefer to re-discuss based on the knowledge that some
>         significant players have actually changed their minds.
>
>         At the moment, I don't see signs that any of the poll
>         respondents have said "I have changed my mind".
>
>         Harald
>
>
>
>
>
>         On Oct 16, 2014, at 2:28 PM, Martin Thomson
>         <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>>
>         wrote:
>
>         On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at
>         <mailto:matthew@matthew.at>> wrote:
>         And that's because something substantive has changed, or
>         simply because
>         wasting the WG time on this again is more entertaining than
>         actually
>         finishing a specification that can be independently
>         implemented by all
>         browser vendors? (A specification that we are nowhere near
>         having, as far as
>         I can tell)
>
>         Personally, I've found the reprieve from this fight
>         refreshing.  And
>         it would appear that we've made some real progress as a
>         result.  I'd
>         suggest that if we don't have new information, we continue to
>         spend
>         our time productively.  If we can't find topics to occupy our
>         meeting
>         agenda time, then maybe we can free an agenda slot.
>
>         _______________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/listinfo/rtcweb
>
>         _______________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>         _______________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>          
>
>         -- 
>
>         Alex. Gouaillard, PhD, PhD, MBA
>
>         ------------------------------------------------------------------------------------
>
>         CTO - Temasys Communications, S'pore / Mountain View
>
>         President - CoSMo Software, Cambridge, MA
>
>         ------------------------------------------------------------------------------------
>
>         sg.linkedin.com/agouaillard <http://sg.linkedin.com/agouaillard>
>
>         Â·          
>
>
>         _______________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>          
>
>         -- 
>
>         Jonathan Rosenberg, Ph.D.
>         jdrosen@jdrosen.net <mailto:jdrosen@jdrosen.net>
>         http://www.jdrosen.net
>
>
>         _______________________________________________
>         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
>
>          
>
>         -- 
>
>         Surveillance is pervasive. Go Dark.
>
>
>         _______________________________________________
>         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
>
>


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 10/21/2014 12:29 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-0CrYEXspFU+urB-STkuD=P4jjkBOmfEWVhP_uJuQtFeA@mail.gmail.com"
      type="cite">
      <div dir="ltr">From the subject "Plan for MTI video codec", I
        thought this discussion was about video codecs, not AMR interop.
        <br>
      </div>
    </blockquote>
    <br>
    All discussions branch, but I strongly recommend changing the
    subject line when changing the subject.<br>
    <br>
    The discussion is supposed to be managed by the WG chairs; since
    they have not spoken up on this thread at all, I must assume that
    they do not wish to constrain the discussion at this time.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-0CrYEXspFU+urB-STkuD=P4jjkBOmfEWVhP_uJuQtFeA@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div><br>
        </div>
        <div>An astute observer might take note of the fact that our
          inability to close the door on H.264 is providing an
          opportunity to other royalty-bearing codecs that want to join
          the party.</div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Mon, Oct 20, 2014 at 7:34 AM, DRAGE,
          Keith (Keith) <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:keith.drage@alcatel-lucent.com"
              target="_blank">keith.drage@alcatel-lucent.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div vlink="purple" link="blue" lang="EN-US">
              <div dir="ltr" align="left"><span><font color="#0000ff"
                    face="Arial">The scenario where you might want to
                    transcode the audio codec is where you have wideband
                    audio in both the OPUS and the AMR, and you believe
                    by transcoding you can achieve a wideband quality,
                    rather than falling back to G.711. And for
                    interworking between 3GPP devices that are not using
                    webRTC, the audio will still be AMR or AMR-WB or
                    EVC, not OPUS.</font></span></div>
              <div dir="ltr" align="left"><span></span>Â </div>
              <div dir="ltr" align="left"><span><font color="#0000ff"
                    face="Arial">But audio transcoding is not
                    particularly a concern for me. It is trying where
                    possible to avoid video transcoding.</font></span></div>
              <div dir="ltr" align="left"><span></span>Â </div>
              <div dir="ltr" align="left"><span><font color="#0000ff"
                    face="Arial">Keith</font></span></div>
              <br>
              <blockquote dir="ltr"
                style="PADDING-LEFT:5px;MARGIN-LEFT:5px;BORDER-LEFT:#0000ff
                2px solid;MARGIN-RIGHT:0px">
                <div dir="ltr" align="left" lang="en-us">
                  <hr>
                  <font face="Tahoma"><b>From:</b> rtcweb [mailto:<a
                      moz-do-not-send="true"
                      href="mailto:rtcweb-bounces@ietf.org"
                      target="_blank">rtcweb-bounces@ietf.org</a>]
                    <b>On Behalf Of </b>Andrew Allen<br>
                    <b>Sent:</b> 20 October 2014 14:52<br>
                    <b>To:</b> <a moz-do-not-send="true"
                      href="mailto:chris.cavigioli@intel.com"
                      target="_blank">chris.cavigioli@intel.com</a>; <a
                      moz-do-not-send="true"
                      href="mailto:harald@alvestrand.no" target="_blank">harald@alvestrand.no</a>;
                    <a moz-do-not-send="true"
                      href="mailto:bernard.aboba@gmail.com"
                      target="_blank">bernard.aboba@gmail.com</a>
                    <div>
                      <div class="h5"><br>
                        <b>Cc:</b> <a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                        <b>Subject:</b> Re: [rtcweb] Plan for MTI video
                        codec?<br>
                      </div>
                    </div>
                  </font><br>
                </div>
                <div>
                  <div class="h5">
                    <font
                      style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'"><br>
                      Chris<br>
                      <br>
                      What is the scenario that dictates the need for
                      direct transcoding between OPUS and AMR?
                      <br>
                      <br>
                      All RTCweb clients MUST support OPUS so for Web
                      browsers on mobile LTE devices OPUS can be
                      negotiated with non mobile RTCweb devices end to
                      end.
                      <br>
                      <br>
                      My expectation is that for mobile devices the
                      RTCweb client will also have access to the
                      embedded AMR codec so mobile RTCweb devices can
                      interoperate using AMR with non-RTCweb enabled
                      mobile IMS devices that support AMR.<br>
                      <br>
                      For interoperation between non mobile RTCweb
                      devices and mobile non RTCweb IMS devices the non
                      mobile RTCweb device can use G.711 to the
                      transcoder for transcoding to AMR on the other
                      side. While not ideal this will be no worse than
                      non mobile RTCweb devices interoperating with
                      circuit switched only mobile devices which will be
                      forced to interoperate via the PSTN.<br>
                      <br>
                      As RTCweb becomes universal on mobile devices I
                      expect that support for OPUS will become native
                      and mobile IMS devices will also be able to use
                      OPUS even when operating without the use of the
                      browser (i.e. in non RTCweb mode) for better
                      interoperation with non-mobile RTCweb devices. <br>
                      <br>
                      The lifetime of most mobile devices is 2 years or
                      less so this is likely to be mainly a short term
                      transition issue only.<br>
                      <br>
                      Andrew</font><br>
                    Â <br>
                    <div style="BORDER-RIGHT:medium
                      none;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt
                      solid;PADDING-LEFT:0in;PADDING-BOTTOM:0in;BORDER-LEFT:medium
                      none;PADDING-TOP:3pt;BORDER-BOTTOM:medium none">
                      <font
                        style="FONT-SIZE:10pt;FONT-FAMILY:'Tahoma','sans-serif'"><b>From</b>:
                        Cavigioli, Chris [mailto:<a
                          moz-do-not-send="true"
                          href="mailto:chris.cavigioli@intel.com"
                          target="_blank">chris.cavigioli@intel.com</a>]
                        <br>
                        <b>Sent</b>: Sunday, October 19, 2014 09:20 PM
                        Eastern Standard Time<br>
                        <b>To</b>: Harald Alvestrand &lt;<a
                          moz-do-not-send="true"
                          href="mailto:harald@alvestrand.no"
                          target="_blank">harald@alvestrand.no</a>&gt;;
                        Bernard Aboba &lt;<a moz-do-not-send="true"
                          href="mailto:bernard.aboba@gmail.com"
                          target="_blank">bernard.aboba@gmail.com</a>&gt;
                        <br>
                        <b>Cc</b>: <a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>
                        &lt;<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>&gt;
                        <br>
                        <b>Subject</b>: Re: [rtcweb] Plan for MTI video
                        codec? <br>
                      </font>Â <br>
                    </div>
                    <div>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Harald
                          said:Â 
                        </span></p>
                      <p class="MsoNormal">â€œWe made the mistake of
                        having an MTI discussion previously with not
                        enough details on that subjectâ€¦â€<span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'"></span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">We
                          didnâ€™t have quantitative data earlier for a
                          data-driven decision.Â  Future Web â€“ LTE
                          interop is important.Â  A group of us in GSMA
                          have been looking at WebRTC interop with LTE
                          (where 3GPP / GSMA defined IMS-based
                          voice/video/messaging), specifically looking
                          at user experience impact of latency
                          introduced by added in-network transcoding.Â 
                          GSMA has approved the whitepaper and it is
                          being prepared for public distribution.Â 
                        </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">WebRTC
                          â€“ LTE transcoding is now MANDATORY because of
                          â€œWebRTC Audio Codec and Processing
                          Requirementsâ€.
                          <a moz-do-not-send="true"
                            href="http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05"
                            target="_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>,
                          where MTI audio codecs in WebRTC and in LTE
                          are dissimilar.Â  Thus, REGARDLESS of the video
                          codecs, there is already a significant
                          degradation imposed on Web â€“ LTE links if only
                          MTI codecs are used.Â  The only systems to
                          avoid this are those which implement OPTIONAL
                          audio codecs to be transcoder-free with LTE.Â 
                          The same can apply for video codecs.</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">The
                          GSMA whitepaper refers to:</span></p>
                      <p>
                        <span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'"><span>1.<span
                              style="FONT:7pt 'Times New Roman'">Â Â Â Â 
                            </span></span></span><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Quantitative
                          voice-over-LTE (VoLTE) delay limits were
                          agreed-to by 3GPP SA4 in August 2014.Â  To
                          paraphrase, performance objectives for maximum
                          VoLTE device delays in error and jitter free
                          conditions and AMR speech codec operation, the
                          sum of the UE delays in sending and receiving
                          directions (T<sub>S</sub> + T<sub>R</sub>)
                          should be â‰¤ 150ms. If this performance
                          objective cannot be met, the sum of the UE
                          delays in sending and receiving directions (T<sub>S</sub>
                          + T<sub>R</sub>) shall in any case be â‰¤ 190ms.</span></p>
                      <p>
                        <span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'"><span>2.<span
                              style="FONT:7pt 'Times New Roman'">Â Â Â Â 
                            </span></span></span><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">OPUS
                          â€“ AMR transcoding adds an additional 211.5 ms
                          implementation-agnostic (T<sub>S</sub> + T<sub>R</sub>)
                          delay, including 40 ms for jitter buffer and
                          40 ms for discontinuous reception (DRX).</span></p>
                      <p>
                        <span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'"><span>3.<span
                              style="FONT:7pt 'Times New Roman'">Â Â Â Â 
                            </span></span></span><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">To
                          put these numbers in perspective, it is in
                          general desirable to minimize UE delays to
                          ensure low enough end-to-end delays and hence
                          a good conversational experience.Â  Increasing
                          delay causes people to double-talk making
                          conversations increasingly frustrating.Â 
                          Guidance to stay below 400 ms maximum one-way
                          delay is found in ITU-T Recommendation G.114
                          and the computational E-model is in ITU-T
                          Recommendation G.107.</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">This
                          is a complex topic with many network factors
                          in play.Â  LTE operators in 3GPP seek UE delays
                          (T<sub>S</sub> + T<sub>R</sub>) around 150
                          ms.Â  Adding OPUS â€“ AMR transcoding in the
                          network imposes an additional 211.5 ms on top
                          of that, just for audio.Â  Optionally using AMR
                          in WebRTC, those 211.5 ms can be eliminated,
                          delivering a superior end-user experience.Â 
                        </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">CONCLUSION:
                          Â regardless of MTI codec choices in WebRTC, to
                          provide a good WebRTC â€“ LTE interop
                          experience, emerging WebRTC systems must
                          accommodate MTI codecs already deployed in the
                          LTE domain and in mobile operating systems,
                          namely AMR/AMR-WB and H.264.</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">POSITION:Â 
                          To be clear, several Intel SoCs launched at
                          MWC this year for smartphones and tablets
                          support both VP8 and H.264 hardware encode and
                          decode â€“ so Intel is codec-neutral in this MTI
                          debate.Â  Additionally, Intel has
                          high-performance solutions for transcoding.Â 
                          However, we wish to inform the industry, with
                          quantitative data, about impact to end-user
                          experience of mandating such transcoding so
                          IETF can make a data-driven decision on video
                          codecs as well as reflect on the impact of
                          already having mandated transcoding on the
                          audio side.</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><span
                          style="COLOR:#1f497d;FONT-FAMILY:'Lucida
                          Console'">Chris Cavigioli</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'">Intel
                          Corp,
                        </span><span
                          style="FONT-SIZE:8pt;COLOR:#0070c0;FONT-FAMILY:'Verdana','sans-serif'">System
                          Architecture and Planning</span><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'">,
                          Video/Multimedia, Mobile and Communications
                          Group (MCG)</span><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"></span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"><a
                            moz-do-not-send="true"
                            href="tel:%2B1%20%28415%29%20254-4545"
                            value="+14152544545" target="_blank">+1
                            (415) 254-4545</a> mobile</span><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"></span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"><a
                            moz-do-not-send="true"
                            href="tel:%2B1%20%28408%29%20653-8348"
                            value="+14086538348" target="_blank">+1
                            (408) 653-8348</a> desk</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"><a
                            moz-do-not-send="true"
                            href="tel:%2B1%20%28408%29%20884-2400"
                            value="+14088842400" target="_blank">+1
                            (408) 884-2400</a> fax</span><span
                          style="FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:'Verdana','sans-serif'"></span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:7.5pt;COLOR:gray;FONT-FAMILY:'Verdana','sans-serif'">This
                          e-mail may contain confidential and privileged
                          material for the sole use of the intended
                          recipient(s).Â  Any review or distribution by
                          others is strictly prohibited. If you are not
                          the intended recipient, please contact the
                          sender and delete all copies.</span></p>
                      <p class="MsoNormal"><span
                          style="FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:'Arial','sans-serif'">Â </span></p>
                      <p class="MsoNormal"><b><span
                            style="FONT-SIZE:10pt;FONT-FAMILY:'Tahoma','sans-serif'">From:</span></b><span
style="FONT-SIZE:10pt;FONT-FAMILY:'Tahoma','sans-serif'"> rtcweb
                          [mailto:<a moz-do-not-send="true"
                            href="mailto:rtcweb-bounces@ietf.org"
                            target="_blank">rtcweb-bounces@ietf.org</a>]
                          <b>On Behalf Of </b>Bernard Aboba<br>
                          <b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
                          <b>To:</b> Harald Alvestrand<br>
                          <b>Cc:</b> <a moz-do-not-send="true"
                            href="mailto:rtcweb@ietf.org"
                            target="_blank">rtcweb@ietf.org</a><br>
                          <b>Subject:</b> Re: [rtcweb] Plan for MTI
                          video codec?</span></p>
                      <p class="MsoNormal">Â </p>
                      <div>
                        <p class="MsoNormal">Harald said:Â </p>
                        <div>
                          <p class="MsoNormal">Â </p>
                        </div>
                        <div>
                          <p class="MsoNormal">"<span
                              style="FONT-SIZE:10pt;FONT-FAMILY:'Arial','sans-serif'">Bernard,</span></p>
                        </div>
                        <p class="MsoNormal" style="MARGIN-BOTTOM:12pt"><span
style="FONT-SIZE:10pt;FONT-FAMILY:'Arial','sans-serif'"><br>
                            I think this is, to a large degree, codec
                            independent.<br>
                            <br>
                            We have demonstrated interoperability on VP8
                            between Firefox and Chrome, and usage of
                            various mechanisms for congestion control
                            and repair of packet loss being applied in
                            live services.</span></p>
                        <div>
                          <p class="MsoNormal"><span
                              style="FONT-SIZE:10pt;FONT-FAMILY:'Arial','sans-serif'">So
                              it can't be all bad.....</span>"</p>
                        </div>
                        <div>
                          <p class="MsoNormal">Â </p>
                        </div>
                        <div>
                          <p class="MsoNormal">[BA] Â I agree that the
                            lack of interoperability is largely "codec
                            independent": Â  that is, implementations
                            based on different mechanisms for congestion
                            control, repair, etc. will exhibit
                            interoperability problems, regardless of the
                            codec chosen. Â  The real test for RTCWEB
                            will be to move beyond "interoperability of
                            implementations sharing source code" Â to
                            "interoperability of independent
                            implementations, based on open standards".
                            Â Â </p>
                        </div>
                      </div>
                      <div>
                        <p class="MsoNormal">Â </p>
                        <div>
                          <p class="MsoNormal">On Sun, Oct 19, 2014 at
                            1:38 PM, Harald Alvestrand &lt;<a
                              moz-do-not-send="true"
                              href="mailto:harald@alvestrand.no"
                              target="_blank">harald@alvestrand.no</a>&gt;
                            wrote:</p>
                          <div>
                            <div>
                              <p class="MsoNormal">On 10/19/2014 05:43
                                PM, Bernard Aboba wrote:</p>
                            </div>
                            <blockquote
                              style="MARGIN-TOP:5pt;MARGIN-BOTTOM:5pt">
                              <div>
                                <p class="MsoNormal">"And its one of the
                                  issues holding up wider adoption of
                                  the technology"
                                </p>
                                <div>
                                  <p class="MsoNormal">Â </p>
                                </div>
                                <div>
                                  <p class="MsoNormal">[BA] Specifying
                                    an MTI encoder/decoder is not
                                    sufficient for interoperability.Â  It
                                    is also necessary to specify the
                                    transport in enough detail to allow
                                    independent implementations that
                                    interoperate well enough to be
                                    usable in a wide variety of
                                    scenarios, including wireless
                                    networks where loss is commonly
                                    experienced.Â  </p>
                                </div>
                              </div>
                            </blockquote>
                            <p class="MsoNormal"><br>
                              Bernard,<br>
                              <br>
                              I think this is, to a large degree, codec
                              independent.<br>
                              <br>
                              We have demonstrated interoperability on
                              VP8 between Firefox and Chrome, and usage
                              of various mechanisms for congestion
                              control and repair of packet loss being
                              applied in live services.<br>
                              <br>
                              So it can't be all bad.....</p>
                            <div>
                              <div>
                                <p class="MsoNormal"><br>
                                  <br>
                                  <br>
                                </p>
                                <div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">We made the
                                      mistake of having an MTI
                                      discussion previously with not
                                      enough details on that subject,
                                      particularly with respect to
                                      H.264. draft-ietf-rtcweb-video
                                      sections 4.2 - 4.4 remain sketchy
                                      at best. Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">So if we are to
                                      have the discussion again, it
                                      should occur in the context of
                                      detailed specifications and
                                      interoperability reports.</p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                  <div>
                                    <p class="MsoNormal">Â </p>
                                  </div>
                                </div>
                                <div>
                                  <p class="MsoNormal">Â </p>
                                  <div>
                                    <p class="MsoNormal">On Sun, Oct 19,
                                      2014 at 8:14 AM, Jonathan
                                      Rosenberg &lt;<a
                                        moz-do-not-send="true"
                                        href="mailto:jdrosen@jdrosen.net"
                                        target="_blank">jdrosen@jdrosen.net</a>&gt;
                                      wrote:</p>
                                    <div>
                                      <p class="MsoNormal">I'm in favor
                                        of taking another run at this. </p>
                                      <div>
                                        <p class="MsoNormal">Â </p>
                                      </div>
                                      <div>
                                        <p class="MsoNormal">The working
                                          group has repeatedly said that
                                          an MTI codec is something we
                                          need to produce. And its one
                                          of the issues holding up wider
                                          adoption of the technology
                                          (not the only one for sure).</p>
                                      </div>
                                      <div>
                                        <p class="MsoNormal">Â </p>
                                      </div>
                                      <div>
                                        <p class="MsoNormal">And things
                                          have changed since the last
                                          meeting, a year ago now
                                          (November in Vancouver).
                                          Cisco's open264 plugin shipped
                                          and now just recently is
                                          integrated into Firefox. iOS8
                                          shipped with APIs for H264.
                                          There are other things worth
                                          noting. Will this change the
                                          minds of everyone? Surely not.
                                          Will it sway enough for us to
                                          achieve rough consensus?
                                          Maybe. IMHO - worth finding
                                          out.</p>
                                      </div>
                                    </div>
                                    <div>
                                      <div>
                                        <div>
                                          <p class="MsoNormal">Â </p>
                                          <div>
                                            <p class="MsoNormal">On Sat,
                                              Oct 18, 2014 at 5:08 PM,
                                              Alexandre GOUAILLARD &lt;<a
                                                moz-do-not-send="true"
                                                href="mailto:agouaillard@gmail.com"
                                                target="_blank">agouaillard@gmail.com</a>&gt;
                                              wrote:</p>
                                            <div>
                                              <p class="MsoNormal">+1 to
                                                not having MTI codec
                                                discussion unless some
                                                progress has been made
                                                on VP8 at MPEG.Â Any news
                                                on that? I'm sharing
                                                harald's Â feeling that
                                                there was no change on
                                                the members position.Â </p>
                                            </div>
                                            <div>
                                              <div>
                                                <div>
                                                  <p class="MsoNormal">Â </p>
                                                  <div>
                                                    <p class="MsoNormal">On
                                                      Fri, Oct 17, 2014
                                                      at 9:22 PM, Harald
                                                      Alvestrand &lt;<a
moz-do-not-send="true" href="mailto:harald@alvestrand.no"
                                                        target="_blank">harald@alvestrand.no</a>&gt;
                                                      wrote:</p>
                                                    <p class="MsoNormal">On
                                                      10/17/2014 12:02
                                                      AM, Bernard Aboba
                                                      wrote:</p>
                                                    <p class="MsoNormal">One
                                                      thing we could do
                                                      instead of wasting
                                                      time on MTI is to
                                                      actually make
                                                      progress on
                                                      Sections 4.2 - 4.4
                                                      of
                                                      draft-IETF-RTCWEB-video,
                                                      so we could
                                                      actually
                                                      interoperate
                                                      regardless of the
                                                      codec.</p>
                                                    <p class="MsoNormal"><br>
                                                      The big argument
                                                      for an MTI is
                                                      actually the one
                                                      stated in
                                                      -overview: It
                                                      guards against
                                                      interoperability
                                                      failure.<br>
                                                      <br>
                                                      I would like to
                                                      have an ecosystem
                                                      where one can
                                                      field a box,
                                                      connect it to
                                                      everything else,
                                                      and run well for
                                                      *some* level of
                                                      "well" - and I
                                                      would prefer that
                                                      ecosystem to be
                                                      one where it's
                                                      possible to field
                                                      the box without
                                                      making prior
                                                      arrangements with
                                                      anyone about
                                                      licenses.<br>
                                                      <br>
                                                      This argument
                                                      hasn't changed one
                                                      whit since last
                                                      time we discussed
                                                      it. And I don't
                                                      see much movement
                                                      on the specifics
                                                      of the proposals
                                                      either.<br>
                                                      <br>
                                                      We'll have to
                                                      interoperate well
                                                      with the codecs we
                                                      field. So using
                                                      some time to
                                                      discuss
                                                      draft-ietf-rtcweb-video
                                                      seems to make
                                                      sense. (And 4.1
                                                      isn't finished
                                                      either. There's
                                                      one sentence that
                                                      needs to be
                                                      removed.)<br>
                                                      <br>
                                                      I wouldn't say I'd
                                                      be happy to not
                                                      discuss this in
                                                      Honolulu. But I'd
                                                      prefer to
                                                      re-discuss based
                                                      on the knowledge
                                                      that some
                                                      significant
                                                      players have
                                                      actually changed
                                                      their minds.<br>
                                                      <br>
                                                      At the moment, I
                                                      don't see signs
                                                      that any of the
                                                      poll respondents
                                                      have said "I have
                                                      changed my mind".<span
style="COLOR:#888888"><br>
                                                        <br>
                                                        Harald</span> </p>
                                                    <div>
                                                      <div>
                                                        <p
                                                          class="MsoNormal"
style="MARGIN-BOTTOM:12pt"><br>
                                                          <br>
                                                        </p>
                                                        <p
                                                          class="MsoNormal"
style="MARGIN-BOTTOM:12pt"><br>
                                                          <br>
                                                        </p>
                                                        <p
                                                          class="MsoNormal"
style="MARGIN-BOTTOM:12pt">On Oct 16, 2014, at 2:28 PM, Martin Thomson
                                                          &lt;<a
                                                          moz-do-not-send="true"
href="mailto:martin.thomson@gmail.com" target="_blank">martin.thomson@gmail.com</a>&gt;
                                                          wrote:</p>
                                                        <p
                                                          class="MsoNormal">On
                                                          16 October
                                                          2014 14:17,
                                                          Matthew
                                                          Kaufman &lt;<a
moz-do-not-send="true" href="mailto:matthew@matthew.at" target="_blank">matthew@matthew.at</a>&gt;
                                                          wrote:<br>
                                                          And that's
                                                          because
                                                          something
                                                          substantive
                                                          has changed,
                                                          or simply
                                                          because<br>
                                                          wasting the WG
                                                          time on this
                                                          again is more
                                                          entertaining
                                                          than actually<br>
                                                          finishing a
                                                          specification
                                                          that can be
                                                          independently
                                                          implemented by
                                                          all<br>
                                                          browser
                                                          vendors? (A
                                                          specification
                                                          that we are
                                                          nowhere near
                                                          having, as far
                                                          as<br>
                                                          I can tell)</p>
                                                        <p
                                                          class="MsoNormal">Personally,
                                                          I've found the
                                                          reprieve from
                                                          this fight
                                                          refreshing.Â 
                                                          And<br>
                                                          it would
                                                          appear that
                                                          we've made
                                                          some real
                                                          progress as a
                                                          result.Â  I'd<br>
                                                          suggest that
                                                          if we don't
                                                          have new
                                                          information,
                                                          we continue to
                                                          spend<br>
                                                          our time
                                                          productively.Â 
                                                          If we can't
                                                          find topics to
                                                          occupy our
                                                          meeting<br>
                                                          agenda time,
                                                          then maybe we
                                                          can free an
                                                          agenda slot.<br>
                                                          <br>
_______________________________________________<br>
                                                          rtcweb mailing
                                                          list<br>
                                                          <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                          <a
                                                          moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                                                        <p
                                                          class="MsoNormal">_______________________________________________<br>
                                                          rtcweb mailing
                                                          list<br>
                                                          <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                          <a
                                                          moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                                                        <p
                                                          class="MsoNormal"><br>
_______________________________________________<br>
                                                          rtcweb mailing
                                                          list<br>
                                                          <a
                                                          moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                          <a
                                                          moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                                                      </div>
                                                    </div>
                                                  </div>
                                                  <p class="MsoNormal"><br>
                                                    <br clear="all">
                                                  </p>
                                                  <div>
                                                    <p class="MsoNormal">Â </p>
                                                  </div>
                                                </div>
                                              </div>
                                              <p class="MsoNormal"><span
                                                  style="COLOR:#888888">--
                                                </span></p>
                                              <div>
                                                <p class="MsoNormal"><span
style="COLOR:#888888">Alex. Gouaillard, PhD, PhD, MBA
                                                  </span></p>
                                                <div>
                                                  <p class="MsoNormal"><span
style="COLOR:#888888">------------------------------------------------------------------------------------</span></p>
                                                </div>
                                                <div>
                                                  <p class="MsoNormal"><span
style="COLOR:#888888">CTO - Temasys Communications, S'pore / Mountain
                                                      View</span></p>
                                                </div>
                                                <div>
                                                  <p class="MsoNormal"><span
style="COLOR:#888888">President - CoSMo Software, Cambridge, MA</span></p>
                                                </div>
                                                <div>
                                                  <p class="MsoNormal"><span
style="COLOR:#888888">------------------------------------------------------------------------------------</span></p>
                                                </div>
                                                <div>
                                                  <p class="MsoNormal"><span
style="COLOR:#888888"><a moz-do-not-send="true"
                                                        href="http://sg.linkedin.com/agouaillard"
                                                        target="_blank">sg.linkedin.com/agouaillard</a></span></p>
                                                </div>
                                                <div>
                                                  <p class="MsoNormal"
                                                    style="MARGIN-LEFT:0in;VERTICAL-ALIGN:baseline;LINE-HEIGHT:14.4pt">
                                                    <span
                                                      style="FONT-SIZE:10pt;COLOR:#333333;FONT-FAMILY:Symbol"><span>Â·<span
                                                          style="FONT:7pt
                                                          'Times New
                                                          Roman'">Â Â Â Â Â Â Â Â 
                                                        </span></span></span><span
style="FONT-SIZE:8.5pt;COLOR:#333333;FONT-FAMILY:inherit">Â </span></p>
                                                </div>
                                              </div>
                                            </div>
                                            <p class="MsoNormal"
                                              style="MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
                                              rtcweb mailing list<br>
                                              <a moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                              <a moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                                          </div>
                                          <p class="MsoNormal"><br>
                                            <br clear="all">
                                          </p>
                                          <div>
                                            <p class="MsoNormal">Â </p>
                                          </div>
                                          <p class="MsoNormal">-- </p>
                                        </div>
                                      </div>
                                      <div>
                                        <p class="MsoNormal">Jonathan
                                          Rosenberg, Ph.D.<br>
                                          <a moz-do-not-send="true"
                                            href="mailto:jdrosen@jdrosen.net"
                                            target="_blank">jdrosen@jdrosen.net</a><br>
                                          <a moz-do-not-send="true"
                                            href="http://www.jdrosen.net"
                                            target="_blank">http://www.jdrosen.net</a></p>
                                      </div>
                                    </div>
                                    <p class="MsoNormal"
                                      style="MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
                                      rtcweb mailing list<br>
                                      <a moz-do-not-send="true"
                                        href="mailto:rtcweb@ietf.org"
                                        target="_blank">rtcweb@ietf.org</a><br>
                                      <a moz-do-not-send="true"
                                        href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                        target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                                  </div>
                                  <p class="MsoNormal">Â </p>
                                </div>
                                <p class="MsoNormal"
                                  style="MARGIN-BOTTOM:12pt">Â </p>
                                <pre>_______________________________________________</pre>
                                <pre>rtcweb mailing list</pre>
                                <pre><a moz-do-not-send="true" href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a></pre>
                                <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></pre>
                                <p class="MsoNormal"
                                  style="MARGIN-BOTTOM:12pt">Â </p>
                              </div>
                            </div>
                            <pre><span style="COLOR:#888888">-- </span></pre>
                            <pre><span style="COLOR:#888888">Surveillance is pervasive. Go Dark.</span></pre>
                          </div>
                          <p class="MsoNormal"
                            style="MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
                            rtcweb mailing list<br>
                            <a moz-do-not-send="true"
                              href="mailto:rtcweb@ietf.org"
                              target="_blank">rtcweb@ietf.org</a><br>
                            <a moz-do-not-send="true"
                              href="https://www.ietf.org/mailman/listinfo/rtcweb"
                              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></p>
                        </div>
                        <p class="MsoNormal">Â </p>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
            </div>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
            <br>
          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

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On 21 Oct 2014, at 00:50, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2014-10-21 0:46 GMT+02:00 Roman Shpount <roman@telurix.com>:
>> Unless H.264 became royalty free or royalty free profile for H.264 is
>> developed (and I am talking about ability to include any =
implementation of
>> this H.264 profile into any project without paying royalties, not =
linking to
>> something the end user must  download), H.264 would not be acceptable =
to me
>> as an MTI. All the other news updates regarding H.264 are irrelevant. =
I
>> would assume most open source developer or smaller software =
developers would
>> share my opinion.
>=20
> +1

+pi

/O=


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Date: Tue, 21 Oct 2014 07:35:39 +0000
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References: <20141010004836.12666.88765.idtracker@ietfa.amsl.com> <03a3bbbc282b4e6d88d587931b46b5f8@CY1PR0501MB1579.namprd05.prod.outlook.com> <57B40110-2F21-4893-B77C-54F46D18567F@lurchi.franken.de> <1DE35922-E890-40C2-87E9-C8C12235920E@phonefromhere.com> <E53B9B7E-37F0-495C-B1BA-DE0A48AC6D73@lurchi.franken.de> <abdfd3ef58dd40028e8d5e2248b5a995@CY1PR0501MB1579.namprd05.prod.outlook.com> <543A5570.7060208@alvestrand.no> <B53499C4-6B51-4E8E-87C2-C8E5C10DBC34@lurchi.franken.de> <543A9E11.2030605@alvestrand.no> <F5C54F5E-C301-4EC7-9536-E43EA3A16863@lurchi.franken.de> <543ABE69.8030104@alvestrand.no> <99078EA5-5FE7-431F-9C70-EEA882F4C3C6@lurchi.franken.de> <E1FE4C082A89A246A11D7F32A95A17828E5DA2AC@US70UWXCHMBA02.zam.alcatel-lucent.com> <71e2b21516cb496eb4d1ad2b26e53a29@CY1PR0501MB1579.namprd05.prod.outlook.com> <543CD1CD.3000001@alvestrand.no> <5440E38F.9070809@jesup.org> <c4ff2fdcd48f4388b67f3e1ffed8edfe@CY1PR0501MB1579.namprd05.prod.outlook.com> <5442B41D.6040307@jesup.org>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> Per above, if it's simply asking the application for an offer, then
> forwarding it, there's no issue: the offer is a true offer, you never
> have to "convert" an SDP offer into an answer or vice versa, and the
> offer will be actpass and roles will be consistent (assuming the
> middlebox isn't trying to intercept/sniff the decrypted DataChannel data)=
.

You are right, that case will work.  I was thinking that the middlebox coul=
d treat the first leg as more independent and not having to wait for the an=
swer on the second leg for faster connection establishment.  =20


> So, if the middle box uses sip-style "empty offers" (ask the app to make
> an offer), then I think there are no issues with DTLS role.  If it tries
> to avoid the 3-way handshake style, or work with apps that don't
> understand it then there'd be an issue (but in webrtc signaling isn't
> specified, so the middlebox MUST be designed to be compatible with the
> signaling used by the apps).
>=20
> Also, if the middlebox acts as an "answerer-in-the-middle", DTLS roles
> should work:
> A->middlebox offer, B->middlebox offer, middlebox->A answer,
> middlebox->B answer (where the answers are derived from the offer on the
> other side).  This again requires that the two Offers be safely
> transformable by the middlebox into answers without a renegotiation. The
> middlebox would select the roles in the answers to be complimentary.

The "empty offer" approach makes more sense in the SIP world because endpoi=
nts are rather heterogeneous in their codec support.  It thus allows the mi=
ddlebox more control over codec selection by being the answerer.  One of th=
e big advantages of WebRTC and its mandatory-to-implement codec is that it =
safe to simply offer Opus and be done with it.  Having to resort to the "an=
swerer in the middle" approach just so the DTLS roles can be picked is actu=
ally another good example of how the DTLS role abstraction leak has uninten=
ded consequences and is bad. =20


> I disagree that role won't work if the middlebox is just forwarding
> offers and answers - that's what every WebRTC signaling server is doing
> today.   However, as I think you implied, you're also terminating DTLS
> but forwarding SCTP (and media).  This a) implies packet-relay; b)
> implicit decryption of all media at the server (NOTE this is a
> security/privacy/MITM issue).

Yes, that's the intended use-case: The middlebox is a trusted man-in-the-mi=
ddle.  In order to record and/or perform analytics on the media flowing thr=
ough it, it obviously has to decrypt it.  That can only be done by terminat=
ing DTLS (and decrypting/re-encrypting SRTP.)   =20


> Also, assuming some bits of my analysis are incorrect, or there are edge
> cases we care about we don't handle (the non-SIP-like start-in-middle
> case), then (and only then) as a working group, we need to decide if
> this is a usecase we support.  Obviously B2BUA's (which this
> more-or-less is) are always possible, at varying levels of ease.  The
> question (partly) is should the protocol be designed to ease the use of
> a B2BUA with datachannels *assuming* it doesn't work per above.

Without inband ID negotiation or an SDP parameter, an "offerer in the middl=
e" middlebox or B2BUA won't be possible.  I think having the flexibility to=
 use either/any scenario is highly desirable.=20


> >> An SDP parameter would
> >> allow a middlebox to specify to the endpoints compatible
> (complementary)
> >> values; however from a DataChannels perspective, this is almost the sa=
me
> as
> >> external negotiation if the middlebox and applications are 'matched'.
> > If in-band negotiation and collision resolution is not an option, addin=
g an
> SDP attribute would be my second choice.  If it's not present, use
> offerer/answerer role.
>=20
> If we *cannot* safely use DTLS role for middlebox cases, and care, then
> we should allow SDP to override the DTLS role (not use the SDP offer
> state as the base).  If an offerer doesn't specify even/odd in the SDP,
> but the answer does specify, then the offerer must use the SDP of the
> answerer to select even/odd (i.e. SDP overrides role).

Having an SDP parameter that overrides the DTLS role instead of the offer/a=
nswer role is fine by me.  As long as there is a way to override ID selecti=
on instead of using the DTLS role.  I nevertheless feel an in-band solution=
 that doesn't depend on signaling or lower-level transport state would be b=
etter as it minimizes dependencies and localizes the complexity in DCEP.

--Felix


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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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On 21 Oct 2014, at 08:35, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>=20
>=20
>> I disagree that role won't work if the middlebox is just forwarding
>> offers and answers - that's what every WebRTC signaling server is =
doing
>> today.   However, as I think you implied, you're also terminating =
DTLS
>> but forwarding SCTP (and media).  This a) implies packet-relay; b)
>> implicit decryption of all media at the server (NOTE this is a
>> security/privacy/MITM issue).
>=20
> Yes, that's the intended use-case: The middlebox is a trusted =
man-in-the-middle.  In order to record and/or perform analytics on the =
media flowing through it, it obviously has to decrypt it.  That can only =
be done by terminating DTLS (and decrypting/re-encrypting SRTP.)   =20

If the middle box is trusted, only interested in media and known about =
by the application, then the application should create a separate=20
direct data channel only DTLS connection between the endpoints that =
bypasses the middle box, with a small amount of extra application
effort we remove the need for even more random extra SDP lines. Remember =
these are programmable endpoints - not dumb SIP devices with fixed =
behaviours.

Unless the middle box is performing =91analytics=92 on the DC data too, =
but that isn=92t the use-case you mentioned.

Tim.


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To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
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--_000_949EF20990823C4C85C18D59AA11AD8B263123FR712WXCHMBA11zeu_
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With respect to audio codecs I was replying to a question from the previous=
 message in the thread. But...

The MTI discussion does not close the door on any codec, as any codec can b=
e supported, just not mandatory to implement.

Your argument to eliminate H.264 only applies if you can either eliminate i=
nterworking with the rest of the world or you can persuade all those legacy=
 devices to remove H.264 and use something else, which in my view is totall=
y impractical.

MTI might persuade usage of that codec, but it will only do that if the MTI=
 codec actually supports end to end communication of the required video cap=
abilities. Neither VP8 or H.264 fit all those requirements.

Keith

________________________________
From: Justin Uberti [mailto:juberti@google.com]
Sent: 20 October 2014 23:29
To: DRAGE, Keith (Keith)
Cc: Andrew Allen; chris.cavigioli@intel.com; harald@alvestrand.no; bernard.=
aboba@gmail.com; rtcweb@ietf.org
Subject: Re: [rtcweb] Plan for MTI video codec?

>From the subject "Plan for MTI video codec", I thought this discussion was =
about video codecs, not AMR interop.

An astute observer might take note of the fact that our inability to close =
the door on H.264 is providing an opportunity to other royalty-bearing code=
cs that want to join the party.

On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (Keith) <keith.drage@alcatel-=
lucent.com<mailto:keith.drage@alcatel-lucent.com>> wrote:
The scenario where you might want to transcode the audio codec is where you=
 have wideband audio in both the OPUS and the AMR, and you believe by trans=
coding you can achieve a wideband quality, rather than falling back to G.71=
1. And for interworking between 3GPP devices that are not using webRTC, the=
 audio will still be AMR or AMR-WB or EVC, not OPUS.

But audio transcoding is not particularly a concern for me. It is trying wh=
ere possible to avoid video transcoding.

Keith

________________________________
From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org=
>] On Behalf Of Andrew Allen
Sent: 20 October 2014 14:52
To: chris.cavigioli@intel.com<mailto:chris.cavigioli@intel.com>; harald@alv=
estrand.no<mailto:harald@alvestrand.no>; bernard.aboba@gmail.com<mailto:ber=
nard.aboba@gmail.com>

Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?


Chris

What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?

All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.

My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.

For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices interoperating with circuit switched only=
 mobile devices which will be forced to interoperate via the PSTN.

As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with non-mobile RTCweb devices.

The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.

Andrew

From: Cavigioli, Chris [mailto:chris.cavigioli@intel.com<mailto:chris.cavig=
ioli@intel.com>]
Sent: Sunday, October 19, 2014 09:20 PM Eastern Standard Time
To: Harald Alvestrand <harald@alvestrand.no<mailto:harald@alvestrand.no>>; =
Bernard Aboba <bernard.aboba@gmail.com<mailto:bernard.aboba@gmail.com>>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org> <rtcweb@ietf.org<mailto:rtcweb@=
ietf.org>>
Subject: Re: [rtcweb] Plan for MTI video codec?

Harald said:
"We made the mistake of having an MTI discussion previously with not enough=
 details on that subject..."

We didn't have quantitative data earlier for a data-driven decision.  Futur=
e Web - LTE interop is important.  A group of us in GSMA have been looking =
at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based voice/video=
/messaging), specifically looking at user experience impact of latency intr=
oduced by added in-network transcoding.  GSMA has approved the whitepaper a=
nd it is being prepared for public distribution.

WebRTC - LTE transcoding is now MANDATORY because of "WebRTC Audio Codec an=
d Processing Requirements". http://tools.ietf.org/html/draft-ietf-rtcweb-au=
dio-05, where MTI audio codecs in WebRTC and in LTE are dissimilar.  Thus, =
REGARDLESS of the video codecs, there is already a significant degradation =
imposed on Web - LTE links if only MTI codecs are used.  The only systems t=
o avoid this are those which implement OPTIONAL audio codecs to be transcod=
er-free with LTE.  The same can apply for video codecs.

The GSMA whitepaper refers to:

1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by 3=
GPP SA4 in August 2014.  To paraphrase, performance objectives for maximum =
VoLTE device delays in error and jitter free conditions and AMR speech code=
c operation, the sum of the UE delays in sending and receiving directions (=
TS + TR) should be =98 150ms. If this performance objective cannot be met, =
the sum of the UE delays in sending and receiving directions (TS + TR) shal=
l in any case be =98 190ms.

2.     OPUS - AMR transcoding adds an additional 211.5 ms implementation-ag=
nostic (TS + TR) delay, including 40 ms for jitter buffer and 40 ms for dis=
continuous reception (DRX).

3.     To put these numbers in perspective, it is in general desirable to m=
inimize UE delays to ensure low enough end-to-end delays and hence a good c=
onversational experience.  Increasing delay causes people to double-talk ma=
king conversations increasingly frustrating.  Guidance to stay below 400 ms=
 maximum one-way delay is found in ITU-T Recommendation G.114 and the compu=
tational E-model is in ITU-T Recommendation G.107.

This is a complex topic with many network factors in play.  LTE operators i=
n 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS - AMR transcodi=
ng in the network imposes an additional 211.5 ms on top of that, just for a=
udio.  Optionally using AMR in WebRTC, those 211.5 ms can be eliminated, de=
livering a superior end-user experience.

CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good W=
ebRTC - LTE interop experience, emerging WebRTC systems must accommodate MT=
I codecs already deployed in the LTE domain and in mobile operating systems=
, namely AMR/AMR-WB and H.264.

POSITION:  To be clear, several Intel SoCs launched at MWC this year for sm=
artphones and tablets support both VP8 and H.264 hardware encode and decode=
 - so Intel is codec-neutral in this MTI debate.  Additionally, Intel has h=
igh-performance solutions for transcoding.  However, we wish to inform the =
industry, with quantitative data, about impact to end-user experience of ma=
ndating such transcoding so IETF can make a data-driven decision on video c=
odecs as well as reflect on the impact of already having mandated transcodi=
ng on the audio side.

Chris Cavigioli
Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile and =
Communications Group (MCG)
+1 (415) 254-4545<tel:%2B1%20%28415%29%20254-4545> mobile
+1 (408) 653-8348<tel:%2B1%20%28408%29%20653-8348> desk
+1 (408) 884-2400<tel:%2B1%20%28408%29%20884-2400> fax
This e-mail may contain confidential and privileged material for the sole u=
se of the intended recipient(s).  Any review or distribution by others is s=
trictly prohibited. If you are not the intended recipient, please contact t=
he sender and delete all copies.

From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org=
>] On Behalf Of Bernard Aboba
Sent: Sunday, October 19, 2014 2:29 PM
To: Harald Alvestrand
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?

Harald said:

"Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.
So it can't be all bad....."

[BA]  I agree that the lack of interoperability is largely "codec independe=
nt":   that is, implementations based on different mechanisms for congestio=
n control, repair, etc. will exhibit interoperability problems, regardless =
of the codec chosen.   The real test for RTCWEB will be to move beyond "int=
eroperability of implementations sharing source code"  to "interoperability=
 of independent implementations, based on open standards".

On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
On 10/19/2014 05:43 PM, Bernard Aboba wrote:
"And its one of the issues holding up wider adoption of the technology"

[BA] Specifying an MTI encoder/decoder is not sufficient for interoperabili=
ty.  It is also necessary to specify the transport in enough detail to allo=
w independent implementations that interoperate well enough to be usable in=
 a wide variety of scenarios, including wireless networks where loss is com=
monly experienced.

Bernard,

I think this is, to a large degree, codec independent.

We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.

So it can't be all bad.....




We made the mistake of having an MTI discussion previously with not enough =
details on that subject, particularly with respect to H.264. draft-ietf-rtc=
web-video sections 4.2 - 4.4 remain sketchy at best.

So if we are to have the discussion again, it should occur in the context o=
f detailed specifications and interoperability reports.





On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net<ma=
ilto:jdrosen@jdrosen.net>> wrote:
I'm in favor of taking another run at this.

The working group has repeatedly said that an MTI codec is something we nee=
d to produce. And its one of the issues holding up wider adoption of the te=
chnology (not the only one for sure).

And things have changed since the last meeting, a year ago now (November in=
 Vancouver). Cisco's open264 plugin shipped and now just recently is integr=
ated into Firefox. iOS8 shipped with APIs for H264. There are other things =
worth noting. Will this change the minds of everyone? Surely not. Will it s=
way enough for us to achieve rough consensus? Maybe. IMHO - worth finding o=
ut.

On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <agouaillard@gmail.co=
m<mailto:agouaillard@gmail.com>> wrote:
+1 to not having MTI codec discussion unless some progress has been made on=
 VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that there wa=
s no change on the members position.

On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
On 10/17/2014 12:02 AM, Bernard Aboba wrote:
One thing we could do instead of wasting time on MTI is to actually make pr=
ogress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could actual=
ly interoperate regardless of the codec.

The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.

I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of "well" - and I would pref=
er that ecosystem to be one where it's possible to field the box without ma=
king prior arrangements with anyone about licenses.

This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.

We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)

I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.

At the moment, I don't see signs that any of the poll respondents have said=
 "I have changed my mind".

Harald




On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com<mailt=
o:martin.thomson@gmail.com>> wrote:
On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at<mailto:matthe=
w@matthew.at>> wrote:
And that's because something substantive has changed, or simply because
wasting the WG time on this again is more entertaining than actually
finishing a specification that can be independently implemented by all
browser vendors? (A specification that we are nowhere near having, as far a=
s
I can tell)
Personally, I've found the reprieve from this fight refreshing.  And
it would appear that we've made some real progress as a result.  I'd
suggest that if we don't have new information, we continue to spend
our time productively.  If we can't find topics to occupy our meeting
agenda time, then maybe we can free an agenda slot.

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

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



--
Alex. Gouaillard, PhD, PhD, MBA
---------------------------------------------------------------------------=
---------
CTO - Temasys Communications, S'pore / Mountain View
President - CoSMo Software, Cambridge, MA
---------------------------------------------------------------------------=
---------
sg.linkedin.com/agouaillard<http://sg.linkedin.com/agouaillard>
*

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



--
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net<mailto:jdrosen@jdrosen.net>
http://www.jdrosen.net

_______________________________________________
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


--

Surveillance is pervasive. Go Dark.

_______________________________________________
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



--_000_949EF20990823C4C85C18D59AA11AD8B263123FR712WXCHMBA11zeu_
Content-Type: text/html; charset="koi8-r"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dkoi8-r">
<meta content=3D"MSHTML 6.00.2900.6550" name=3D"GENERATOR">
</head>
<body>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">With respect to audio codecs I wa=
s replying to a question from the previous message in the thread. But...</f=
ont></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">The MTI discussion does not close=
 the door on any codec, as any codec can be supported, just not mandatory t=
o implement.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">Your argument to eliminate H.264 =
only applies if you can either eliminate interworking with the rest of the =
world or you can persuade all those legacy devices
 to remove H.264 and use something else, which in my view is totally imprac=
tical.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">MTI might persuade usage of that =
codec, but it will only do that if the MTI codec actually supports end to e=
nd communication of the required video capabilities.
 Neither VP8 or H.264 fit all those requirements.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span class=3D"577594807-21102014"><font fa=
ce=3D"Arial" color=3D"#0000ff" size=3D"2">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDE=
R-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<div class=3D"OutlookMessageHeader" lang=3D"en-us" dir=3D"ltr" align=3D"lef=
t">
<hr tabindex=3D"-1">
<font face=3D"Tahoma" size=3D"2"><b>From:</b> Justin Uberti [mailto:juberti=
@google.com]
<br>
<b>Sent:</b> 20 October 2014 23:29<br>
<b>To:</b> DRAGE, Keith (Keith)<br>
<b>Cc:</b> Andrew Allen; chris.cavigioli@intel.com; harald@alvestrand.no; b=
ernard.aboba@gmail.com; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</font><br>
</div>
<div></div>
<div dir=3D"ltr">From the subject &quot;Plan for MTI video codec&quot;, I t=
hought this discussion was about video codecs, not AMR interop.&nbsp;
<div><br>
</div>
<div>An astute observer might take note of the fact that our inability to c=
lose the door on H.264 is providing an opportunity to other royalty-bearing=
 codecs that want to join the party.</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (K=
eith) <span dir=3D"ltr">
&lt;<a href=3D"mailto:keith.drage@alcatel-lucent.com" target=3D"_blank">kei=
th.drage@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<u></u>
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">The scenario where you might want to transcode the audio codec is where y=
ou have wideband audio in both the OPUS and the AMR, and you believe by tra=
nscoding you can achieve a wideband quality,
 rather than falling back to G.711. And for interworking between 3GPP devic=
es that are not using webRTC, the audio will still be AMR or AMR-WB or EVC,=
 not OPUS.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">But audio transcoding is not particularly a concern for me. It is trying =
where possible to avoid video transcoding.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>&nbsp;</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDE=
R-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<div lang=3D"en-us" dir=3D"ltr" align=3D"left">
<hr>
<font face=3D"Tahoma"><b>From:</b> rtcweb [mailto:<a href=3D"mailto:rtcweb-=
bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Andrew Allen<br>
<b>Sent:</b> 20 October 2014 14:52<br>
<b>To:</b> <a href=3D"mailto:chris.cavigioli@intel.com" target=3D"_blank">c=
hris.cavigioli@intel.com</a>;
<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand=
.no</a>; <a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">
bernard.aboba@gmail.com</a>
<div>
<div class=3D"h5"><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</div>
</div>
</font><br>
</div>
<div>
<div class=3D"h5">
<div></div>
<font style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','san=
s-serif'"><br>
Chris<br>
<br>
What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?
<br>
<br>
All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.
<br>
<br>
My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.<br>
<br>
For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices
 interoperating with circuit switched only mobile devices which will be for=
ced to interoperate via the PSTN.<br>
<br>
As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with
 non-mobile RTCweb devices. <br>
<br>
The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.<br>
<br>
Andrew</font><br>
&nbsp;<br>
<div style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b=
5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: mediu=
m none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<font style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"><b>From=
</b>: Cavigioli, Chris [mailto:<a href=3D"mailto:chris.cavigioli@intel.com"=
 target=3D"_blank">chris.cavigioli@intel.com</a>]
<br>
<b>Sent</b>: Sunday, October 19, 2014 09:20 PM Eastern Standard Time<br>
<b>To</b>: Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;; Bernard Aboba &lt;<a href=3D"=
mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</=
a>&gt;
<br>
<b>Cc</b>: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a> &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt;
<br>
<b>Subject</b>: Re: [rtcweb] Plan for MTI video codec? <br>
</font>&nbsp;<br>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">Harald said:&nbsp;
<u></u><u></u></span></p>
<p class=3D"MsoNormal">&#8220;We made the mistake of having an MTI discussi=
on previously with not enough details on that subject&#8230;&#8221;<span st=
yle=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Arial','sans-serif'">=
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">We didn&#8217;t have quantitative data earli=
er for a data-driven decision.&nbsp; Future Web &#8211; LTE interop is impo=
rtant.&nbsp; A group of us in GSMA have been looking at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.&nbsp; GSMA has approved the whitepaper and i=
t is being prepared for public distribution.&nbsp;
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">WebRTC &#8211; LTE transcoding is now MANDAT=
ORY because of &#8220;WebRTC Audio Codec and Processing Requirements&#8221;=
.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.&nbsp; Thus, REGARDLESS=
 of the video codecs, there is already a
 significant degradation imposed on Web &#8211; LTE links if only MTI codec=
s are used.&nbsp; The only systems to avoid this are those which implement =
OPTIONAL audio codecs to be transcoder-free with LTE.&nbsp; The same can ap=
ply for video codecs.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">The GSMA whitepaper refers to:<u></u><u></u>=
</span></p>
<p><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Ari=
al','sans-serif'"><span>1.<span style=3D"FONT: 7pt 'Times New Roman'">&nbsp=
;&nbsp;&nbsp;&nbsp;
</span></span></span><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d;=
 FONT-FAMILY: 'Arial','sans-serif'">Quantitative voice-over-LTE (VoLTE) del=
ay limits were agreed-to by 3GPP SA4 in August 2014.&nbsp; To paraphrase, p=
erformance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> &#43; T<sub>R</sub>) should be =98 150ms. If this performance obje=
ctive cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> &#43; T<sub>R</s=
ub>) shall in any case be =98 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Ari=
al','sans-serif'"><span>2.<span style=3D"FONT: 7pt 'Times New Roman'">&nbsp=
;&nbsp;&nbsp;&nbsp;
</span></span></span><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d;=
 FONT-FAMILY: 'Arial','sans-serif'">OPUS &#8211; AMR transcoding adds an ad=
ditional 211.5 ms implementation-agnostic (T<sub>S</sub> &#43; T<sub>R</sub=
>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Ari=
al','sans-serif'"><span>3.<span style=3D"FONT: 7pt 'Times New Roman'">&nbsp=
;&nbsp;&nbsp;&nbsp;
</span></span></span><u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d;=
 FONT-FAMILY: 'Arial','sans-serif'">To put these numbers in perspective, it=
 is in general desirable to minimize UE delays to ensure low enough end-to-=
end delays and hence a good conversational
 experience.&nbsp; Increasing delay causes people to double-talk making con=
versations increasingly frustrating.&nbsp; Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">This is a complex topic with many network fa=
ctors in play.&nbsp; LTE operators in 3GPP seek UE delays (T<sub>S</sub> &#=
43; T<sub>R</sub>) around 150 ms.&nbsp; Adding OPUS
 &#8211; AMR transcoding in the network imposes an additional 211.5 ms on t=
op of that, just for audio.&nbsp; Optionally using AMR in WebRTC, those 211=
.5 ms can be eliminated, delivering a superior end-user experience.&nbsp;
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">CONCLUSION: &nbsp;regardless of MTI codec ch=
oices in WebRTC, to provide a good WebRTC &#8211; LTE interop experience, e=
merging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'">POSITION:&nbsp; To be clear, several Intel S=
oCs launched at MWC this year for smartphones and tablets support both VP8 =
and H.264 hardware encode and decode &#8211; so
 Intel is codec-neutral in this MTI debate.&nbsp; Additionally, Intel has h=
igh-performance solutions for transcoding.&nbsp; However, we wish to inform=
 the industry, with quantitative data, about impact to end-user experience =
of mandating such transcoding so IETF can
 make a data-driven decision on video codecs as well as reflect on the impa=
ct of already having mandated transcoding on the audio side.<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><span style=3D"COLOR: #1f497d; FONT-FAMILY: 'Lucida =
Console'">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'">Intel Corp,
</span><span style=3D"FONT-SIZE: 8pt; COLOR: #0070c0; FONT-FAMILY: 'Verdana=
','sans-serif'">System Architecture and Planning</span><span style=3D"FONT-=
SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verdana','sans-serif'">, Video/Mul=
timedia, Mobile and Communications Group
 (MCG)</span><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'V=
erdana','sans-serif'"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'"><a href=3D"tel:%2B1%20%28415%29%20254-4545"=
 target=3D"_blank" value=3D"&#43;14152544545">&#43;1 (415) 254-4545</a> mob=
ile</span><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verd=
ana','sans-serif'"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'"><a href=3D"tel:%2B1%20%28408%29%20653-8348"=
 target=3D"_blank" value=3D"&#43;14086538348">&#43;1 (408) 653-8348</a> des=
k<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-=
FAMILY: 'Verdana','sans-serif'"><a href=3D"tel:%2B1%20%28408%29%20884-2400"=
 target=3D"_blank" value=3D"&#43;14088842400">&#43;1 (408) 884-2400</a> fax=
</span><span style=3D"FONT-SIZE: 8pt; COLOR: #1f497d; FONT-FAMILY: 'Verdana=
','sans-serif'"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 7.5pt; COLOR: gray; FONT-F=
AMILY: 'Verdana','sans-serif'">This e-mail may contain confidential and pri=
vileged material for the sole use of the intended recipient(s).&nbsp; Any r=
eview or distribution by others is strictly
 prohibited. If you are not the intended recipient, please contact the send=
er and delete all copies.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT=
-FAMILY: 'Arial','sans-serif'"><u></u><u></u></span>&nbsp;</p>
<p class=3D"MsoNormal"><b><span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tah=
oma','sans-serif'">From:</span></b><span style=3D"FONT-SIZE: 10pt; FONT-FAM=
ILY: 'Tahoma','sans-serif'"> rtcweb [mailto:<a href=3D"mailto:rtcweb-bounce=
s@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
<div>
<p class=3D"MsoNormal">Harald said:&nbsp;<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: '=
Arial','sans-serif'">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><span style=3D"FONT-SI=
ZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial'=
,'sans-serif'">So it can't be all bad.....</span>&quot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">[BA] &nbsp;I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: &nbsp; that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. &nbsp; The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; &nbsp;to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. &nbsp;&nbsp;<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"MARGIN-TOP: 5pt; MARGIN-BOTTOM: 5pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.&nbsp; It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.&nbsp; <u>
</u><u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can't be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 &nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I'm in favor of taking another run at this. <u></u><=
u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco's open264 plugin shipped and now =
just recently is integrated into Firefox. iOS8 shipped with APIs for H264. =
There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">&#43;1 to not having MTI codec discussion unless som=
e progress has been made on VP8 at MPEG.&nbsp;Any news on that? I'm sharing=
 harald's &nbsp;feeling that there was no change on the members position.&n=
bsp;<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it's possible to field the box =
without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn't changed one whit since last time we discussed it. And =
I don't see much movement on the specifics of the proposals either.<br>
<br>
We'll have to interoperate well with the codecs we field. So using some tim=
e to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn't fi=
nished either. There's one sentence that needs to be removed.)<br>
<br>
I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd prefer=
 to re-discuss based on the knowledge that some significant players have ac=
tually changed their minds.<br>
<br>
At the moment, I don't see signs that any of the poll respondents have said=
 &quot;I have changed my mind&quot;.<span style=3D"COLOR: #888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt">On Oct 16, 2014, at 2:=
28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targe=
t=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that's because something substantive has changed, or simply because<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I've found the reprieve from this fight =
refreshing.&nbsp; And<br>
it would appear that we've made some real progress as a result.&nbsp; I'd<b=
r>
suggest that if we don't have new information, we continue to spend<br>
our time productively.&nbsp; If we can't find topics to occupy our meeting<=
br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">-- <u></u><u></u></sp=
an></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">Alex. Gouaillard, PhD=
, PhD, MBA
<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">---------------------=
---------------------------------------------------------------<u></u><u></=
u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">CTO - Temasys Communi=
cations, S'pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">President - CoSMo Sof=
tware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888">---------------------=
---------------------------------------------------------------<u></u><u></=
u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR: #888888"><a href=3D"http://sg.=
linkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a>=
<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-LEFT: 0in; VERTICAL-ALIGN: baseline;=
 LINE-HEIGHT: 14.4pt">
<u></u><span style=3D"FONT-SIZE: 10pt; COLOR: #333333; FONT-FAMILY: Symbol"=
><span>=9E<span style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><u></u><span style=3D"FONT-SIZE: 8.5pt; COLOR: #333333=
; FONT-FAMILY: inherit"><u></u>&nbsp;<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><u></u><u></u>&nbsp;</=
p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><u></u><u></u>&nbsp;</=
p>
</div>
</div>
<pre><span style=3D"COLOR: #888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"COLOR: #888888">Surveillance is pervasive. Go Dark.<u><=
/u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM: 12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u><u></u>&nbsp;</p>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</body>
</html>

--_000_949EF20990823C4C85C18D59AA11AD8B263123FR712WXCHMBA11zeu_--


From nobody Tue Oct 21 07:32:27 2014
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To: Emil Ivov <emcho@jitsi.org>, "Cullen Jenning (work)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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On: 20 October 2014 20:42 Emil Ivov Wrote:
> On 20 Oct 2014 8:55 PM, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
> wrote:
> >
> >
> > So lets be clear here.
>=20
> Good idea!
>=20
> > Every time we checked, there has been consensus that we need an MTI
> codec.
> > So right now this work is sort of blocked on that
>=20
> So clearly, what work is blocked on that exactly?
>=20

The only thing that seems to be blocked is the MTI Codec decision and here =
I think we need to accept that no decision is better than a wrong decision =
or a decision that divides the community.=20

Therefore I suggest we continue to ignore the issue and move on.

Andy


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From: Ted Hardie <ted.ietf@gmail.com>
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--001a1140cac02877a00505efe220
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I would appreciate folks moving off the question of whether the video codec
discussion should be part of the agenda could change the topic.

thanks,

Ted

On Tue, Oct 21, 2014 at 2:08 AM, DRAGE, Keith (Keith) <
keith.drage@alcatel-lucent.com> wrote:

>  With respect to audio codecs I was replying to a question from the
> previous message in the thread. But...
>
> The MTI discussion does not close the door on any codec, as any codec can
> be supported, just not mandatory to implement.
>
> Your argument to eliminate H.264 only applies if you can either eliminate
> interworking with the rest of the world or you can persuade all those
> legacy devices to remove H.264 and use something else, which in my view i=
s
> totally impractical.
>
> MTI might persuade usage of that codec, but it will only do that if the
> MTI codec actually supports end to end communication of the required vide=
o
> capabilities. Neither VP8 or H.264 fit all those requirements.
>
> Keith
>
>  ------------------------------
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* 20 October 2014 23:29
> *To:* DRAGE, Keith (Keith)
> *Cc:* Andrew Allen; chris.cavigioli@intel.com; harald@alvestrand.no;
> bernard.aboba@gmail.com; rtcweb@ietf.org
>
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>  From the subject "Plan for MTI video codec", I thought this discussion
> was about video codecs, not AMR interop.
>
>  An astute observer might take note of the fact that our inability to
> close the door on H.264 is providing an opportunity to other
> royalty-bearing codecs that want to join the party.
>
> On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (Keith) <
> keith.drage@alcatel-lucent.com> wrote:
>
>>  The scenario where you might want to transcode the audio codec is where
>> you have wideband audio in both the OPUS and the AMR, and you believe by
>> transcoding you can achieve a wideband quality, rather than falling back=
 to
>> G.711. And for interworking between 3GPP devices that are not using webR=
TC,
>> the audio will still be AMR or AMR-WB or EVC, not OPUS.
>>
>> But audio transcoding is not particularly a concern for me. It is trying
>> where possible to avoid video transcoding.
>>
>> Keith
>>
>>  ------------------------------
>> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Andrew
>> Allen
>> *Sent:* 20 October 2014 14:52
>> *To:* chris.cavigioli@intel.com; harald@alvestrand.no;
>> bernard.aboba@gmail.com
>>
>> *Cc:* rtcweb@ietf.org
>> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>>
>>
>> Chris
>>
>> What is the scenario that dictates the need for direct transcoding
>> between OPUS and AMR?
>>
>> All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE
>> devices OPUS can be negotiated with non mobile RTCweb devices end to end=
.
>>
>> My expectation is that for mobile devices the RTCweb client will also
>> have access to the embedded AMR codec so mobile RTCweb devices can
>> interoperate using AMR with non-RTCweb enabled mobile IMS devices that
>> support AMR.
>>
>> For interoperation between non mobile RTCweb devices and mobile non
>> RTCweb IMS devices the non mobile RTCweb device can use G.711 to the
>> transcoder for transcoding to AMR on the other side. While not ideal thi=
s
>> will be no worse than non mobile RTCweb devices interoperating with circ=
uit
>> switched only mobile devices which will be forced to interoperate via th=
e
>> PSTN.
>>
>> As RTCweb becomes universal on mobile devices I expect that support for
>> OPUS will become native and mobile IMS devices will also be able to use
>> OPUS even when operating without the use of the browser (i.e. in non RTC=
web
>> mode) for better interoperation with non-mobile RTCweb devices.
>>
>> The lifetime of most mobile devices is 2 years or less so this is likely
>> to be mainly a short term transition issue only.
>>
>> Andrew
>>
>>  *From*: Cavigioli, Chris [mailto:chris.cavigioli@intel.com]
>> *Sent*: Sunday, October 19, 2014 09:20 PM Eastern Standard Time
>> *To*: Harald Alvestrand <harald@alvestrand.no>; Bernard Aboba <
>> bernard.aboba@gmail.com>
>> *Cc*: rtcweb@ietf.org <rtcweb@ietf.org>
>> *Subject*: Re: [rtcweb] Plan for MTI video codec?
>>
>>
>> Harald said:
>>
>> =E2=80=9CWe made the mistake of having an MTI discussion previously with=
 not
>> enough details on that subject=E2=80=A6=E2=80=9D
>>
>>
>>
>> We didn=E2=80=99t have quantitative data earlier for a data-driven decis=
ion.
>> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA ha=
ve been
>> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
>> voice/video/messaging), specifically looking at user experience impact o=
f
>> latency introduced by added in-network transcoding.  GSMA has approved t=
he
>> whitepaper and it is being prepared for public distribution.
>>
>>
>>
>> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWe=
bRTC Audio Codec
>> and Processing Requirements=E2=80=9D.
>> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
>> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vid=
eo
>> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
>> links if only MTI codecs are used.  The only systems to avoid this are
>> those which implement OPTIONAL audio codecs to be transcoder-free with
>> LTE.  The same can apply for video codecs.
>>
>>
>>
>> The GSMA whitepaper refers to:
>>
>> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to
>> by 3GPP SA4 in August 2014.  To paraphrase, performance objectives for
>> maximum VoLTE device delays in error and jitter free conditions and AMR
>> speech codec operation, the sum of the UE delays in sending and receivin=
g
>> directions (TS + TR) should be =E2=89=A4 150ms. If this performance obje=
ctive
>> cannot be met, the sum of the UE delays in sending and receiving directi=
ons
>> (TS + TR) shall in any case be =E2=89=A4 190ms.
>>
>> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
>> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
>> buffer and 40 ms for discontinuous reception (DRX).
>>
>> 3.     To put these numbers in perspective, it is in general desirable
>> to minimize UE delays to ensure low enough end-to-end delays and hence a
>> good conversational experience.  Increasing delay causes people to
>> double-talk making conversations increasingly frustrating.  Guidance to
>> stay below 400 ms maximum one-way delay is found in ITU-T Recommendation
>> G.114 and the computational E-model is in ITU-T Recommendation G.107.
>>
>>
>>
>> This is a complex topic with many network factors in play.  LTE operator=
s
>> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 A=
MR
>> transcoding in the network imposes an additional 211.5 ms on top of that=
,
>> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
>> eliminated, delivering a superior end-user experience.
>>
>>
>>
>> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a goo=
d
>> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must ac=
commodate
>> MTI codecs already deployed in the LTE domain and in mobile operating
>> systems, namely AMR/AMR-WB and H.264.
>>
>>
>>
>> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
>> smartphones and tablets support both VP8 and H.264 hardware encode and
>> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Addition=
ally, Intel
>> has high-performance solutions for transcoding.  However, we wish to inf=
orm
>> the industry, with quantitative data, about impact to end-user experienc=
e
>> of mandating such transcoding so IETF can make a data-driven decision on
>> video codecs as well as reflect on the impact of already having mandated
>> transcoding on the audio side.
>>
>>
>>
>> Chris Cavigioli
>>
>> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
>> and Communications Group (MCG)
>>
>> +1 (415) 254-4545 mobile
>>
>> +1 (408) 653-8348 desk
>>
>> +1 (408) 884-2400 fax
>>
>> This e-mail may contain confidential and privileged material for the sol=
e
>> use of the intended recipient(s).  Any review or distribution by others =
is
>> strictly prohibited. If you are not the intended recipient, please conta=
ct
>> the sender and delete all copies.
>>
>>
>>
>> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Bernard
>> Aboba
>> *Sent:* Sunday, October 19, 2014 2:29 PM
>> *To:* Harald Alvestrand
>> *Cc:* rtcweb@ietf.org
>> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>>
>>
>>
>> Harald said:
>>
>>
>>
>> "Bernard,
>>
>>
>> I think this is, to a large degree, codec independent.
>>
>> We have demonstrated interoperability on VP8 between Firefox and Chrome,
>> and usage of various mechanisms for congestion control and repair of pac=
ket
>> loss being applied in live services.
>>
>> So it can't be all bad....."
>>
>>
>>
>> [BA]  I agree that the lack of interoperability is largely "codec
>> independent":   that is, implementations based on different mechanisms f=
or
>> congestion control, repair, etc. will exhibit interoperability problems,
>> regardless of the codec chosen.   The real test for RTCWEB will be to mo=
ve
>> beyond "interoperability of implementations sharing source code"  to
>> "interoperability of independent implementations, based on open standard=
s".
>>
>>
>>
>>
>> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no=
>
>> wrote:
>>
>> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>>
>>  "And its one of the issues holding up wider adoption of the technology"
>>
>>
>>
>> [BA] Specifying an MTI encoder/decoder is not sufficient for
>> interoperability.  It is also necessary to specify the transport in enou=
gh
>> detail to allow independent implementations that interoperate well enoug=
h
>> to be usable in a wide variety of scenarios, including wireless networks
>> where loss is commonly experienced.
>>
>>
>> Bernard,
>>
>> I think this is, to a large degree, codec independent.
>>
>> We have demonstrated interoperability on VP8 between Firefox and Chrome,
>> and usage of various mechanisms for congestion control and repair of pac=
ket
>> loss being applied in live services.
>>
>> So it can't be all bad.....
>>
>>
>>
>>
>>
>>
>> We made the mistake of having an MTI discussion previously with not
>> enough details on that subject, particularly with respect to H.264.
>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>>
>>
>>
>> So if we are to have the discussion again, it should occur in the contex=
t
>> of detailed specifications and interoperability reports.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net=
>
>> wrote:
>>
>> I'm in favor of taking another run at this.
>>
>>
>>
>> The working group has repeatedly said that an MTI codec is something we
>> need to produce. And its one of the issues holding up wider adoption of =
the
>> technology (not the only one for sure).
>>
>>
>>
>> And things have changed since the last meeting, a year ago now (November
>> in Vancouver). Cisco's open264 plugin shipped and now just recently is
>> integrated into Firefox. iOS8 shipped with APIs for H264. There are othe=
r
>> things worth noting. Will this change the minds of everyone? Surely not.
>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wor=
th
>> finding out.
>>
>>
>>
>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
>> agouaillard@gmail.com> wrote:
>>
>> +1 to not having MTI codec discussion unless some progress has been made
>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that the=
re
>> was no change on the members position.
>>
>>
>>
>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no=
>
>> wrote:
>>
>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>>
>> One thing we could do instead of wasting time on MTI is to actually make
>> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
>> actually interoperate regardless of the codec.
>>
>>
>> The big argument for an MTI is actually the one stated in -overview: It
>> guards against interoperability failure.
>>
>> I would like to have an ecosystem where one can field a box, connect it
>> to everything else, and run well for *some* level of "well" - and I woul=
d
>> prefer that ecosystem to be one where it's possible to field the box
>> without making prior arrangements with anyone about licenses.
>>
>> This argument hasn't changed one whit since last time we discussed it.
>> And I don't see much movement on the specifics of the proposals either.
>>
>> We'll have to interoperate well with the codecs we field. So using some
>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 is=
n't
>> finished either. There's one sentence that needs to be removed.)
>>
>> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
>> prefer to re-discuss based on the knowledge that some significant player=
s
>> have actually changed their minds.
>>
>> At the moment, I don't see signs that any of the poll respondents have
>> said "I have changed my mind".
>>
>> Harald
>>
>>
>>
>>
>>
>>  On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
>> wrote:
>>
>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
>> And that's because something substantive has changed, or simply because
>> wasting the WG time on this again is more entertaining than actually
>> finishing a specification that can be independently implemented by all
>> browser vendors? (A specification that we are nowhere near having, as fa=
r
>> as
>> I can tell)
>>
>> Personally, I've found the reprieve from this fight refreshing.  And
>> it would appear that we've made some real progress as a result.  I'd
>> suggest that if we don't have new information, we continue to spend
>> our time productively.  If we can't find topics to occupy our meeting
>> agenda time, then maybe we can free an agenda slot.
>>
>> _______________________________________________
>> 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
>>
>>
>>
>>
>>
>> --
>>
>> Alex. Gouaillard, PhD, PhD, MBA
>>
>>
>> ------------------------------------------------------------------------=
------------
>>
>> CTO - Temasys Communications, S'pore / Mountain View
>>
>> President - CoSMo Software, Cambridge, MA
>>
>>
>> ------------------------------------------------------------------------=
------------
>>
>> sg.linkedin.com/agouaillard
>>
>> =C2=B7
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>>
>>
>> --
>>
>> Jonathan Rosenberg, Ph.D.
>> jdrosen@jdrosen.net
>> 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
>>
>>
>>
>> --
>>
>> Surveillance is pervasive. Go Dark.
>>
>>
>> _______________________________________________
>> 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
>
>

--001a1140cac02877a00505efe220
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">I would appreciate folks moving off the question=
 of whether the video codec discussion should be part of the agenda could c=
hange the topic.<br><br>thanks,<br><br>Ted<br></div></div><div class=3D"gma=
il_extra"><br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 2:08 AM, D=
RAGE, Keith (Keith) <span dir=3D"ltr">&lt;<a href=3D"mailto:keith.drage@alc=
atel-lucent.com" target=3D"_blank">keith.drage@alcatel-lucent.com</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><u></u>





<div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">With respect to audio codecs I was replying to a question from the previo=
us message in the thread. But...</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">The MTI discussion does not close the door on any codec, as any codec can=
 be supported, just not mandatory to implement.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">Your argument to eliminate H.264 only applies if you can either eliminate=
 interworking with the rest of the world or you can persuade all those lega=
cy devices
 to remove H.264 and use something else, which in my view is totally imprac=
tical.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">MTI might persuade usage of that codec, but it will only do that if the M=
TI codec actually supports end to end communication of the required video c=
apabilities.
 Neither VP8 or H.264 fit all those requirements.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT:5px;MARGIN-LEFT:5px;BORDER-LE=
FT:#0000ff 2px solid;MARGIN-RIGHT:0px">
<div dir=3D"ltr" lang=3D"en-us" align=3D"left">
<hr>
<font face=3D"Tahoma"><span class=3D""><b>From:</b> Justin Uberti [mailto:<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>]
<br>
</span><b>Sent:</b> 20 October 2014 23:29<br>
<b>To:</b> DRAGE, Keith (Keith)<br>
<b>Cc:</b> Andrew Allen; <a href=3D"mailto:chris.cavigioli@intel.com" targe=
t=3D"_blank">chris.cavigioli@intel.com</a>; <a href=3D"mailto:harald@alvest=
rand.no" target=3D"_blank">harald@alvestrand.no</a>; <a href=3D"mailto:bern=
ard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>; <a href=
=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><div><div =
class=3D"h5"><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</div></div></font><br>
</div><div><div class=3D"h5">
<div></div>
<div dir=3D"ltr">From the subject &quot;Plan for MTI video codec&quot;, I t=
hought this discussion was about video codecs, not AMR interop.=C2=A0
<div><br>
</div>
<div>An astute observer might take note of the fact that our inability to c=
lose the door on H.264 is providing an opportunity to other royalty-bearing=
 codecs that want to join the party.</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Mon, Oct 20, 2014 at 7:34 AM, DRAGE, Keith (K=
eith) <span dir=3D"ltr">
&lt;<a href=3D"mailto:keith.drage@alcatel-lucent.com" target=3D"_blank">kei=
th.drage@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT:1ex;MARGIN:0px 0px =
0px 0.8ex;BORDER-LEFT:#ccc 1px solid">
<u></u>
<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">The scenario where you might want to transcode the audio codec is where y=
ou have wideband audio in both the OPUS and the AMR, and you believe by tra=
nscoding you can achieve a wideband quality,
 rather than falling back to G.711. And for interworking between 3GPP devic=
es that are not using webRTC, the audio will still be AMR or AMR-WB or EVC,=
 not OPUS.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">But audio transcoding is not particularly a concern for me. It is trying =
where possible to avoid video transcoding.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
"></font></span>=C2=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
">Keith</font></span></div>
<br>
<blockquote dir=3D"ltr" style=3D"PADDING-LEFT:5px;MARGIN-LEFT:5px;BORDER-LE=
FT:#0000ff 2px solid;MARGIN-RIGHT:0px">
<div dir=3D"ltr" lang=3D"en-us" align=3D"left">
<hr>
<font face=3D"Tahoma"><b>From:</b> rtcweb [mailto:<a href=3D"mailto:rtcweb-=
bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Andrew Allen<br>
<b>Sent:</b> 20 October 2014 14:52<br>
<b>To:</b> <a href=3D"mailto:chris.cavigioli@intel.com" target=3D"_blank">c=
hris.cavigioli@intel.com</a>;
<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand=
.no</a>; <a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">
bernard.aboba@gmail.com</a>
<div>
<div><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<br>
</div>
</div>
</font><br>
</div>
<div>
<div>
<div></div>
<font style=3D"FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:&#39;Calibri&#39;,&=
#39;sans-serif&#39;"><br>
Chris<br>
<br>
What is the scenario that dictates the need for direct transcoding between =
OPUS and AMR?
<br>
<br>
All RTCweb clients MUST support OPUS so for Web browsers on mobile LTE devi=
ces OPUS can be negotiated with non mobile RTCweb devices end to end.
<br>
<br>
My expectation is that for mobile devices the RTCweb client will also have =
access to the embedded AMR codec so mobile RTCweb devices can interoperate =
using AMR with non-RTCweb enabled mobile IMS devices that support AMR.<br>
<br>
For interoperation between non mobile RTCweb devices and mobile non RTCweb =
IMS devices the non mobile RTCweb device can use G.711 to the transcoder fo=
r transcoding to AMR on the other side. While not ideal this will be no wor=
se than non mobile RTCweb devices
 interoperating with circuit switched only mobile devices which will be for=
ced to interoperate via the PSTN.<br>
<br>
As RTCweb becomes universal on mobile devices I expect that support for OPU=
S will become native and mobile IMS devices will also be able to use OPUS e=
ven when operating without the use of the browser (i.e. in non RTCweb mode)=
 for better interoperation with
 non-mobile RTCweb devices. <br>
<br>
The lifetime of most mobile devices is 2 years or less so this is likely to=
 be mainly a short term transition issue only.<br>
<br>
Andrew</font><br>
=C2=A0<br>
<div style=3D"BORDER-RIGHT:medium none;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df=
 1pt solid;PADDING-LEFT:0in;PADDING-BOTTOM:0in;BORDER-LEFT:medium none;PADD=
ING-TOP:3pt;BORDER-BOTTOM:medium none">
<font style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Tahoma&#39;,&#39;sans-serif&=
#39;"><b>From</b>: Cavigioli, Chris [mailto:<a href=3D"mailto:chris.cavigio=
li@intel.com" target=3D"_blank">chris.cavigioli@intel.com</a>]
<br>
<b>Sent</b>: Sunday, October 19, 2014 09:20 PM Eastern Standard Time<br>
<b>To</b>: Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;; Bernard Aboba &lt;<a href=3D"=
mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</=
a>&gt;
<br>
<b>Cc</b>: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a> &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt;
<br>
<b>Subject</b>: Re: [rtcweb] Plan for MTI video codec? <br>
</font>=C2=A0<br>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Arial&#39;,&#39;=
sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">We didn=E2=80=99t have quantitati=
ve data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=93 LTE =
interop is important.=C2=A0 A group of us in GSMA have been looking at WebR=
TC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">WebRTC =E2=80=93 LTE transcoding =
is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processing Requ=
irements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a
 significant degradation imposed on Web =E2=80=93 LTE links if only MTI cod=
ecs are used.=C2=A0 The only systems to avoid this are those which implemen=
t OPTIONAL audio codecs to be transcoder-free with LTE.=C2=A0 The same can =
apply for video codecs.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">The GSMA whitepaper refers to:<u>=
</u><u></u></span></p>
<p><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Aria=
l&#39;,&#39;sans-serif&#39;"><span>1.<span style=3D"FONT:7pt &#39;Times New=
 Roman&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">Quantitative voice-over-LTE =
(VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0 To pa=
raphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Aria=
l&#39;,&#39;sans-serif&#39;"><span>2.<span style=3D"FONT:7pt &#39;Times New=
 Roman&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">OPUS =E2=80=93 AMR transcodi=
ng adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub> + T<s=
ub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAMILY:&#39;Aria=
l&#39;,&#39;sans-serif&#39;"><span>3.<span style=3D"FONT:7pt &#39;Times New=
 Roman&#39;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FON=
T-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;">To put these numbers in pers=
pective, it is in general desirable to minimize UE delays to ensure low eno=
ugh end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">This is a complex topic with many=
 network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays (T<sub=
>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS
 =E2=80=93 AMR transcoding in the network imposes an additional 211.5 ms on=
 top of that, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 2=
11.5 ms can be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">CONCLUSION: =C2=A0regardless of M=
TI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE interop =
experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;">POSITION:=C2=A0 To be clear, seve=
ral Intel SoCs launched at MWC this year for smartphones and tablets suppor=
t both VP8 and H.264 hardware encode and decode =E2=80=93 so
 Intel is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has h=
igh-performance solutions for transcoding.=C2=A0 However, we wish to inform=
 the industry, with quantitative data, about impact to end-user experience =
of mandating such transcoding so IETF can
 make a data-driven decision on video codecs as well as reflect on the impa=
ct of already having mandated transcoding on the audio side.<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><span style=3D"COLOR:#1f497d;FONT-FAMILY:&#39;Lucida=
 Console&#39;">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;">Intel Corp,
</span><span style=3D"FONT-SIZE:8pt;COLOR:#0070c0;FONT-FAMILY:&#39;Verdana&=
#39;,&#39;sans-serif&#39;">System Architecture and Planning</span><span sty=
le=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Verdana&#39;,&#39;sans-s=
erif&#39;">, Video/Multimedia, Mobile and Communications Group
 (MCG)</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Ve=
rdana&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28415%29%=
20254-4545" value=3D"+14152544545" target=3D"_blank">+1 (415) 254-4545</a> =
mobile</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Ve=
rdana&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28408%29%=
20653-8348" value=3D"+14086538348" target=3D"_blank">+1 (408) 653-8348</a> =
desk<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMI=
LY:&#39;Verdana&#39;,&#39;sans-serif&#39;"><a href=3D"tel:%2B1%20%28408%29%=
20884-2400" value=3D"+14088842400" target=3D"_blank">+1 (408) 884-2400</a> =
fax</span><span style=3D"FONT-SIZE:8pt;COLOR:#1f497d;FONT-FAMILY:&#39;Verda=
na&#39;,&#39;sans-serif&#39;"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:7.5pt;COLOR:gray;FONT-FAMIL=
Y:&#39;Verdana&#39;,&#39;sans-serif&#39;">This e-mail may contain confident=
ial and privileged material for the sole use of the intended recipient(s).=
=C2=A0 Any review or distribution by others is strictly
 prohibited. If you are not the intended recipient, please contact the send=
er and delete all copies.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;COLOR:#1f497d;FONT-FAM=
ILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><u></u><u></u></span>=C2=A0</p>
<p class=3D"MsoNormal"><b><span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Ta=
homa&#39;,&#39;sans-serif&#39;">From:</span></b><span style=3D"FONT-SIZE:10=
pt;FONT-FAMILY:&#39;Tahoma&#39;,&#39;sans-serif&#39;"> rtcweb [mailto:<a hr=
ef=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf=
.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39=
;Arial&#39;,&#39;sans-serif&#39;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><span style=3D"FONT-SIZ=
E:10pt;FONT-FAMILY:&#39;Arial&#39;,&#39;sans-serif&#39;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-SIZE:10pt;FONT-FAMILY:&#39;Arial=
&#39;,&#39;sans-serif&#39;">So it can&#39;t be all bad.....</span>&quot;<u>=
</u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"MARGIN-TOP:5pt;MARGIN-BOTTOM:5pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u>
</u><u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"COLOR:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt">On Oct 16, 2014, at 2:2=
8 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" target=
=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA
<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"COLOR:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"MARGIN-LEFT:0in;VERTICAL-ALIGN:baseline;LIN=
E-HEIGHT:14.4pt">
<u></u><span style=3D"FONT-SIZE:10pt;COLOR:#333333;FONT-FAMILY:Symbol"><spa=
n>=C2=B7<span style=3D"FONT:7pt &#39;Times New Roman&#39;">=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"FONT-SIZE:8.5pt;COLOR:#333333;FO=
NT-FAMILY:inherit"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><u></u><u></u>=C2=A0</p=
>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><u></u><u></u>=C2=A0</p=
>
</div>
</div>
<pre><span style=3D"COLOR:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"COLOR:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"MARGIN-BOTTOM:12pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u><u></u>=C2=A0</p>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div></div></blockquote>
</div>

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

--001a1140cac02877a00505efe220--


From nobody Tue Oct 21 09:26:06 2014
Return-Path: <ibc@aliax.net>
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 21 Oct 2014 18:25:38 +0200
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Subject: [rtcweb] SDP and ssrc-group,
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Hi,

May I know which SSRC (345259865 or 2693756249) will be used for the
real media stream (plus RED and FEC) and which SSRC will be used for
RTX?



--------------------------
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
a=3Dmid:video
a=3Drtpmap:100 VP8/90000
a=3Drtpmap:116 red/90000
a=3Drtpmap:117 ulpfec/90000
a=3Drtpmap:96 rtx/90000
a=3Dfmtp:96 apt=3D100
a=3Dssrc-group:FID 345259865 2693756249
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
-------------------------------




RFC 5576 does not clarify it at all:

http://tools.ietf.org/html/rfc5576#section-4.2

--------------------------------------------------
4.2.  The "ssrc-group" Media Attribute

   a=3Dssrc-group:<semantics> <ssrc-id> ...

   [..]

   The <semantics> parameter is taken from the specification of the
   "group" attribute [RFC3388].  The initial semantic values defined for
   the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
   and FEC (Forward Error Correction) [RFC4756].  In each case, the
   relationship among the grouped sources is the same as the
   relationship among corresponding sources in media streams grouped
   using the SDP "group" attribute.
--------------------------------------------------



The referenced RFC 3388 neither clarifies it:

---------------------------------------------------
7.4 FID Semantics

   Several "m" lines grouped together using FID semantics form a media
   flow.  A media agent handling a media flow that comprises several "m"
   lines MUST send a copy of the media to every "m" line part of the
   flow as long as the codecs and the direction attribute present in a
   particular "m" line allow it.

   It is assumed that the application uses only one codec at a time to
   encode the media produced.  This codec MAY change dynamically during
   the session, but at any particular moment only one codec is in use.

   The application encodes the media using the current codec and checks
   one by one all the "m" lines that are part of the flow.  If a
   particular "m" line contains the codec being used and the direction
   attribute is "sendonly" or "sendrecv", a copy of the encoded media is
   sent to the address/port specified in that particular media stream.
   If either the "m" line does not contain the codec being used or the
   direction attribute is neither "sendonly" nor "sendrecv", nothing is
   sent over this media stream.
----------------------------------------------------




So, how is the usage of ssrc-group? Where is it really defined?

Can I put more than 2 ssrc together in the same ssrc-group line?

How should the receiver interpret it?

Does a ssrc-group always mean RTX usage? Where is that specified in
the above SDP?

Does not the above SDP look a complete mixture of hacks and workarounds?




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


From nobody Tue Oct 21 10:31:48 2014
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From: Peter Thatcher <pthatcher@google.com>
Date: Tue, 21 Oct 2014 10:31:01 -0700
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Subject: Re: [rtcweb] SDP and ssrc-group,
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--001a11c1c3245946dd0505f234d1
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345259865 is "real"
2693756249 is rtx

On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> Hi,
>
> May I know which SSRC (345259865 or 2693756249) will be used for the
> real media stream (plus RED and FEC) and which SSRC will be used for
> RTX?
>
>
>
> --------------------------
> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
> a=3Dmid:video
> a=3Drtpmap:100 VP8/90000
> a=3Drtpmap:116 red/90000
> a=3Drtpmap:117 ulpfec/90000
> a=3Drtpmap:96 rtx/90000
> a=3Dfmtp:96 apt=3D100
> a=3Dssrc-group:FID 345259865 2693756249
> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> -------------------------------
>
>
>
>
> RFC 5576 does not clarify it at all:
>
> http://tools.ietf.org/html/rfc5576#section-4.2
>
> --------------------------------------------------
> 4.2.  The "ssrc-group" Media Attribute
>
>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>
>    [..]
>
>    The <semantics> parameter is taken from the specification of the
>    "group" attribute [RFC3388].  The initial semantic values defined for
>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>    relationship among the grouped sources is the same as the
>    relationship among corresponding sources in media streams grouped
>    using the SDP "group" attribute.
> --------------------------------------------------
>
>
>
> The referenced RFC 3388 neither clarifies it:
>
> ---------------------------------------------------
> 7.4 FID Semantics
>
>    Several "m" lines grouped together using FID semantics form a media
>    flow.  A media agent handling a media flow that comprises several "m"
>    lines MUST send a copy of the media to every "m" line part of the
>    flow as long as the codecs and the direction attribute present in a
>    particular "m" line allow it.
>
>    It is assumed that the application uses only one codec at a time to
>    encode the media produced.  This codec MAY change dynamically during
>    the session, but at any particular moment only one codec is in use.
>
>    The application encodes the media using the current codec and checks
>    one by one all the "m" lines that are part of the flow.  If a
>    particular "m" line contains the codec being used and the direction
>    attribute is "sendonly" or "sendrecv", a copy of the encoded media is
>    sent to the address/port specified in that particular media stream.
>    If either the "m" line does not contain the codec being used or the
>    direction attribute is neither "sendonly" nor "sendrecv", nothing is
>    sent over this media stream.
> ----------------------------------------------------
>
>
>
>
> So, how is the usage of ssrc-group? Where is it really defined?
>
> Can I put more than 2 ssrc together in the same ssrc-group line?
>
> How should the receiver interpret it?
>
> Does a ssrc-group always mean RTX usage? Where is that specified in
> the above SDP?
>
> Does not the above SDP look a complete mixture of hacks and workarounds?
>
>
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c1c3245946dd0505f234d1
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><span style=3D"font-family:arial,sans-serif;font-size:1=
2.7272720336914px">345259865 is &quot;real&quot;</span><br></div><div class=
=3D"gmail_default" style><font face=3D"arial, sans-serif">2693756249 is rtx=
</font><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_q=
uote">On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <span dir=3D=
"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 2693756249<br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
=C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the specificatio=
n of the<br>
=C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initial seman=
tic values defined for<br>
=C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Identificat=
ion) [RFC3388]<br>
=C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In each ca=
se, the<br>
=C2=A0 =C2=A0relationship among the grouped sources is the same as the<br>
=C2=A0 =C2=A0relationship among corresponding sources in media streams grou=
ped<br>
=C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID semanti=
cs form a media<br>
=C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that comprises=
 several &quot;m&quot;<br>
=C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&quot; lin=
e part of the<br>
=C2=A0 =C2=A0flow as long as the codecs and the direction attribute present=
 in a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0It is assumed that the application uses only one codec at a ti=
me to<br>
=C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change dynamic=
ally during<br>
=C2=A0 =C2=A0the session, but at any particular moment only one codec is in=
 use.<br>
<br>
=C2=A0 =C2=A0The application encodes the media using the current codec and =
checks<br>
=C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of the fl=
ow.=C2=A0 If a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being used an=
d the direction<br>
=C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quot;, a c=
opy of the encoded media is<br>
=C2=A0 =C2=A0sent to the address/port specified in that particular media st=
ream.<br>
=C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the codec be=
ing used or the<br>
=C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor &quot;=
sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

--001a11c1c3245946dd0505f234d1--


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Date: Tue, 21 Oct 2014 19:40:46 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Peter Thatcher <pthatcher@google.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP and ssrc-group,
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How is that? Where is that specified? What about if I include 3 ssrc values
in the ssrc-group? What is each one for?
On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:

> 345259865 is "real"
> 2693756249 is rtx
>
> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>
>> Hi,
>>
>> May I know which SSRC (345259865 or 2693756249) will be used for the
>> real media stream (plus RED and FEC) and which SSRC will be used for
>> RTX?
>>
>>
>>
>> --------------------------
>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>> a=3Dmid:video
>> a=3Drtpmap:100 VP8/90000
>> a=3Drtpmap:116 red/90000
>> a=3Drtpmap:117 ulpfec/90000
>> a=3Drtpmap:96 rtx/90000
>> a=3Dfmtp:96 apt=3D100
>> a=3Dssrc-group:FID 345259865 2693756249
>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>> -------------------------------
>>
>>
>>
>>
>> RFC 5576 does not clarify it at all:
>>
>> http://tools.ietf.org/html/rfc5576#section-4.2
>>
>> --------------------------------------------------
>> 4.2.  The "ssrc-group" Media Attribute
>>
>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>
>>    [..]
>>
>>    The <semantics> parameter is taken from the specification of the
>>    "group" attribute [RFC3388].  The initial semantic values defined for
>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>    relationship among the grouped sources is the same as the
>>    relationship among corresponding sources in media streams grouped
>>    using the SDP "group" attribute.
>> --------------------------------------------------
>>
>>
>>
>> The referenced RFC 3388 neither clarifies it:
>>
>> ---------------------------------------------------
>> 7.4 FID Semantics
>>
>>    Several "m" lines grouped together using FID semantics form a media
>>    flow.  A media agent handling a media flow that comprises several "m"
>>    lines MUST send a copy of the media to every "m" line part of the
>>    flow as long as the codecs and the direction attribute present in a
>>    particular "m" line allow it.
>>
>>    It is assumed that the application uses only one codec at a time to
>>    encode the media produced.  This codec MAY change dynamically during
>>    the session, but at any particular moment only one codec is in use.
>>
>>    The application encodes the media using the current codec and checks
>>    one by one all the "m" lines that are part of the flow.  If a
>>    particular "m" line contains the codec being used and the direction
>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media is
>>    sent to the address/port specified in that particular media stream.
>>    If either the "m" line does not contain the codec being used or the
>>    direction attribute is neither "sendonly" nor "sendrecv", nothing is
>>    sent over this media stream.
>> ----------------------------------------------------
>>
>>
>>
>>
>> So, how is the usage of ssrc-group? Where is it really defined?
>>
>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>
>> How should the receiver interpret it?
>>
>> Does a ssrc-group always mean RTX usage? Where is that specified in
>> the above SDP?
>>
>> Does not the above SDP look a complete mixture of hacks and workarounds?
>>
>>
>>
>>
>> --
>> I=C3=B1aki Baz Castillo
>> <ibc@aliax.net>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>

--047d7b624e72c76d2e0505f25493
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">How is that? Where is that specified? What about if I includ=
e 3 ssrc values in the ssrc-group? What is each one for?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; w=
rote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"lt=
r"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif"><span style=3D"font-family:arial,sans-serif;font-size:12.727272033691=
4px">345259865 is &quot;real&quot;</span><br></div><div class=3D"gmail_defa=
ult"><font face=3D"arial, sans-serif">2693756249 is rtx</font><br></div></d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Oct 21=
, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D=
"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 2693756249<br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
=C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the specificatio=
n of the<br>
=C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initial seman=
tic values defined for<br>
=C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Identificat=
ion) [RFC3388]<br>
=C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In each ca=
se, the<br>
=C2=A0 =C2=A0relationship among the grouped sources is the same as the<br>
=C2=A0 =C2=A0relationship among corresponding sources in media streams grou=
ped<br>
=C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID semanti=
cs form a media<br>
=C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that comprises=
 several &quot;m&quot;<br>
=C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&quot; lin=
e part of the<br>
=C2=A0 =C2=A0flow as long as the codecs and the direction attribute present=
 in a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0It is assumed that the application uses only one codec at a ti=
me to<br>
=C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change dynamic=
ally during<br>
=C2=A0 =C2=A0the session, but at any particular moment only one codec is in=
 use.<br>
<br>
=C2=A0 =C2=A0The application encodes the media using the current codec and =
checks<br>
=C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of the fl=
ow.=C2=A0 If a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being used an=
d the direction<br>
=C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quot;, a c=
opy of the encoded media is<br>
=C2=A0 =C2=A0sent to the address/port specified in that particular media st=
ream.<br>
=C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the codec be=
ing used or the<br>
=C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor &quot;=
sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<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>
</font></span></blockquote></div><br></div>
</blockquote></div>

--047d7b624e72c76d2e0505f25493--


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From: Peter Thatcher <pthatcher@google.com>
Date: Tue, 21 Oct 2014 10:57:09 -0700
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Subject: Re: [rtcweb] SDP and ssrc-group,
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I'm not sure where it's specified, but here's where it's implemented:

https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webr=
tcvideoengine.cc#4108

It only supports 2 SSRCs for a FID group.  It would ignore any more than
that.

On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:

> How is that? Where is that specified? What about if I include 3 ssrc
> values in the ssrc-group? What is each one for?
> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>
>> 345259865 is "real"
>> 2693756249 is rtx
>>
>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net>
>> wrote:
>>
>>> Hi,
>>>
>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>> RTX?
>>>
>>>
>>>
>>> --------------------------
>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>> a=3Dmid:video
>>> a=3Drtpmap:100 VP8/90000
>>> a=3Drtpmap:116 red/90000
>>> a=3Drtpmap:117 ulpfec/90000
>>> a=3Drtpmap:96 rtx/90000
>>> a=3Dfmtp:96 apt=3D100
>>> a=3Dssrc-group:FID 345259865 2693756249
>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> -------------------------------
>>>
>>>
>>>
>>>
>>> RFC 5576 does not clarify it at all:
>>>
>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>
>>> --------------------------------------------------
>>> 4.2.  The "ssrc-group" Media Attribute
>>>
>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>
>>>    [..]
>>>
>>>    The <semantics> parameter is taken from the specification of the
>>>    "group" attribute [RFC3388].  The initial semantic values defined fo=
r
>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>    relationship among the grouped sources is the same as the
>>>    relationship among corresponding sources in media streams grouped
>>>    using the SDP "group" attribute.
>>> --------------------------------------------------
>>>
>>>
>>>
>>> The referenced RFC 3388 neither clarifies it:
>>>
>>> ---------------------------------------------------
>>> 7.4 FID Semantics
>>>
>>>    Several "m" lines grouped together using FID semantics form a media
>>>    flow.  A media agent handling a media flow that comprises several "m=
"
>>>    lines MUST send a copy of the media to every "m" line part of the
>>>    flow as long as the codecs and the direction attribute present in a
>>>    particular "m" line allow it.
>>>
>>>    It is assumed that the application uses only one codec at a time to
>>>    encode the media produced.  This codec MAY change dynamically during
>>>    the session, but at any particular moment only one codec is in use.
>>>
>>>    The application encodes the media using the current codec and checks
>>>    one by one all the "m" lines that are part of the flow.  If a
>>>    particular "m" line contains the codec being used and the direction
>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media i=
s
>>>    sent to the address/port specified in that particular media stream.
>>>    If either the "m" line does not contain the codec being used or the
>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing is
>>>    sent over this media stream.
>>> ----------------------------------------------------
>>>
>>>
>>>
>>>
>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>
>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>
>>> How should the receiver interpret it?
>>>
>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>> the above SDP?
>>>
>>> Does not the above SDP look a complete mixture of hacks and workarounds=
?
>>>
>>>
>>>
>>>
>>> --
>>> I=C3=B1aki Baz Castillo
>>> <ibc@aliax.net>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>

--001a11c34bced2842a0505f29196
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I&#39;m not sure where it&#39;s specified, but here&#39=
;s where it&#39;s implemented:</div><div class=3D"gmail_default" style=3D"f=
ont-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_defaul=
t" style><font face=3D"arial, helvetica, sans-serif"><a href=3D"https://cod=
e.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengi=
ne.cc#4108">https://code.google.com/p/webrtc/source/browse/trunk/talk/media=
/webrtc/webrtcvideoengine.cc#4108</a></font><br></div><div class=3D"gmail_d=
efault" style><font face=3D"arial, helvetica, sans-serif"><br></font></div>=
<div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-ser=
if">It only supports 2 SSRCs for a FID group.=C2=A0 It would ignore any mor=
e than that.</font></div></div><div class=3D"gmail_extra"><br><div class=3D=
"gmail_quote">On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@a=
liax.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"><p dir=
=3D"ltr">How is that? Where is that specified? What about if I include 3 ss=
rc values in the ssrc-group? What is each one for?</p><div class=3D"HOEnZb"=
><div class=3D"h5">
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@go=
ogle.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif"><span style=3D"font-family:arial,sans-serif;font-si=
ze:12.7272720336914px">345259865 is &quot;real&quot;</span><br></div><div c=
lass=3D"gmail_default"><font face=3D"arial, sans-serif"><a href=3D"tel:2693=
756249" value=3D"+12693756249" target=3D"_blank">2693756249</a> is rtx</fon=
t><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+126937562=
49" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
=C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the specificatio=
n of the<br>
=C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initial seman=
tic values defined for<br>
=C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Identificat=
ion) [RFC3388]<br>
=C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In each ca=
se, the<br>
=C2=A0 =C2=A0relationship among the grouped sources is the same as the<br>
=C2=A0 =C2=A0relationship among corresponding sources in media streams grou=
ped<br>
=C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID semanti=
cs form a media<br>
=C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that comprises=
 several &quot;m&quot;<br>
=C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&quot; lin=
e part of the<br>
=C2=A0 =C2=A0flow as long as the codecs and the direction attribute present=
 in a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0It is assumed that the application uses only one codec at a ti=
me to<br>
=C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change dynamic=
ally during<br>
=C2=A0 =C2=A0the session, but at any particular moment only one codec is in=
 use.<br>
<br>
=C2=A0 =C2=A0The application encodes the media using the current codec and =
checks<br>
=C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of the fl=
ow.=C2=A0 If a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being used an=
d the direction<br>
=C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quot;, a c=
opy of the encoded media is<br>
=C2=A0 =C2=A0sent to the address/port specified in that particular media st=
ream.<br>
=C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the codec be=
ing used or the<br>
=C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor &quot;=
sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<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>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></div></blockquote></div><br></div>

--001a11c34bced2842a0505f29196--


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Subject: Re: [rtcweb] SDP and ssrc-group,
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Looking at the SDP, the Payload Type seems like the clearest way to differen=
tiate.  As it is, SSRC does not appear linked to the potential payloads.



> On Oct 21, 2014, at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrot=
e:
>=20
> How is that? Where is that specified? What about if I include 3 ssrc value=
s in the ssrc-group? What is each one for?
>=20
>> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>> 345259865 is "real"
>> 2693756249 is rtx
>>=20
>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net>=
 wrote:
>>> Hi,
>>>=20
>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>> RTX?
>>>=20
>>>=20
>>>=20
>>> --------------------------
>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>> a=3Dmid:video
>>> a=3Drtpmap:100 VP8/90000
>>> a=3Drtpmap:116 red/90000
>>> a=3Drtpmap:117 ulpfec/90000
>>> a=3Drtpmap:96 rtx/90000
>>> a=3Dfmtp:96 apt=3D100
>>> a=3Dssrc-group:FID 345259865 2693756249
>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>> -------------------------------
>>>=20
>>>=20
>>>=20
>>>=20
>>> RFC 5576 does not clarify it at all:
>>>=20
>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>=20
>>> --------------------------------------------------
>>> 4.2.  The "ssrc-group" Media Attribute
>>>=20
>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>=20
>>>    [..]
>>>=20
>>>    The <semantics> parameter is taken from the specification of the
>>>    "group" attribute [RFC3388].  The initial semantic values defined for=

>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>    relationship among the grouped sources is the same as the
>>>    relationship among corresponding sources in media streams grouped
>>>    using the SDP "group" attribute.
>>> --------------------------------------------------
>>>=20
>>>=20
>>>=20
>>> The referenced RFC 3388 neither clarifies it:
>>>=20
>>> ---------------------------------------------------
>>> 7.4 FID Semantics
>>>=20
>>>    Several "m" lines grouped together using FID semantics form a media
>>>    flow.  A media agent handling a media flow that comprises several "m"=

>>>    lines MUST send a copy of the media to every "m" line part of the
>>>    flow as long as the codecs and the direction attribute present in a
>>>    particular "m" line allow it.
>>>=20
>>>    It is assumed that the application uses only one codec at a time to
>>>    encode the media produced.  This codec MAY change dynamically during
>>>    the session, but at any particular moment only one codec is in use.
>>>=20
>>>    The application encodes the media using the current codec and checks
>>>    one by one all the "m" lines that are part of the flow.  If a
>>>    particular "m" line contains the codec being used and the direction
>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media is=

>>>    sent to the address/port specified in that particular media stream.
>>>    If either the "m" line does not contain the codec being used or the
>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing is
>>>    sent over this media stream.
>>> ----------------------------------------------------
>>>=20
>>>=20
>>>=20
>>>=20
>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>=20
>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>=20
>>> How should the receiver interpret it?
>>>=20
>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>> the above SDP?
>>>=20
>>> Does not the above SDP look a complete mixture of hacks and workarounds?=

>>>=20
>>>=20
>>>=20
>>>=20
>>> --
>>> I=C3=B1aki Baz Castillo
>>> <ibc@aliax.net>
>>>=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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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>Looking at the SDP, the Payload Type s=
eems like the clearest way to differentiate. &nbsp;As it is, SSRC does not a=
ppear linked to the potential payloads.<br><br><br></div><div><br>On Oct 21,=
 2014, at 10:40 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.=
net">ibc@aliax.net</a>&gt; wrote:<br><br></div><blockquote type=3D"cite"><di=
v><p dir=3D"ltr">How is that? Where is that specified? What about if I inclu=
de 3 ssrc values in the ssrc-group? What is each one for?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, "Peter Thatcher" &lt;<a hre=
f=3D"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; wrote:<br typ=
e=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><span style=
=3D"font-family:arial,sans-serif;font-size:12.7272720336914px">345259865 is "=
real"</span><br></div><div class=3D"gmail_default"><font face=3D"arial, sans=
-serif">2693756249 is rtx</font><br></div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Ba=
z Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"=
_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+126=
93756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 2693756249<br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank"=
>http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.&nbsp; The "ssrc-group" Media Attribute<br>
<br>
&nbsp; &nbsp;a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
&nbsp; &nbsp;[..]<br>
<br>
&nbsp; &nbsp;The &lt;semantics&gt; parameter is taken from the specification=
 of the<br>
&nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The initial semantic values d=
efined for<br>
&nbsp; &nbsp;the "ssrc-group" attribute are FID (Flow Identification) [RFC33=
88]<br>
&nbsp; &nbsp;and FEC (Forward Error Correction) [RFC4756].&nbsp; In each cas=
e, the<br>
&nbsp; &nbsp;relationship among the grouped sources is the same as the<br>
&nbsp; &nbsp;relationship among corresponding sources in media streams group=
ed<br>
&nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
&nbsp; &nbsp;Several "m" lines grouped together using FID semantics form a m=
edia<br>
&nbsp; &nbsp;flow.&nbsp; A media agent handling a media flow that comprises s=
everal "m"<br>
&nbsp; &nbsp;lines MUST send a copy of the media to every "m" line part of t=
he<br>
&nbsp; &nbsp;flow as long as the codecs and the direction attribute present i=
n a<br>
&nbsp; &nbsp;particular "m" line allow it.<br>
<br>
&nbsp; &nbsp;It is assumed that the application uses only one codec at a tim=
e to<br>
&nbsp; &nbsp;encode the media produced.&nbsp; This codec MAY change dynamica=
lly during<br>
&nbsp; &nbsp;the session, but at any particular moment only one codec is in u=
se.<br>
<br>
&nbsp; &nbsp;The application encodes the media using the current codec and c=
hecks<br>
&nbsp; &nbsp;one by one all the "m" lines that are part of the flow.&nbsp; I=
f a<br>
&nbsp; &nbsp;particular "m" line contains the codec being used and the direc=
tion<br>
&nbsp; &nbsp;attribute is "sendonly" or "sendrecv", a copy of the encoded me=
dia is<br>
&nbsp; &nbsp;sent to the address/port specified in that particular media str=
eam.<br>
&nbsp; &nbsp;If either the "m" line does not contain the codec being used or=
 the<br>
&nbsp; &nbsp;direction attribute is neither "sendonly" nor "sendrecv", nothi=
ng is<br>
&nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br>=

<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
<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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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Subject: Re: [rtcweb] SDP and ssrc-group,
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Hi Peter,

This is the rtcweb list, so while I appreciate the info about how Chrome 
implements it, in regards to this discussion it is irrelevant. As far as 
I know there is no spec that allow us to know just by looking at the SDP 
which ssrc belongs to each stream. Please, correct me if I am wrong.

Best regards
Sergio
On 21/10/2014 19:57, Peter Thatcher wrote:
> I'm not sure where it's specified, but here's where it's implemented:
>
> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108
>
> It only supports 2 SSRCs for a FID group.  It would ignore any more 
> than that.
>
> On Tue, Oct 21, 2014 at 10:40 AM, Iñaki Baz Castillo <ibc@aliax.net 
> <mailto:ibc@aliax.net>> wrote:
>
>     How is that? Where is that specified? What about if I include 3
>     ssrc values in the ssrc-group? What is each one for?
>
>     On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com
>     <mailto:pthatcher@google.com>> wrote:
>
>         345259865 is "real"
>         2693756249 <tel:2693756249> is rtx
>
>         On Tue, Oct 21, 2014 at 9:25 AM, Iñaki Baz Castillo
>         <ibc@aliax.net <mailto:ibc@aliax.net>> wrote:
>
>             Hi,
>
>             May I know which SSRC (345259865 or 2693756249
>             <tel:2693756249>) will be used for the
>             real media stream (plus RED and FEC) and which SSRC will
>             be used for
>             RTX?
>
>
>
>             --------------------------
>             m=video 62164 RTP/SAVPF 100 116 117 96
>             a=mid:video
>             a=rtpmap:100 VP8/90000
>             a=rtpmap:116 red/90000
>             a=rtpmap:117 ulpfec/90000
>             a=rtpmap:96 rtx/90000
>             a=fmtp:96 apt=100
>             a=ssrc-group:FID 345259865 2693756249 <tel:2693756249>
>             a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>             a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>             c0134f05-e7c2-4afd-a979-4e224de5eb91
>             a=ssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>             a=ssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>             a=ssrc:2693756249 <tel:2693756249> cname:erS7E/KHLYKTejNs
>             a=ssrc:2693756249 <tel:2693756249>
>             msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>             c0134f05-e7c2-4afd-a979-4e224de5eb91
>             a=ssrc:2693756249 <tel:2693756249>
>             mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>             a=ssrc:2693756249 <tel:2693756249>
>             label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>             -------------------------------
>
>
>
>
>             RFC 5576 does not clarify it at all:
>
>             http://tools.ietf.org/html/rfc5576#section-4.2
>
>             --------------------------------------------------
>             4.2.  The "ssrc-group" Media Attribute
>
>                a=ssrc-group:<semantics> <ssrc-id> ...
>
>                [..]
>
>                The <semantics> parameter is taken from the
>             specification of the
>                "group" attribute [RFC3388].  The initial semantic
>             values defined for
>                the "ssrc-group" attribute are FID (Flow
>             Identification) [RFC3388]
>                and FEC (Forward Error Correction) [RFC4756].  In each
>             case, the
>                relationship among the grouped sources is the same as the
>                relationship among corresponding sources in media
>             streams grouped
>                using the SDP "group" attribute.
>             --------------------------------------------------
>
>
>
>             The referenced RFC 3388 neither clarifies it:
>
>             ---------------------------------------------------
>             7.4 FID Semantics
>
>                Several "m" lines grouped together using FID semantics
>             form a media
>                flow.  A media agent handling a media flow that
>             comprises several "m"
>                lines MUST send a copy of the media to every "m" line
>             part of the
>                flow as long as the codecs and the direction attribute
>             present in a
>                particular "m" line allow it.
>
>                It is assumed that the application uses only one codec
>             at a time to
>                encode the media produced.  This codec MAY change
>             dynamically during
>                the session, but at any particular moment only one
>             codec is in use.
>
>                The application encodes the media using the current
>             codec and checks
>                one by one all the "m" lines that are part of the
>             flow.  If a
>                particular "m" line contains the codec being used and
>             the direction
>                attribute is "sendonly" or "sendrecv", a copy of the
>             encoded media is
>                sent to the address/port specified in that particular
>             media stream.
>                If either the "m" line does not contain the codec being
>             used or the
>                direction attribute is neither "sendonly" nor
>             "sendrecv", nothing is
>                sent over this media stream.
>             ----------------------------------------------------
>
>
>
>
>             So, how is the usage of ssrc-group? Where is it really
>             defined?
>
>             Can I put more than 2 ssrc together in the same ssrc-group
>             line?
>
>             How should the receiver interpret it?
>
>             Does a ssrc-group always mean RTX usage? Where is that
>             specified in
>             the above SDP?
>
>             Does not the above SDP look a complete mixture of hacks
>             and workarounds?
>
>
>
>
>             --
>             Iñaki Baz Castillo
>             <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>             _______________________________________________
>             rtcweb mailing list
>             rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>             https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------060203090304080301040400
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    <div class="moz-cite-prefix">Hi Peter,<br>
      <br>
      This is the rtcweb list, so while I appreciate the info about how
      Chrome implements it, in regards to this discussion it is
      irrelevant. As far as I know there is no spec that allow us to
      know just by looking at the SDP which ssrc belongs to each stream.
      Please, correct me if I am wrong.<br>
      <br>
      Best regards<br>
      Sergio<br>
      On 21/10/2014 19:57, Peter Thatcher wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrXDUF-AXcDuQmYhH91vbgPAxLkYkB==GY9opoRk7zrEP8A7A@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif">I'm not sure
          where it's specified, but here's where it's implemented:</div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class="gmail_default" style=""><font face="arial,
            helvetica, sans-serif"><a moz-do-not-send="true"
href="https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108">https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108</a></font><br>
        </div>
        <div class="gmail_default" style=""><font face="arial,
            helvetica, sans-serif"><br>
          </font></div>
        <div class="gmail_default" style=""><font face="arial,
            helvetica, sans-serif">It only supports 2 SSRCs for a FID
            group.&nbsp; It would ignore any more than that.</font></div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Tue, Oct 21, 2014 at 10:40 AM, I&ntilde;aki
          Baz Castillo <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <p dir="ltr">How is that? Where is that specified? What
              about if I include 3 ssrc values in the ssrc-group? What
              is each one for?</p>
            <div class="HOEnZb">
              <div class="h5">
                <div class="gmail_quote">On 21 Oct 2014 19:31, "Peter
                  Thatcher" &lt;<a moz-do-not-send="true"
                    href="mailto:pthatcher@google.com" target="_blank">pthatcher@google.com</a>&gt;
                  wrote:<br type="attribution">
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    <div dir="ltr">
                      <div class="gmail_default"
                        style="font-family:arial,helvetica,sans-serif"><span
style="font-family:arial,sans-serif;font-size:12.7272720336914px">345259865
                          is "real"</span><br>
                      </div>
                      <div class="gmail_default"><font face="arial,
                          sans-serif"><a moz-do-not-send="true"
                            href="tel:2693756249" value="+12693756249"
                            target="_blank">2693756249</a> is rtx</font><br>
                      </div>
                    </div>
                    <div class="gmail_extra"><br>
                      <div class="gmail_quote">On Tue, Oct 21, 2014 at
                        9:25 AM, I&ntilde;aki Baz Castillo <span dir="ltr">&lt;<a
                            moz-do-not-send="true"
                            href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;</span>
                        wrote:<br>
                        <blockquote class="gmail_quote" style="margin:0
                          0 0 .8ex;border-left:1px #ccc
                          solid;padding-left:1ex">Hi,<br>
                          <br>
                          May I know which SSRC (345259865 or <a
                            moz-do-not-send="true" href="tel:2693756249"
                            value="+12693756249" target="_blank">2693756249</a>)
                          will be used for the<br>
                          real media stream (plus RED and FEC) and which
                          SSRC will be used for<br>
                          RTX?<br>
                          <br>
                          <br>
                          <br>
                          --------------------------<br>
                          m=video 62164 RTP/SAVPF 100 116 117 96<br>
                          a=mid:video<br>
                          a=rtpmap:100 VP8/90000<br>
                          a=rtpmap:116 red/90000<br>
                          a=rtpmap:117 ulpfec/90000<br>
                          a=rtpmap:96 rtx/90000<br>
                          a=fmtp:96 apt=100<br>
                          a=ssrc-group:FID 345259865 <a
                            moz-do-not-send="true" href="tel:2693756249"
                            value="+12693756249" target="_blank">2693756249</a><br>
                          a=ssrc:345259865 cname:erS7E/KHLYKTejNs<br>
                          a=ssrc:345259865
                          msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                          c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                          a=ssrc:345259865
                          mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                          a=ssrc:345259865
                          <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
                          a=ssrc:<a moz-do-not-send="true"
                            href="tel:2693756249" value="+12693756249"
                            target="_blank">2693756249</a>
                          cname:erS7E/KHLYKTejNs<br>
                          a=ssrc:<a moz-do-not-send="true"
                            href="tel:2693756249" value="+12693756249"
                            target="_blank">2693756249</a>
                          msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                          c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                          a=ssrc:<a moz-do-not-send="true"
                            href="tel:2693756249" value="+12693756249"
                            target="_blank">2693756249</a>
                          mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                          a=ssrc:<a moz-do-not-send="true"
                            href="tel:2693756249" value="+12693756249"
                            target="_blank">2693756249</a>
                          <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
                          -------------------------------<br>
                          <br>
                          <br>
                          <br>
                          <br>
                          RFC 5576 does not clarify it at all:<br>
                          <br>
                          <a moz-do-not-send="true"
                            href="http://tools.ietf.org/html/rfc5576#section-4.2"
                            target="_blank">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
                          <br>
--------------------------------------------------<br>
                          4.2.&nbsp; The "ssrc-group" Media Attribute<br>
                          <br>
                          &nbsp; &nbsp;a=ssrc-group:&lt;semantics&gt;
                          &lt;ssrc-id&gt; ...<br>
                          <br>
                          &nbsp; &nbsp;[..]<br>
                          <br>
                          &nbsp; &nbsp;The &lt;semantics&gt; parameter is taken
                          from the specification of the<br>
                          &nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The initial
                          semantic values defined for<br>
                          &nbsp; &nbsp;the "ssrc-group" attribute are FID (Flow
                          Identification) [RFC3388]<br>
                          &nbsp; &nbsp;and FEC (Forward Error Correction)
                          [RFC4756].&nbsp; In each case, the<br>
                          &nbsp; &nbsp;relationship among the grouped sources is
                          the same as the<br>
                          &nbsp; &nbsp;relationship among corresponding sources in
                          media streams grouped<br>
                          &nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
                          <br>
                          <br>
                          <br>
                          The referenced RFC 3388 neither clarifies it:<br>
                          <br>
---------------------------------------------------<br>
                          7.4 FID Semantics<br>
                          <br>
                          &nbsp; &nbsp;Several "m" lines grouped together using
                          FID semantics form a media<br>
                          &nbsp; &nbsp;flow.&nbsp; A media agent handling a media flow
                          that comprises several "m"<br>
                          &nbsp; &nbsp;lines MUST send a copy of the media to
                          every "m" line part of the<br>
                          &nbsp; &nbsp;flow as long as the codecs and the
                          direction attribute present in a<br>
                          &nbsp; &nbsp;particular "m" line allow it.<br>
                          <br>
                          &nbsp; &nbsp;It is assumed that the application uses
                          only one codec at a time to<br>
                          &nbsp; &nbsp;encode the media produced.&nbsp; This codec MAY
                          change dynamically during<br>
                          &nbsp; &nbsp;the session, but at any particular moment
                          only one codec is in use.<br>
                          <br>
                          &nbsp; &nbsp;The application encodes the media using the
                          current codec and checks<br>
                          &nbsp; &nbsp;one by one all the "m" lines that are part
                          of the flow.&nbsp; If a<br>
                          &nbsp; &nbsp;particular "m" line contains the codec
                          being used and the direction<br>
                          &nbsp; &nbsp;attribute is "sendonly" or "sendrecv", a
                          copy of the encoded media is<br>
                          &nbsp; &nbsp;sent to the address/port specified in that
                          particular media stream.<br>
                          &nbsp; &nbsp;If either the "m" line does not contain the
                          codec being used or the<br>
                          &nbsp; &nbsp;direction attribute is neither "sendonly"
                          nor "sendrecv", nothing is<br>
                          &nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
                          <br>
                          <br>
                          <br>
                          <br>
                          So, how is the usage of ssrc-group? Where is
                          it really defined?<br>
                          <br>
                          Can I put more than 2 ssrc together in the
                          same ssrc-group line?<br>
                          <br>
                          How should the receiver interpret it?<br>
                          <br>
                          Does a ssrc-group always mean RTX usage? Where
                          is that specified in<br>
                          the above SDP?<br>
                          <br>
                          Does not the above SDP look a complete mixture
                          of hacks and workarounds?<br>
                          <span><font color="#888888"><br>
                              <br>
                              <br>
                              <br>
                              --<br>
                              I&ntilde;aki Baz Castillo<br>
                              &lt;<a moz-do-not-send="true"
                                href="mailto:ibc@aliax.net"
                                target="_blank">ibc@aliax.net</a>&gt;<br>
                              <br>
_______________________________________________<br>
                              rtcweb mailing list<br>
                              <a moz-do-not-send="true"
                                href="mailto:rtcweb@ietf.org"
                                target="_blank">rtcweb@ietf.org</a><br>
                              <a moz-do-not-send="true"
                                href="https://www.ietf.org/mailman/listinfo/rtcweb"
                                target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                            </font></span></blockquote>
                      </div>
                      <br>
                    </div>
                  </blockquote>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
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I hope I can rule my SDP logic based on standards rather than on how Chrome
implements some features.

My question was generic: if I receive the above SDP, how do I know which
payloads each ssrc is supposed to transport?
On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:

> I'm not sure where it's specified, but here's where it's implemented:
>
>
> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/we=
brtcvideoengine.cc#4108
>
> It only supports 2 SSRCs for a FID group.  It would ignore any more than
> that.
>
> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net>
> wrote:
>
>> How is that? Where is that specified? What about if I include 3 ssrc
>> values in the ssrc-group? What is each one for?
>> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>>
>>> 345259865 is "real"
>>> 2693756249 is rtx
>>>
>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net=
>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>>> RTX?
>>>>
>>>>
>>>>
>>>> --------------------------
>>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>>> a=3Dmid:video
>>>> a=3Drtpmap:100 VP8/90000
>>>> a=3Drtpmap:116 red/90000
>>>> a=3Drtpmap:117 ulpfec/90000
>>>> a=3Drtpmap:96 rtx/90000
>>>> a=3Dfmtp:96 apt=3D100
>>>> a=3Dssrc-group:FID 345259865 2693756249
>>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> -------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> RFC 5576 does not clarify it at all:
>>>>
>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>
>>>> --------------------------------------------------
>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>
>>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>>
>>>>    [..]
>>>>
>>>>    The <semantics> parameter is taken from the specification of the
>>>>    "group" attribute [RFC3388].  The initial semantic values defined f=
or
>>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>    relationship among the grouped sources is the same as the
>>>>    relationship among corresponding sources in media streams grouped
>>>>    using the SDP "group" attribute.
>>>> --------------------------------------------------
>>>>
>>>>
>>>>
>>>> The referenced RFC 3388 neither clarifies it:
>>>>
>>>> ---------------------------------------------------
>>>> 7.4 FID Semantics
>>>>
>>>>    Several "m" lines grouped together using FID semantics form a media
>>>>    flow.  A media agent handling a media flow that comprises several "=
m"
>>>>    lines MUST send a copy of the media to every "m" line part of the
>>>>    flow as long as the codecs and the direction attribute present in a
>>>>    particular "m" line allow it.
>>>>
>>>>    It is assumed that the application uses only one codec at a time to
>>>>    encode the media produced.  This codec MAY change dynamically durin=
g
>>>>    the session, but at any particular moment only one codec is in use.
>>>>
>>>>    The application encodes the media using the current codec and check=
s
>>>>    one by one all the "m" lines that are part of the flow.  If a
>>>>    particular "m" line contains the codec being used and the direction
>>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media =
is
>>>>    sent to the address/port specified in that particular media stream.
>>>>    If either the "m" line does not contain the codec being used or the
>>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing i=
s
>>>>    sent over this media stream.
>>>> ----------------------------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>
>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>
>>>> How should the receiver interpret it?
>>>>
>>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>>> the above SDP?
>>>>
>>>> Does not the above SDP look a complete mixture of hacks and workaround=
s?
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> I=C3=B1aki Baz Castillo
>>>> <ibc@aliax.net>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>>
>

--001a1134a5ca4df8110505f31b86
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">I hope I can rule my SDP logic based on standards rather tha=
n on how Chrome implements some features.</p>
<p dir=3D"ltr">My question was generic: if I receive the above SDP, how do =
I know which payloads each ssrc is supposed to transport?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:57, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; w=
rote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"lt=
r"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif">I&#39;m not sure where it&#39;s specified, but here&#39;s where it&#3=
9;s implemented:</div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif"><br></div><div class=3D"gmail_default"><font face=
=3D"arial, helvetica, sans-serif"><a href=3D"https://code.google.com/p/webr=
tc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108" target=
=3D"_blank">https://code.google.com/p/webrtc/source/browse/trunk/talk/media=
/webrtc/webrtcvideoengine.cc#4108</a></font><br></div><div class=3D"gmail_d=
efault"><font face=3D"arial, helvetica, sans-serif"><br></font></div><div c=
lass=3D"gmail_default"><font face=3D"arial, helvetica, sans-serif">It only =
supports 2 SSRCs for a FID group.=C2=A0 It would ignore any more than that.=
</font></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <span dir=3D"lt=
r">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir=3D"ltr">How is =
that? Where is that specified? What about if I include 3 ssrc values in the=
 ssrc-group? What is each one for?</p><div><div>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@go=
ogle.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif"><span style=3D"font-family:arial,sans-serif;font-si=
ze:12.7272720336914px">345259865 is &quot;real&quot;</span><br></div><div c=
lass=3D"gmail_default"><font face=3D"arial, sans-serif"><a href=3D"tel:2693=
756249" value=3D"+12693756249" target=3D"_blank">2693756249</a> is rtx</fon=
t><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+126937562=
49" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
=C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the specificatio=
n of the<br>
=C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initial seman=
tic values defined for<br>
=C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Identificat=
ion) [RFC3388]<br>
=C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In each ca=
se, the<br>
=C2=A0 =C2=A0relationship among the grouped sources is the same as the<br>
=C2=A0 =C2=A0relationship among corresponding sources in media streams grou=
ped<br>
=C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID semanti=
cs form a media<br>
=C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that comprises=
 several &quot;m&quot;<br>
=C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&quot; lin=
e part of the<br>
=C2=A0 =C2=A0flow as long as the codecs and the direction attribute present=
 in a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0It is assumed that the application uses only one codec at a ti=
me to<br>
=C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change dynamic=
ally during<br>
=C2=A0 =C2=A0the session, but at any particular moment only one codec is in=
 use.<br>
<br>
=C2=A0 =C2=A0The application encodes the media using the current codec and =
checks<br>
=C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of the fl=
ow.=C2=A0 If a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being used an=
d the direction<br>
=C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quot;, a c=
opy of the encoded media is<br>
=C2=A0 =C2=A0sent to the address/port specified in that particular media st=
ream.<br>
=C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the codec be=
ing used or the<br>
=C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor &quot;=
sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<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>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></div></blockquote></div><br></div>
</blockquote></div>

--001a1134a5ca4df8110505f31b86--


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Date: Tue, 21 Oct 2014 20:58:32 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/RGcS9my1Z6PCcRruAU1l9dZS-MU
Subject: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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Hi all,

Not sure if it is done on pourpose, but according to the RTP usage 
draft, it may seem that full RFC 4588 is mandated at the recevier side:

     Receivers are REQUIRED to implement support for RTP retransmission
     packets [RFC4588  <https://tools.ietf.org/html/rfc4588>].


That would include both modes, session and ssrc multiplexing. Given the 
extensive usage of bundle and current implementations, session 
multiplexing support doesn't make much sense.

Should we drop it, and state that only ssrc-multiplexing shall be 
supported at the receiving end?

Best regards
Sergio

--------------020003060701020508050205
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Hi all,<br>
    <br>
    Not sure if it is done on pourpose, but according to the RTP usage
    draft, it may seem that full RFC 4588 is mandated at the recevier
    side:<br>
    <br>
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are REQUIRED to implement support for RTP retransmission
   &nbsp;packets [<a href="https://tools.ietf.org/html/rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].</pre>
    <br>
    That would include both modes, session and ssrc multiplexing. Given
    the extensive usage of bundle and current implementations, session
    multiplexing support doesn't make much sense.<br>
    <br>
    Should we drop it, and state that only ssrc-multiplexing shall be
    supported at the receiving end?<br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------020003060701020508050205--


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Subject: Re: [rtcweb] SDP and ssrc-group,
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>
>
> Does not the above SDP look a complete mixture of hacks and workarounds?
>
>
Are you new here?

In all seriousness, it would be nice if this behavior was standardized, so
that applications could have deterministic behavior. If we can come up with
appropriate text about the ordering of SSRCs, this could be written into
JSEP's handling of setLocalDescription.

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span><font color=3D"#888888"><br></font></span></blockquote><div><br></div=
><div>Are you new here?=C2=A0</div><div><br></div><div>In all seriousness, =
it would be nice if this behavior was standardized, so that applications co=
uld have deterministic behavior. If we can come up with appropriate text ab=
out the ordering of SSRCs, this could be written into JSEP&#39;s handling o=
f setLocalDescription.</div></div></div></div>

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Subject: Re: [rtcweb] SDP and ssrc-group,
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On 21/10/2014 22:14, Makaraju, Maridi Raju (Raju) wrote:
>
> I hope I can rule my SDP logic based on standards rather than on how 
> Chrome implements some features.
>
> My question was generic: if I receive the above SDP, how do I know 
> which payloads each ssrc is supposed to transport?
>
> */<Raju >/*
>
> */Right. Firefox may have a different interpretation./*
>
> */If a=ssrc-group and retransmissions are negotiated at SDP answer, 
> then the order of ssrcs probably does not matter as the RTP payload 
> values can determine the retransmission vs. original payloads./*
>
> */If answerer wants to know the original ssrc for rejecting 
> a=ssrc-group and retransmission payload then it need to know the 
> original ssrc./*
>
> *//*
>
> */http://tools.ietf.org/html/rfc5576#section-6.3 defines a=ssrc with 
> format. In the example below,/*
>
> */won't having the following SDP line be sufficient to assign 
> 2693756249 <tel:2693756249> for retransmission and, indirectly, assign 
>  345259865 for original ./*
>
> */a=ssrc:2693756249 <tel:2693756249> fmtp:96 /*
>
> --------------------------
> m=video 62164 RTP/SAVPF 100 116 117 96
> a=mid:video
> a=rtpmap:100 VP8/90000
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> a=ssrc-group:FID 345259865 2693756249 <tel:2693756249>
> a=ssrc:345259865 cname:erS7E/KHLYKTejNs
> a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=ssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=ssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=ssrc:2693756249 <tel:2693756249> cname:erS7E/KHLYKTejNs
> a=ssrc:2693756249 <tel:2693756249> 
> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=ssrc:2693756249 <tel:2693756249> 
> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=ssrc:2693756249 <tel:2693756249> 
> label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> -------------------------------
>
> */On the question of what if there are 3 ssrcs?/*
>
> */Per my understanding, there is only one ssrc for original payload 
> and another for retransmission. So, a=ssrc-group containing 3 ssrcs is 
> probably invalid./*
>
> *//*
>
> */</Raju>/*
>
>
Hi Raju

I foresee that when we start working on FEC we will end up using a 
different ssrc to avoid asking retransmissions of FEC redundant data. So 
3 ssrcs will be needed for each media stream.

Regarding the ftmp ssrc attribute, syntactically it solves perfectly the 
problem (even if we have fec in its own ssrc), but I am not sure if 
semantically it is the intended usage, as it seems more to define format 
attributes for an specific ssrc an not to restrict the usage of the 
payload type to that ssrc.

Also, I am not sure how that would play together with plan c/plan b/plan 
c/no plan (I have not followed that thread actively).

Best regards
Sergio

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    <div class="moz-cite-prefix"><br>
      On 21/10/2014 22:14, Makaraju, Maridi Raju (Raju) wrote:<br>
    </div>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5EAE2A@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
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      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <p>I hope I can rule my SDP logic based on standards rather
            than on how Chrome implements some features.<o:p></o:p></p>
          <p>My question was generic: if I receive the above SDP, how do
            I know which payloads each ssrc is supposed to transport?<span
              style="color:#1F497D"><o:p></o:p></span></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju
                  &gt;<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Right.
                  Firefox may have a different interpretation.<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If
                  a=ssrc-group and retransmissions are negotiated at SDP
                  answer, then the order of ssrcs probably does not
                  matter as the RTP payload values can determine the
                  retransmission vs. original payloads.<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If
                  answerer wants to know the original ssrc for rejecting
                  a=ssrc-group and retransmission payload then it need
                  to know the original ssrc.<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><a
                    moz-do-not-send="true"
                    href="http://tools.ietf.org/html/rfc5576#section-6.3">http://tools.ietf.org/html/rfc5576#section-6.3</a>
                  defines a=ssrc with format. In the example below,<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">won&#8217;t
                  having the following SDP line be sufficient to assign
                  <a moz-do-not-send="true" href="tel:2693756249"
                    target="_blank"><span
                      style="color:#1F497D;text-decoration:none">2693756249</span></a>
                  for retransmission and, indirectly, assign &nbsp;345259865
                  for original .<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">a=ssrc:<a
                    moz-do-not-send="true" href="tel:2693756249"
                    target="_blank"><span
                      style="color:#1F497D;text-decoration:none">2693756249</span></a>
                  fmtp:96
                  <o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt">--------------------------<br>
            m=video 62164 RTP/SAVPF 100 116 117 96<br>
            a=mid:video<br>
            a=rtpmap:100 VP8/90000<br>
            a=rtpmap:116 red/90000<br>
            a=rtpmap:117 ulpfec/90000<br>
            a=rtpmap:96 rtx/90000<br>
            a=fmtp:96 apt=100<br>
            a=ssrc-group:FID 345259865 <a moz-do-not-send="true"
              href="tel:2693756249" target="_blank">2693756249</a><br>
            a=ssrc:345259865 cname:erS7E/KHLYKTejNs<br>
            a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
            c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
            a=ssrc:345259865
            mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
            a=ssrc:345259865 <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
            a=ssrc:<a moz-do-not-send="true" href="tel:2693756249"
              target="_blank">2693756249</a> cname:erS7E/KHLYKTejNs<br>
            a=ssrc:<a moz-do-not-send="true" href="tel:2693756249"
              target="_blank">2693756249</a>
            msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
            c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
            a=ssrc:<a moz-do-not-send="true" href="tel:2693756249"
              target="_blank">2693756249</a>
            mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
            a=ssrc:<a moz-do-not-send="true" href="tel:2693756249"
              target="_blank">2693756249</a>
            <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
            -------------------------------<br>
            <br>
            <b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">On
                  the question of what if there are 3 ssrcs?<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Per
                  my understanding, there is only one ssrc for original
                  payload and another for retransmission. So,
                  a=ssrc-group containing 3 ssrcs is probably invalid.<o:p></o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
          <p style="margin:0in;margin-bottom:.0001pt"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<o:p></o:p></span></i></b></p>
          <br>
        </div>
      </div>
    </blockquote>
    Hi Raju<br>
    <br>
    I foresee that when we start working on FEC we will end up using a
    different ssrc to avoid asking retransmissions of FEC redundant
    data. So 3 ssrcs will be needed for each media stream.<br>
    <br>
    Regarding the ftmp ssrc attribute, syntactically it solves perfectly
    the problem (even if we have fec in its own ssrc), but I am not sure
    if semantically it is the intended usage, as it seems more to define
    format attributes for an specific ssrc an not to restrict the usage
    of the payload type to that ssrc.&nbsp; <br>
    <br>
    Also, I am not sure how that would play together with plan c/plan
    b/plan c/no plan (I have not followed that thread actively).<br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------020009030004050604010503--


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Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> > Yes, that's the intended use-case: The middlebox is a trusted man-in-th=
e-
> middle.  In order to record and/or perform analytics on the media flowing
> through it, it obviously has to decrypt it.  That can only be done by
> terminating DTLS (and decrypting/re-encrypting SRTP.)
>=20
> If the middle box is trusted, only interested in media and known about by
> the application, then the application should create a separate direct dat=
a
> channel only DTLS connection between the endpoints that bypasses the
> middle box, with a small amount of extra application effort we remove the
> need for even more random extra SDP lines. Remember these are
> programmable endpoints - not dumb SIP devices with fixed behaviours.
>=20
> Unless the middle box is performing 'analytics' on the DC data too, but t=
hat
> isn't the use-case you mentioned.

Having to use one of the endpoints to fork the media for recording and anal=
ytics is not acceptable for us.  Many of our customers require that all int=
eractions are recorded exactly as they occur between the endpoints.  Often =
they are required to do that for legal or compliance reasons.  It's actuall=
y pretty certain that some of them will want the data streams recorded too.=
 =20

--Felix


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-18.txt
	Pages           : 44
	Date            : 2014-10-21

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-18

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-18


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

Internet-Drafts are also available by anonymous FTP at:
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From nobody Tue Oct 21 15:18:56 2014
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Date: Tue, 21 Oct 2014 18:18:25 -0400
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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On 10/21/2014 4:58 PM, Wyss, Felix wrote:
>>> Yes, that's the intended use-case: The middlebox is a trusted man-in-the-
>> middle.  In order to record and/or perform analytics on the media flowing
>> through it, it obviously has to decrypt it.  That can only be done by
>> terminating DTLS (and decrypting/re-encrypting SRTP.)
>>
>> If the middle box is trusted, only interested in media and known about by
>> the application, then the application should create a separate direct data
>> channel only DTLS connection between the endpoints that bypasses the
>> middle box, with a small amount of extra application effort we remove the
>> need for even more random extra SDP lines. Remember these are
>> programmable endpoints - not dumb SIP devices with fixed behaviours.
>>
>> Unless the middle box is performing 'analytics' on the DC data too, but that
>> isn't the use-case you mentioned.
> Having to use one of the endpoints to fork the media for recording and analytics is not acceptable for us.  Many of our customers require that all interactions are recorded exactly as they occur between the endpoints.  Often they are required to do that for legal or compliance reasons.  It's actually pretty certain that some of them will want the data streams recorded too.

I had a suspicion that this was the intended use-case - and it's a real 
use-case for certain applications/industries.

So, for those applications (remembering that webrtc applications are not 
random external devices, but basically code linked to the usage) just 
either use pre-defined or external negotiation for channel assignments, 
or implement *in the application* allocation-choosing strategy.  Such as:

middlebox: offer to A and B (actpass)
A: accept (active)
B: accept (active)
ok, now both are on the same "evenness"
both statically open (pre-defined) channel 0 always
Both send a random token value
The side with the lower random value is even, and the other is odd
All channel allocations are preceeded by message on channel 0 that 
amounts to an "I'm opening channel N with these params" that takes the 
place of DCEP
Encapsulate this behavior in some simple functions that take the place 
of the default createDataChannel()/etc

There are other solutions of course for glare resolution.  If you want, 
if the two sides chose the opposite evenness, you just leave 
createDataChannel alone.  There are a LOT of ways to deal with this at 
the application level, since the application "should" know it's talking 
to (or may be talking to) a middlebox that will decrypt/forward things.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Tue Oct 21 15:19:34 2014
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This version attempts to align the terminology with the latest =
rtcweb-overview draft. Aside from (hopefully) fixing the terminology, =
there are no other changes.

Colin



On 21 Oct 2014, at 18:13, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
>        Authors         : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-18.txt
> 	Pages           : 44
> 	Date            : 2014-10-21
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-18
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-18
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
https://csperkins.org/





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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-video Section 4.2 (H.264)
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On 14/10/14 00:58, Bernard Aboba wrote:=0A=
> This section needs more work.  Aside from requiring support for=0A=
> packetization mode 1, we could use additional text on robustness.=0A=
>=0A=
> Could aspects of existing H.264 robustness recommendations (such 3GPP=0A=
> S4-140961, designed for Video over LTE) could be incorporated?=0A=
=0A=
You mean the part saying more or less:=0A=
=0A=
* NACK messages mean lost frames=0A=
* PLI must be supported=0A=
* FIR must be supported=0A=
=0A=
To me that would make sense. But should it go into section 4.2 or section 5=
?=0A=
=0A=
Stefan=0A=
=0A=


From nobody Wed Oct 22 07:22:21 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-07.txt
	Pages           : 15
	Date            : 2014-10-22

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-07


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

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


From nobody Wed Oct 22 07:44:09 2014
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Section 3.4

" WebRTC browsers MUST support configuration of STUN and TURN servers,
   both from browser configuration and from an application."


Should we mention which takes precedence? If configured in the browser, =
start with that server.
If not use the one provided by the application.

Maybe we should clarify that turn DNS discovery MUST be used to provide =
failover and load balancing.

/O=


From nobody Wed Oct 22 07:51:39 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Olle E. Johansso=
n
Sent: 22 October 2014 17:44
To: rtcweb@ietf.org
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt

Section 3.4

" WebRTC browsers MUST support configuration of STUN and TURN servers,
   both from browser configuration and from an application."


Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.
If not use the one provided by the application.

Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.

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


From nobody Wed Oct 22 07:53:07 2014
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This is a multi-part message in MIME format.
--------------090809050209020800060208
Content-Type: text/plain; charset=windows-1252
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On 08/25/2014 07:36 AM, Justin Uberti wrote:
> Yes, 4571 must be used for ICE connectivity checks, when using ICE-TCP.=

>
> However, TURN/TCP has its own framing for STUN messages.
>
> I would simplify your sentence to:
>
>    If ICE-TCP connections are used, framing according to [RFC4571] MUST=

>    be used for all packets, unless the content has its own framing
> mechanism.

Checking the list for long-delayed updates....

Checking back: The change from TCP to ICE-TCP is good (this excludes
TURN-TCP from this paragraph), but the exchange leaves me uncertain
about the status of STUN packets on ICE-TCP connections.

Do they happen at all? If they happen, should they be RFC 4571
encapsulated, or not?

>
>
> On Fri, Aug 22, 2014 at 8:38 AM, I=F1aki Baz Castillo <ibc@aliax.net
> <mailto:ibc@aliax.net>> wrote:
>
>     Hi,
>
>     draft-ietf-rtcweb-transports-06 says:
>
>        If TCP connections are used, RTP framing according to [RFC4571]
>     MUST
>        be used, both for the RTP packets and for the DTLS packets used =
to
>        carry data channels.
>
>
>     Not just RTP and DTLS, but also for STUN packets must be framed wit=
h
>     RFC4571. And I do not understand why "DTLS packets used to carry da=
ta
>     channels". In fact all the DTLS packets (also the ones for the
>     handshake) must be framed as per RFC 4571.
>
>
>     So I suggest to improve the above text as follows:
>
>        If TCP connections are used, RTP framing according to [RFC4571]
>     MUST
>        be used for the STUN packets, SRTP packets, SRTCP packets and DT=
LS
>        packets.
>
>
>
>
>
>
>     --
>     I=F1aki Baz Castillo
>     <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Surveillance is pervasive. Go Dark.


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<html>
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    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
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    <div class="moz-cite-prefix">On 08/25/2014 07:36 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-1NgDGVahbhJeb=5m1bjTvYdqKPTXmiOWqEw_wev8TV_Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">Yes, 4571 must be used for ICE connectivity checks,
        when using ICE-TCP.
        <div><br>
        </div>
        <div>However, TURN/TCP has its own framing for STUN messages.</div>
        <div><br>
        </div>
        <div>I would simplify your sentence to:</div>
        <div><br>
        </div>
        <div><span style="font-family:arial,sans-serif;font-size:13px"> 
             If ICE-TCP connections are used, framing according to
            [RFC4571] MUST</span><br
            style="font-family:arial,sans-serif;font-size:13px">
          <span style="font-family:arial,sans-serif;font-size:13px"> 
             be used for all packets, unless the content has its own
            framing mechanism.</span><br>
        </div>
      </div>
    </blockquote>
    <br>
    Checking the list for long-delayed updates....<br>
    <br>
    Checking back: The change from TCP to ICE-TCP is good (this excludes
    TURN-TCP from this paragraph), but the exchange leaves me uncertain
    about the status of STUN packets on ICE-TCP connections.<br>
    <br>
    Do they happen at all? If they happen, should they be RFC 4571
    encapsulated, or not?<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-1NgDGVahbhJeb=5m1bjTvYdqKPTXmiOWqEw_wev8TV_Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Fri, Aug 22, 2014 at 8:38 AM, Iñaki
          Baz Castillo <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
            <br>
            draft-ietf-rtcweb-transports-06 says:<br>
            <br>
               If TCP connections are used, RTP framing according to
            [RFC4571] MUST<br>
               be used, both for the RTP packets and for the DTLS
            packets used to<br>
               carry data channels.<br>
            <br>
            <br>
            Not just RTP and DTLS, but also for STUN packets must be
            framed with<br>
            RFC4571. And I do not understand why "DTLS packets used to
            carry data<br>
            channels". In fact all the DTLS packets (also the ones for
            the<br>
            handshake) must be framed as per RFC 4571.<br>
            <br>
            <br>
            So I suggest to improve the above text as follows:<br>
            <br>
               If TCP connections are used, RTP framing according to
            [RFC4571] MUST<br>
               be used for the STUN packets, SRTP packets, SRTCP packets
            and DTLS<br>
               packets.<br>
            <span class="HOEnZb"><font color="#888888"><br>
                <br>
                <br>
                <br>
                <br>
                <br>
                --<br>
                Iñaki Baz Castillo<br>
                &lt;<a moz-do-not-send="true"
                  href="mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </font></span></blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

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Date: Wed, 22 Oct 2014 10:57:53 -0400
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From: Benjamin Schwartz <bemasc@webrtc.org>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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FYI, the purpose of the RETURN draft (
https://tools.ietf.org/html/draft-schwartz-rtcweb-return-03) is to try to
specify the precise interaction and precedence when both the browser and
the application provide TURN servers.  (It's not as simple as "one or the
other".)

I haven't considered the issues related to browser-provided STUN servers,
though.

On Wed, Oct 22, 2014 at 10:51 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> The advantage of an application configuration is that it can be
> dynamically provided/updated, based on location based etc - assuming the
> webpage provider has knowledge about STUN/TURN servers, of course.
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Olle E.
> Johansson
> Sent: 22 October 2014 17:44
> To: rtcweb@ietf.org
> Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt
>
> Section 3.4
>
> " WebRTC browsers MUST support configuration of STUN and TURN servers,
>    both from browser configuration and from an application."
>
>
> Should we mention which takes precedence? If configured in the browser,
> start with that server.
> If not use the one provided by the application.
>
> Maybe we should clarify that turn DNS discovery MUST be used to provide
> failover and load balancing.
>
> /O
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">FYI, the purpose of the RETURN draft (<a href=3D"https://t=
ools.ietf.org/html/draft-schwartz-rtcweb-return-03">https://tools.ietf.org/=
html/draft-schwartz-rtcweb-return-03</a>) is to try to specify the precise =
interaction and precedence when both the browser and the application provid=
e TURN servers. =C2=A0(It&#39;s not as simple as &quot;one or the other&quo=
t;.)<div><br></div><div>I haven&#39;t considered the issues related to brow=
ser-provided STUN servers, though.</div></div><div class=3D"gmail_extra"><b=
r><div class=3D"gmail_quote">On Wed, Oct 22, 2014 at 10:51 AM, Christer Hol=
mberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-boun=
ces@ietf.org</a>] On Behalf Of Olle E. Johansson<br>
Sent: 22 October 2014 17:44<br>
To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt<br>
<br>
Section 3.4<br>
<br>
&quot; WebRTC browsers MUST support configuration of STUN and TURN servers,=
<br>
=C2=A0 =C2=A0both from browser configuration and from an application.&quot;=
<br>
<br>
<br>
Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.<br>
If not use the one provided by the application.<br>
<br>
Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.<br>
<br>
/O<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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Date: Wed, 22 Oct 2014 08:58:44 -0600
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From: Simon Perreault <sperreault@jive.com>
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I just want to mention this:

http://tools.ietf.org/html/draft-ietf-tram-turn-server-discovery-00

I think that both browser and application configuration are included in
step #1 of the discovery algorithm, so that doesn't answer Olle's question.
But we still need to keep the whole discovery process in mind.

Simon

On Wed, Oct 22, 2014 at 8:51 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> The advantage of an application configuration is that it can be
> dynamically provided/updated, based on location based etc - assuming the
> webpage provider has knowledge about STUN/TURN servers, of course.
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Olle E.
> Johansson
> Sent: 22 October 2014 17:44
> To: rtcweb@ietf.org
> Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt
>
> Section 3.4
>
> " WebRTC browsers MUST support configuration of STUN and TURN servers,
>    both from browser configuration and from an application."
>
>
> Should we mention which takes precedence? If configured in the browser,
> start with that server.
> If not use the one provided by the application.
>
> Maybe we should clarify that turn DNS discovery MUST be used to provide
> failover and load balancing.
>
> /O
> _______________________________________________
> 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
>

--001a11c3eeee2feffe0506042fae
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<div dir=3D"ltr">I just want to mention this:<div><br></div><div><a href=3D=
"http://tools.ietf.org/html/draft-ietf-tram-turn-server-discovery-00">http:=
//tools.ietf.org/html/draft-ietf-tram-turn-server-discovery-00</a><br></div=
><div><br></div><div>I think that both browser and application configuratio=
n are included in step #1 of the discovery algorithm, so that doesn&#39;t a=
nswer Olle&#39;s question. But we still need to keep the whole discovery pr=
ocess in mind.</div><div><br></div><div>Simon</div></div><div class=3D"gmai=
l_extra"><br><div class=3D"gmail_quote">On Wed, Oct 22, 2014 at 8:51 AM, Ch=
rister Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@e=
ricsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span=
> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-boun=
ces@ietf.org</a>] On Behalf Of Olle E. Johansson<br>
Sent: 22 October 2014 17:44<br>
To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt<br>
<br>
Section 3.4<br>
<br>
&quot; WebRTC browsers MUST support configuration of STUN and TURN servers,=
<br>
=C2=A0 =C2=A0both from browser configuration and from an application.&quot;=
<br>
<br>
<br>
Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.<br>
If not use the one provided by the application.<br>
<br>
Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.<br>
<br>
/O<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a11c3eeee2feffe0506042fae--


From nobody Wed Oct 22 08:30:32 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 22 Oct 2014 17:29:54 +0200
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Subject: Re: [rtcweb] SDP and ssrc-group,
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2014-10-21 22:02 GMT+02:00 Justin Uberti <juberti@google.com>:
> In all seriousness, it would be nice if this behavior was standardized, s=
o
> that applications could have deterministic behavior. If we can come up wi=
th
> appropriate text about the ordering of SSRCs, this could be written into
> JSEP's handling of setLocalDescription.

So I expect a new draft about how to map ssrc values and payload-types
in SDP. I think we already have 4 or 5 drafts about mapping or
grouping stuff in SDP, but it looks like having them is not enough to
define which payload-types are carried over each declared SSRC within
a transport or m line (or bundled m lines within the same transport
because what a m line represents is no longer clear).

Is that? do we need a new SDP draft? or is it enough if we create a
convention within the WebRTC context?

Regards.


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


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi Sergio,


On 21/10/2014 22:14, Makaraju, Maridi Raju (Raju) wrote:

I hope I can rule my SDP logic based on standards rather than on how Chrome=
 implements some features.

My question was generic: if I receive the above SDP, how do I know which pa=
yloads each ssrc is supposed to transport?

<Raju >

Right. Firefox may have a different interpretation.

If a=3Dssrc-group and retransmissions are negotiated at SDP answer, then th=
e order of ssrcs probably does not matter as the RTP payload values can det=
ermine the retransmission vs. original payloads.

If answerer wants to know the original ssrc for rejecting a=3Dssrc-group an=
d retransmission payload then it need to know the original ssrc.



http://tools.ietf.org/html/rfc5576#section-6.3 defines a=3Dssrc with format=
. In the example below,

won't having the following SDP line be sufficient to assign 2693756249<tel:=
2693756249> for retransmission and, indirectly, assign  345259865 for origi=
nal .

a=3Dssrc:2693756249<tel:2693756249> fmtp:96

--------------------------
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
a=3Dmid:video
a=3Drtpmap:100 VP8/90000
a=3Drtpmap:116 red/90000
a=3Drtpmap:117 ulpfec/90000
a=3Drtpmap:96 rtx/90000
a=3Dfmtp:96 apt=3D100
a=3Dssrc-group:FID 345259865 2693756249<tel:2693756249>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:2693756249<tel:2693756249> cname:erS7E/KHLYKTejNs
a=3Dssrc:2693756249<tel:2693756249> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqB=
Y5
c0134f05-e7c2-4afd-a979-4e224de5eb91
a=3Dssrc:2693756249<tel:2693756249> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2Et=
fqBY5
a=3Dssrc:2693756249<tel:2693756249> label:c0134f05-e7c2-4afd-a979-4e224de5e=
b91
-------------------------------

On the question of what if there are 3 ssrcs?

Per my understanding, there is only one ssrc for original payload and anoth=
er for retransmission. So, a=3Dssrc-group containing 3 ssrcs is probably in=
valid.



</Raju>

Hi Raju

I foresee that when we start working on FEC we will end up using a differen=
t ssrc to avoid asking retransmissions of FEC redundant data. So 3 ssrcs wi=
ll be needed for each media stream.
<Raju2>
But for FEC, you will have to specify a different a=3Dssrc-group:FID. So, I=
 think a=3Dssrc-group:FID will still have 2 ssrcs only.
</Raju2>

Regarding the ftmp ssrc attribute, syntactically it solves perfectly the pr=
oblem (even if we have fec in its own ssrc), but I am not sure if semantica=
lly it is the intended usage, as it seems more to define format attributes =
for an specific ssrc an not to restrict the usage of the payload type to th=
at ssrc.
<Raju2>
Whatever was the initial intended purpose, I think it can be put to use it =
for this (new) usage. By definition, retransmission payload can only be sen=
t on one ssrc only and also the original payloads can only be sent on one s=
src only. So, assuming there is only one retransmission payload independent=
 of  # of original payloads, just associating it to an ssrc unambiguously m=
aps both ssrcs. But still, an explicit mapping for all ssrcs can also be do=
ne, thought which is unnecessary I think.
</Raju2>

BR
Raju

Also, I am not sure how that would play together with plan c/plan b/plan c/=
no plan (I have not followed that thread actively).

Best regards
Sergio

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<p style=3D"mso-margin-top-alt:0in;margin-right:.5in;margin-bottom:0in;marg=
in-left:0in;margin-bottom:.0001pt">
<b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">Hi Sergio,<o:p></o:p></span></i></b></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>
<p class=3D"MsoNormal"><br>
On 21/10/2014 22:14, Makaraju, Maridi Raju (Raju) wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<p>I hope I can rule my SDP logic based on standards rather than on how Chr=
ome implements some features.<o:p></o:p></p>
<p>My question was generic: if I receive the above SDP, how do I know which=
 payloads each ssrc is supposed to transport?<o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">&lt;Raju &gt;</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">Right. Firefox may have a different interpretation.</span></i></b><o:p>=
</o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">If a=3Dssrc-group and retransmissions are negotiated at SDP answer, the=
n the order of ssrcs probably does not matter as the RTP payload
 values can determine the retransmission vs. original payloads.</span></i><=
/b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">If answerer wants to know the original ssrc for rejecting a=3Dssrc-grou=
p and retransmission payload then it need to know the original
 ssrc.</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">&nbsp;</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D"><a href=3D"http://tools.ietf.org/html/rfc5576#section-6.3">http://tools=
.ietf.org/html/rfc5576#section-6.3</a> defines a=3Dssrc with
 format. In the example below,</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">won&#8217;t having the following SDP line be sufficient to assign
<a href=3D"tel:2693756249" target=3D"_blank"><span style=3D"color:#1F497D;t=
ext-decoration:none">2693756249</span></a> for retransmission and, indirect=
ly, assign &nbsp;345259865 for original .</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank"><span style=3D"co=
lor:#1F497D;text-decoration:none">2693756249</span></a> fmtp:96
</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt">--------------------------<br=
>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" target=3D"_blank">2=
693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 <a href=3D"label:c0134f05-e7c2-4afd-a979-4e224de5eb91">l=
abel:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> cname:=
erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> msid:D=
WpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> mslabe=
l:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> <a hre=
f=3D"label:c0134f05-e7c2-4afd-a979-4e224de5eb91">
label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
-------------------------------<br>
<br>
<b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">On the question of what if there are 3 ssr=
cs?</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">Per my understanding, there is only one ssrc for original payload and a=
nother for retransmission. So, a=3Dssrc-group containing 3
 ssrcs is probably invalid.</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">&nbsp;</span></i></b><o:p></o:p></p>
<p style=3D"margin:0in;margin-bottom:.0001pt"><b><i><span style=3D"font-siz=
e:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F49=
7D">&lt;/Raju&gt;</span></i></b><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</blockquote>
<p class=3D"MsoNormal">Hi Raju<br>
<br>
I foresee that when we start working on FEC we will end up using a differen=
t ssrc to avoid asking retransmissions of FEC redundant data. So 3 ssrcs wi=
ll be needed for each media stream.<br>
<b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">But for FEC, you wi=
ll have to specify a different a=3Dssrc-group:FID. So, I think a=3Dssrc-gro=
up:FID will still have 2 ssrcs only.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><br>
Regarding the ftmp ssrc attribute, syntactically it solves perfectly the pr=
oblem (even if we have fec in its own ssrc), but I am not sure if semantica=
lly it is the intended usage, as it seems more to define format attributes =
for an specific ssrc an not to restrict
 the usage of the payload type to that ssrc.&nbsp; <br>
<b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p></o:p></span></i></b></p=
>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Whatever was the in=
itial intended purpose, I think it can be put to use it for this (new) usag=
e. By definition, retransmission payload can only be sent
 on one ssrc only and also the original payloads can only be sent on one ss=
rc only. So, assuming there is only one retransmission payload independent =
of &nbsp;# of original payloads, just associating it to an ssrc unambiguous=
ly maps both ssrcs. But still, an explicit
 mapping for all ssrcs can also be done, thought which is unnecessary I thi=
nk. <o:p>
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span=
></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju<o:p></o:p></sp=
an></i></b></p>
<p class=3D"MsoNormal"><br>
Also, I am not sure how that would play together with plan c/plan b/plan c/=
no plan (I have not followed that thread actively).<br>
<br>
Best regards<br>
Sergio<o:p></o:p></p>
</div>
</div>
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</html>

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Date: Wed, 22 Oct 2014 18:13:20 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
References: <CALiegfmH8rRyEDbJjQ=kzMv0nGC=S9gNsE7roE=kqJyVcfgy8g@mail.gmail.com> <CAJrXDUHekuQCLeCYzsnm8AuTUgiVppQHUqR7MKdQ9Q=eFFAy0w@mail.gmail.com> <CALiegfm_B5KfD5SBPzsH4YYuzD2OXdu47TtatPVmd6ihrMCh1A@mail.gmail.com> <CAJrXDUF-AXcDuQmYhH91vbgPAxLkYkB==GY9opoRk7zrEP8A7A@mail.gmail.com> <CALiegfmxnzZ6_3rKX0paNhHas6Emvu1Mekgb9caj9NLVSf_u+g@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E5EAE2A@US70UWXCHMBA02.zam.alcatel-lucent.com> <5446C6D5.5090804@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E5EC0CA@US70UWXCHMBA02.zam.alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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On 22/10/2014 18:06, Makaraju, Maridi Raju (Raju) wrote:
>
> Hi Raju
>
> I foresee that when we start working on FEC we will end up using a 
> different ssrc to avoid asking retransmissions of FEC redundant data. 
> So 3 ssrcs will be needed for each media stream.
> */<Raju2> /*
>
> */But for FEC, you will have to specify a different a=ssrc-group:FID. 
> So, I think a=ssrc-group:FID will still have 2 ssrcs only./*
>
> */</Raju2>/*
>
Indeed, in fact I was proposing in another thread to use a the 
ssrc-group:FEC (and remove the RED altogether).

> Regarding the ftmp ssrc attribute, syntactically it solves perfectly 
> the problem (even if we have fec in its own ssrc), but I am not sure 
> if semantically it is the intended usage, as it seems more to define 
> format attributes for an specific ssrc an not to restrict the usage of 
> the payload type to that ssrc.
> */<Raju2>/*
>
> */Whatever was the initial intended purpose, I think it can be put to 
> use it for this (new) usage. By definition, retransmission payload can 
> only be sent on one ssrc only and also the original payloads can only 
> be sent on one ssrc only. So, assuming there is only one 
> retransmission payload independent of  # of original payloads, just 
> associating it to an ssrc unambiguously maps both ssrcs. But still, an 
> explicit mapping for all ssrcs can also be done, thought which is 
> unnecessary I think. /*
>
> */</Raju2>/*
>
I am also fine, just not sure if everyone else would agree.

Best regards
Sergio



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    <div class="moz-cite-prefix">On 22/10/2014 18:06, Makaraju, Maridi
      Raju (Raju) wrote:<br>
    </div>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5EC0CA@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <p class="MsoNormal">Hi Raju<br>
        <br>
        I foresee that when we start working on FEC we will end up using
        a different ssrc to avoid asking retransmissions of FEC
        redundant data. So 3 ssrcs will be needed for each media stream.<br>
        <b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;
              <o:p></o:p></span></i></b></p>
      <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">But
              for FEC, you will have to specify a different
              a=ssrc-group:FID. So, I think a=ssrc-group:FID will still
              have 2 ssrcs only.<o:p></o:p></span></i></b></p>
      <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p></o:p></span></i></b></p>
    </blockquote>
    Indeed, in fact I was proposing in another thread to use a the
    ssrc-group:FEC (and remove the RED altogether).<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5EC0CA@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <p class="MsoNormal">
        Regarding the ftmp ssrc attribute, syntactically it solves
        perfectly the problem (even if we have fec in its own ssrc), but
        I am not sure if semantically it is the intended usage, as it
        seems more to define format attributes for an specific ssrc an
        not to restrict the usage of the payload type to that ssrc.&nbsp; <br>
        <b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p></o:p></span></i></b></p>
      <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Whatever

              was the initial intended purpose, I think it can be put to
              use it for this (new) usage. By definition, retransmission
              payload can only be sent on one ssrc only and also the
              original payloads can only be sent on one ssrc only. So,
              assuming there is only one retransmission payload
              independent of &nbsp;# of original payloads, just associating
              it to an ssrc unambiguously maps both ssrcs. But still, an
              explicit mapping for all ssrcs can also be done, thought
              which is unnecessary I think. <o:p>
              </o:p></span></i></b></p>
      <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p></o:p></span></i></b></p>
    </blockquote>
    I am also fine, just not sure if everyone else would agree.<br>
    <br>
    Best regards<br>
    Sergio<br>
    <br>
    <br>
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--------------040506050002070307010906--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Justin Uberti <juberti@google.com>
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anUNCg==


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Subject: Re: [rtcweb] SDP and ssrc-group,
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SDP specifies payload types for RED/ULPFEC/RTX so as to allow differentiatio=
n within an ssrc-group. So while an fmtp: attribute could be used to denote a=
 media stream on an a:ssrc line, it is not necessary.

RFC 5176 Example 3 gives an example with two cameras in which there are two s=
src-group lines (one for each camera), with payload type used to differentia=
te between H.264 and RTX in each ssrc-group:

m=3Dvideo 49174 RTP/AVPF 96 98=20
 a=3Drtpmap:96 H.264/90000=20
 a=3Drtpmap:98 rtx/90000=20
 a=3Dfmtp:98 apt=3D96;rtx-time=3D3000=20
 a=3Dssrc-group:FID 11111 22222=20
 a=3Dssrc:11111 cname:user3@example.com=20
 a=3Dssrc:22222 cname:user3@example.com=20
 a=3Dssrc-group:FID 33333 44444=20
 a=3Dssrc:33333 cname:user3@example.com=20
 a=3Dssrc:44444 cname:user3@example.com

I believe that RFC 5176 includes the two groups to make clear that there are=
 two sources so that the Answerer might choose to reject the m-line if it ca=
nnot handle that. =46rom RFC 5176 Section 8:

When used with the SDP Offer/Answer Model [RFC3264], SDP source-specific att=
ributes describe only the sources that each party is willing to send (whethe=
r it is sending RTP data or RTCP report blocks). No mechanism is provided by=
 which an answer can accept or reject individual sources within a media stre=
am; if the set of sources in a media stream is unacceptable, the answerer's o=
nly option is to reject the media stream or the entire multimedia session.

Note that in the example in RFC 4588 Section 8.8 where there is only one sou=
rce, there are no ssrc-group lines at all:

 v=3D0=20
 o=3Dmascha 2980675221 2980675778 IN IP4 host.example.net=20
 c=3DIN IP4 192.0.2.0=20
 m=3Dvideo 49170 RTP/AVPF 96 97=20
 a=3Drtpmap:96 MP4V-ES/90000=20
 a=3Drtcp-fb:96 nack=20
 a=3Dfmtp:96 profile-level-id=3D8;config=3D01010000012000884006682C2090A21F=20=

 a=3Drtpmap:97 rtx/90000=20
 a=3Dfmtp:97 apt=3D96;rtx-time=3D3000

The situation would be the same for FEC groups created via ssrc-group:FEC. =46rom=
=20
https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :

The FEC and the payload MAY also be multiplexed by SSRC into one single RTP s=
ession, with separate SSRC values, if the association between FEC and payloa=
d streams are communicated to all members of the session. If SDP is used, th=
is association MAY be communicated through the FEC ssrc-group semantic [RFC5=
576]; other mechanisms are also possible. Receivers MUST NOT attempt to inte=
rpret FEC streams for which they do not have information to associate them w=
ith the corresponding payload streams.



> On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrot=
e:
>=20
> I hope I can rule my SDP logic based on standards rather than on how Chrom=
e implements some features.
>=20
> My question was generic: if I receive the above SDP, how do I know which p=
ayloads each ssrc is supposed to transport?
>=20
>> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:
>> I'm not sure where it's specified, but here's where it's implemented:
>>=20
>> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/we=
brtcvideoengine.cc#4108
>>=20
>> It only supports 2 SSRCs for a FID group.  It would ignore any more than t=
hat.
>>=20
>>> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net=
> wrote:
>>> How is that? Where is that specified? What about if I include 3 ssrc val=
ues in the ssrc-group? What is each one for?
>>>=20
>>>> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>>>> 345259865 is "real"
>>>> 2693756249 is rtx
>>>>=20
>>>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.ne=
t> wrote:
>>>>> Hi,
>>>>>=20
>>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>>>> RTX?
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> --------------------------
>>>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>>>> a=3Dmid:video
>>>>> a=3Drtpmap:100 VP8/90000
>>>>> a=3Drtpmap:116 red/90000
>>>>> a=3Drtpmap:117 ulpfec/90000
>>>>> a=3Drtpmap:96 rtx/90000
>>>>> a=3Dfmtp:96 apt=3D100
>>>>> a=3Dssrc-group:FID 345259865 2693756249
>>>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> -------------------------------
>>>>>=20
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> RFC 5576 does not clarify it at all:
>>>>>=20
>>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>>=20
>>>>> --------------------------------------------------
>>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>>=20
>>>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>>>=20
>>>>>    [..]
>>>>>=20
>>>>>    The <semantics> parameter is taken from the specification of the
>>>>>    "group" attribute [RFC3388].  The initial semantic values defined f=
or
>>>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>>    relationship among the grouped sources is the same as the
>>>>>    relationship among corresponding sources in media streams grouped
>>>>>    using the SDP "group" attribute.
>>>>> --------------------------------------------------
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> The referenced RFC 3388 neither clarifies it:
>>>>>=20
>>>>> ---------------------------------------------------
>>>>> 7.4 FID Semantics
>>>>>=20
>>>>>    Several "m" lines grouped together using FID semantics form a media=

>>>>>    flow.  A media agent handling a media flow that comprises several "=
m"
>>>>>    lines MUST send a copy of the media to every "m" line part of the
>>>>>    flow as long as the codecs and the direction attribute present in a=

>>>>>    particular "m" line allow it.
>>>>>=20
>>>>>    It is assumed that the application uses only one codec at a time to=

>>>>>    encode the media produced.  This codec MAY change dynamically durin=
g
>>>>>    the session, but at any particular moment only one codec is in use.=

>>>>>=20
>>>>>    The application encodes the media using the current codec and check=
s
>>>>>    one by one all the "m" lines that are part of the flow.  If a
>>>>>    particular "m" line contains the codec being used and the direction=

>>>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media i=
s
>>>>>    sent to the address/port specified in that particular media stream.=

>>>>>    If either the "m" line does not contain the codec being used or the=

>>>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing i=
s
>>>>>    sent over this media stream.
>>>>> ----------------------------------------------------
>>>>>=20
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>>=20
>>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>>=20
>>>>> How should the receiver interpret it?
>>>>>=20
>>>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>>>> the above SDP?
>>>>>=20
>>>>> Does not the above SDP look a complete mixture of hacks and workaround=
s?
>>>>>=20
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> --
>>>>> I=C3=B1aki Baz Castillo
>>>>> <ibc@aliax.net>
>>>>>=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

--Apple-Mail-10A6CC59-955F-4D31-94E8-2D7F50F9C270
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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>SDP specifies payload types for RED/UL=
PFEC/RTX so as to allow differentiation within an ssrc-group. So while an fm=
tp: attribute could be used to denote a media stream on an a:ssrc line, it i=
s not necessary.</div><div><br></div><div>RFC 5176 Example 3 gives an exampl=
e with two cameras in which there are two ssrc-group lines (one for each cam=
era), with payload type used to differentiate between H.264 and RTX in each s=
src-group:</div><div><br></div><div><pre class=3D"newpage" style=3D"margin-t=
op: 0px; margin-bottom: 0px; page-break-before: always;"><font face=3D"UICTFo=
ntTextStyleBody"><span style=3D"white-space: normal; background-color: rgba(=
255, 255, 255, 0);">
   m=3Dvideo 49174 RTP/AVPF 96 98&nbsp;</span></font></pre><pre class=3D"new=
page" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: alway=
s;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space: normal;=
 background-color: rgba(255, 255, 255, 0);">&nbsp;a=3Drtpmap:96 H.264/90000&=
nbsp;</span></font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; ma=
rgin-bottom: 0px; page-break-before: always;"><font face=3D"UICTFontTextStyl=
eBody"><span style=3D"white-space: normal; background-color: rgba(255, 255, 2=
55, 0);">&nbsp;a=3Drtpmap:98 rtx/90000&nbsp;</span></font></pre><pre class=3D=
"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: a=
lways;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space: nor=
mal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=3Dfmtp:98 apt=3D96;r=
tx-time=3D3000&nbsp;</span></font></pre><pre class=3D"newpage" style=3D"marg=
in-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face=3D"U=
ICTFontTextStyleBody"><span style=3D"white-space: normal; background-color: r=
gba(255, 255, 255, 0);">&nbsp;a=3Dssrc-group:FID 11111 22222&nbsp;</span></f=
ont></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0p=
x; page-break-before: always;"><font face=3D"UICTFontTextStyleBody"><span st=
yle=3D"white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp=
;a=3Dssrc:11111 cname:user3@<a href=3D"http://example.com">example.com</a>&n=
bsp;</span></font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; mar=
gin-bottom: 0px; page-break-before: always;"><font face=3D"UICTFontTextStyle=
Body"><span style=3D"white-space: normal; background-color: rgba(255, 255, 2=
55, 0);">&nbsp;a=3Dssrc:22222 cname:user3@<a href=3D"http://example.com">exa=
mple.com</a>&nbsp;</span></font></pre><pre class=3D"newpage" style=3D"margin=
-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face=3D"UIC=
TFontTextStyleBody"><span style=3D"white-space: normal; background-color: rg=
ba(255, 255, 255, 0);">&nbsp;a=3Dssrc-group:FID 33333 44444&nbsp;</span></fo=
nt></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px=
; page-break-before: always;"><font face=3D"UICTFontTextStyleBody"><span sty=
le=3D"white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;=
a=3Dssrc:33333 cname:user3@<a href=3D"http://example.com">example.com</a>&nb=
sp;</span></font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; marg=
in-bottom: 0px; page-break-before: always;"><font face=3D"UICTFontTextStyleB=
ody"><span style=3D"white-space: normal; background-color: rgba(255, 255, 25=
5, 0);">&nbsp;a=3Dssrc:44444 cname:user3@<a href=3D"http://example.com">exam=
ple.com</a></span></font><span style=3D"font-size: 1em; -webkit-text-size-ad=
just: auto;">
</span></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom:=
 0px; page-break-before: always;"><font face=3D"UICTFontTextStyleBody"><span=
 style=3D"white-space: normal; background-color: rgba(255, 255, 255, 0);"><b=
r></span></font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margi=
n-bottom: 0px; page-break-before: always;"><font face=3D"UICTFontTextStyleBo=
dy"><span style=3D"white-space: normal; background-color: rgba(255, 255, 255=
, 0);">I believe that RFC 5176 includes the two groups to make clear that th=
ere are two sources so that the Answerer might choose to reject the m-line i=
f it cannot handle that.&nbsp;</span></font><span style=3D"white-space: norm=
al; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyle=
Body;">=46rom RFC 5176 Section 8:</span></pre><pre class=3D"newpage" style=3D=
"margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face=
=3D"UICTFontTextStyleBody"><span style=3D"white-space: normal; background-co=
lor: rgba(255, 255, 255, 0);"><br></span></font></pre><pre class=3D"newpage"=
 style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><=
span style=3D"white-space: normal; background-color: rgba(255, 255, 255, 0);=
 font-family: UICTFontTextStyleBody;">When used with the SDP Offer/Answer Mo=
del [</span><a href=3D"http://tools.ietf.org/html/rfc3264" title=3D"&quot;An=
 Offer/Answer Model with Session Description Protocol (SDP)&quot;" style=3D"=
white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: U=
ICTFontTextStyleBody;">RFC3264</a><span style=3D"white-space: normal; backgr=
ound-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">], S=
DP source-specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer's only option
   is to reject the media stream or the entire multimedia session.</span></p=
re><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page=
-break-before: always;"><font face=3D"UICTFontTextStyleBody"><span style=3D"=
white-space: normal; background-color: rgba(255, 255, 255, 0);"><br></span><=
/font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0=
px; page-break-before: always;"><span style=3D"white-space: normal; backgrou=
nd-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">Note t=
hat in the example in RFC 4588 Section 8.8 where there is only one source, t=
here are no ssrc-group lines at all:</span></pre><pre class=3D"newpage" styl=
e=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span s=
tyle=3D"white-space: normal; background-color: rgba(255, 255, 255, 0); font-=
family: UICTFontTextStyleBody;"><br></span></pre><pre class=3D"newpage" styl=
e=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span s=
tyle=3D"white-space: normal; background-color: rgba(255, 255, 255, 0); font-=
family: UICTFontTextStyleBody;">&nbsp;v=3D0&nbsp;</span></pre><pre class=3D"=
newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: al=
ways;"><span style=3D"white-space: normal; background-color: rgba(255, 255, 2=
55, 0); font-family: UICTFontTextStyleBody;">&nbsp;o=3Dmascha 2980675221 298=
0675778 IN IP4 <a href=3D"http://host.example.net">host.example.net</a>&nbsp=
;</span></pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom=
: 0px; page-break-before: always;"><span style=3D"white-space: normal; backg=
round-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&n=
bsp;c=3DIN IP4 192.0.2.0&nbsp;</span></pre><pre class=3D"newpage" style=3D"m=
argin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style=3D=
"white-space: normal; background-color: rgba(255, 255, 255, 0); font-family:=
 UICTFontTextStyleBody;">&nbsp;m=3Dvideo 49170 RTP/AVPF 96 97&nbsp;</span></=
pre><pre class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; pag=
e-break-before: always;"><span style=3D"white-space: normal; background-colo=
r: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=3Drt=
pmap:96 MP4V-ES/90000&nbsp;</span></pre><pre class=3D"newpage" style=3D"marg=
in-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style=3D"=
white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: U=
ICTFontTextStyleBody;">&nbsp;a=3Drtcp-fb:96 nack&nbsp;</span></pre><pre clas=
s=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-befor=
e: always;"><span style=3D"white-space: normal; background-color: rgba(255, 2=
55, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=3Dfmtp:96 profile-=
level-id=3D8;config=3D01010000012000884006682C2090A21F&nbsp;</span></pre><pr=
e class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break=
-before: always;"><span style=3D"white-space: normal; background-color: rgba=
(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=3Drtpmap:97=
 rtx/90000&nbsp;</span></pre><pre class=3D"newpage" style=3D"margin-top: 0px=
; margin-bottom: 0px; page-break-before: always;"><span style=3D"white-space=
: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTex=
tStyleBody;">&nbsp;a=3Dfmtp:97 apt=3D96;rtx-time=3D3000</span></pre></div><d=
iv><br></div><div>The situation would be the same for FEC groups created via=
 ssrc-group:FEC. From&nbsp;</div><div><a href=3D"https://tools.ietf.org/html=
/draft-lennox-payload-ulp-ssrc-mux">https://tools.ietf.org/html/draft-lennox=
-payload-ulp-ssrc-mux</a> &nbsp;:</div><div><br></div><div><pre class=3D"new=
page" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: alway=
s;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space: normal;=
 background-color: rgba(255, 255, 255, 0);">      The FEC and the payload MA=
Y also be multiplexed by SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a href=3D"https://tools.ietf.org=
/html/rfc5576" title=3D"&quot;Source-Specific Media Attributes in the Sessio=
n Description Protocol (SDP)&quot;">RFC5576</a>]; other mechanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style=3D"fo=
nt-size: 1em; -webkit-text-size-adjust: auto;">
</span></pre></div><div><br></div><div><br></div><div><br>On Oct 21, 2014, a=
t 11:36 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net">ibc=
@aliax.net</a>&gt; wrote:<br><br></div><blockquote type=3D"cite"><div><p dir=
=3D"ltr">I hope I can rule my SDP logic based on standards rather than on ho=
w Chrome implements some features.</p>
<p dir=3D"ltr">My question was generic: if I receive the above SDP, how do I=
 know which payloads each ssrc is supposed to transport?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:57, "Peter Thatcher" &lt;<a hre=
f=3D"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; wrote:<br typ=
e=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">I'm not sur=
e where it's specified, but here's where it's implemented:</div><div class=3D=
"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></div><=
div class=3D"gmail_default"><font face=3D"arial, helvetica, sans-serif"><a h=
ref=3D"https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrt=
c/webrtcvideoengine.cc#4108" target=3D"_blank">https://code.google.com/p/web=
rtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108</a></fon=
t><br></div><div class=3D"gmail_default"><font face=3D"arial, helvetica, san=
s-serif"><br></font></div><div class=3D"gmail_default"><font face=3D"arial, h=
elvetica, sans-serif">It only supports 2 SSRCs for a FID group.&nbsp; It wou=
ld ignore any more than that.</font></div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki B=
az Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D=
"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
"><p dir=3D"ltr">How is that? Where is that specified? What about if I inclu=
de 3 ssrc values in the ssrc-group? What is each one for?</p><div><div>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, "Peter Thatcher" &lt;<a hre=
f=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>=
&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"=
ltr"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-=
serif"><span style=3D"font-family:arial,sans-serif;font-size:12.727272033691=
4px">345259865 is "real"</span><br></div><div class=3D"gmail_default"><font f=
ace=3D"arial, sans-serif"><a href=3D"tel:2693756249" value=3D"+12693756249" t=
arget=3D"_blank">2693756249</a> is rtx</font><br></div></div><div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 9:25 AM, I=
=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net=
" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;paddi=
ng-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+126=
93756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+1269375624=
9" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank"=
>http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.&nbsp; The "ssrc-group" Media Attribute<br>
<br>
&nbsp; &nbsp;a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
&nbsp; &nbsp;[..]<br>
<br>
&nbsp; &nbsp;The &lt;semantics&gt; parameter is taken from the specification=
 of the<br>
&nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The initial semantic values d=
efined for<br>
&nbsp; &nbsp;the "ssrc-group" attribute are FID (Flow Identification) [RFC33=
88]<br>
&nbsp; &nbsp;and FEC (Forward Error Correction) [RFC4756].&nbsp; In each cas=
e, the<br>
&nbsp; &nbsp;relationship among the grouped sources is the same as the<br>
&nbsp; &nbsp;relationship among corresponding sources in media streams group=
ed<br>
&nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
&nbsp; &nbsp;Several "m" lines grouped together using FID semantics form a m=
edia<br>
&nbsp; &nbsp;flow.&nbsp; A media agent handling a media flow that comprises s=
everal "m"<br>
&nbsp; &nbsp;lines MUST send a copy of the media to every "m" line part of t=
he<br>
&nbsp; &nbsp;flow as long as the codecs and the direction attribute present i=
n a<br>
&nbsp; &nbsp;particular "m" line allow it.<br>
<br>
&nbsp; &nbsp;It is assumed that the application uses only one codec at a tim=
e to<br>
&nbsp; &nbsp;encode the media produced.&nbsp; This codec MAY change dynamica=
lly during<br>
&nbsp; &nbsp;the session, but at any particular moment only one codec is in u=
se.<br>
<br>
&nbsp; &nbsp;The application encodes the media using the current codec and c=
hecks<br>
&nbsp; &nbsp;one by one all the "m" lines that are part of the flow.&nbsp; I=
f a<br>
&nbsp; &nbsp;particular "m" line contains the codec being used and the direc=
tion<br>
&nbsp; &nbsp;attribute is "sendonly" or "sendrecv", a copy of the encoded me=
dia is<br>
&nbsp; &nbsp;sent to the address/port specified in that particular media str=
eam.<br>
&nbsp; &nbsp;If either the "m" line does not contain the codec being used or=
 the<br>
&nbsp; &nbsp;direction attribute is neither "sendonly" nor "sendrecv", nothi=
ng is<br>
&nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br>=

<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
<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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></div></blockquote></div><br></div>
</blockquote></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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Subject: Re: [rtcweb] SDP and ssrc-group,
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Just thinking loud
 As long as there is enough information to unabigously map the incoming
streams to the SDP, i dont see we need any more information to be added.

I feel the combination of SSRC and Payload Type in the RTP packet is good
enough to find the mapping to the appropriate media line (if the PTs are
not repeated) when needed.

So in the initial examples above, ssrc-group communicated in SDP along with
semantics (FEC,FID) when applied to incoming SSRCs and the associated
payload type has all the needed info for the receiver to demux at the rtp
level as well as map at the SDP level. Thus i dont see a need for
specifying PT association explicitly in SDP for the ssrc-groups.

Cheers
Suhas

On Wed, Oct 22, 2014 at 12:01 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> SDP specifies payload types for RED/ULPFEC/RTX so as to allow
> differentiation within an ssrc-group. So while an fmtp: attribute could b=
e
> used to denote a media stream on an a:ssrc line, it is not necessary.
>
> RFC 5176 Example 3 gives an example with two cameras in which there are
> two ssrc-group lines (one for each camera), with payload type used to
> differentiate between H.264 and RTX in each ssrc-group:
>
>
>    m=3Dvideo 49174 RTP/AVPF 96 98
>
>  a=3Drtpmap:96 H.264/90000
>
>  a=3Drtpmap:98 rtx/90000
>
>  a=3Dfmtp:98 apt=3D96;rtx-time=3D3000
>
>  a=3Dssrc-group:FID 11111 22222
>
>  a=3Dssrc:11111 cname:user3@example.com
>
>  a=3Dssrc:22222 cname:user3@example.com
>
>  a=3Dssrc-group:FID 33333 44444
>
>  a=3Dssrc:33333 cname:user3@example.com
>
>  a=3Dssrc:44444 cname:user3@example.com
>
>
> I believe that RFC 5176 includes the two groups to make clear that there =
are two sources so that the Answerer might choose to reject the m-line if i=
t cannot handle that. From RFC 5176 Section 8:
>
>
> When used with the SDP Offer/Answer Model [RFC3264 <http://tools.ietf.org=
/html/rfc3264>], SDP source-specific attributes describe only the sources t=
hat each party is
>    willing to send (whether it is sending RTP data or RTCP report
>    blocks).  No mechanism is provided by which an answer can accept or
>    reject individual sources within a media stream; if the set of
>    sources in a media stream is unacceptable, the answerer's only option
>    is to reject the media stream or the entire multimedia session.
>
>
> Note that in the example in RFC 4588 Section 8.8 where there is only one =
source, there are no ssrc-group lines at all:
>
>
>  v=3D0
>
>  o=3Dmascha 2980675221 2980675778 IN IP4 host.example.net
>
>  c=3DIN IP4 192.0.2.0
>
>  m=3Dvideo 49170 RTP/AVPF 96 97
>
>  a=3Drtpmap:96 MP4V-ES/90000
>
>  a=3Drtcp-fb:96 nack
>
>  a=3Dfmtp:96 profile-level-id=3D8;config=3D01010000012000884006682C2090A2=
1F
>
>  a=3Drtpmap:97 rtx/90000
>
>  a=3Dfmtp:97 apt=3D96;rtx-time=3D3000
>
>
> The situation would be the same for FEC groups created via ssrc-group:FEC=
.
> From
> https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
>
>       The FEC and the payload MAY also be multiplexed by SSRC into one
>       single RTP session, with separate SSRC values, if the association
>       between FEC and payload streams are communicated to all members of
>       the session.  If SDP is used, this association MAY be communicated
>       through the FEC ssrc-group semantic [RFC5576 <https://tools.ietf.or=
g/html/rfc5576>]; other mechanisms
>       are also possible.  Receivers MUST NOT attempt to interpret FEC
>       streams for which they do not have information to associate them
>       with the corresponding payload streams.
>
>
>
>
> On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:
>
> I hope I can rule my SDP logic based on standards rather than on how
> Chrome implements some features.
>
> My question was generic: if I receive the above SDP, how do I know which
> payloads each ssrc is supposed to transport?
> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:
>
>> I'm not sure where it's specified, but here's where it's implemented:
>>
>>
>> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/w=
ebrtcvideoengine.cc#4108
>>
>> It only supports 2 SSRCs for a FID group.  It would ignore any more than
>> that.
>>
>> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net=
>
>> wrote:
>>
>>> How is that? Where is that specified? What about if I include 3 ssrc
>>> values in the ssrc-group? What is each one for?
>>> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>>>
>>>> 345259865 is "real"
>>>> 2693756249 is rtx
>>>>
>>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.ne=
t>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>>>> RTX?
>>>>>
>>>>>
>>>>>
>>>>> --------------------------
>>>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>>>> a=3Dmid:video
>>>>> a=3Drtpmap:100 VP8/90000
>>>>> a=3Drtpmap:116 red/90000
>>>>> a=3Drtpmap:117 ulpfec/90000
>>>>> a=3Drtpmap:96 rtx/90000
>>>>> a=3Dfmtp:96 apt=3D100
>>>>> a=3Dssrc-group:FID 345259865 2693756249
>>>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> -------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> RFC 5576 does not clarify it at all:
>>>>>
>>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>>
>>>>> --------------------------------------------------
>>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>>
>>>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>>>
>>>>>    [..]
>>>>>
>>>>>    The <semantics> parameter is taken from the specification of the
>>>>>    "group" attribute [RFC3388].  The initial semantic values defined
>>>>> for
>>>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>>    relationship among the grouped sources is the same as the
>>>>>    relationship among corresponding sources in media streams grouped
>>>>>    using the SDP "group" attribute.
>>>>> --------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>> The referenced RFC 3388 neither clarifies it:
>>>>>
>>>>> ---------------------------------------------------
>>>>> 7.4 FID Semantics
>>>>>
>>>>>    Several "m" lines grouped together using FID semantics form a medi=
a
>>>>>    flow.  A media agent handling a media flow that comprises several
>>>>> "m"
>>>>>    lines MUST send a copy of the media to every "m" line part of the
>>>>>    flow as long as the codecs and the direction attribute present in =
a
>>>>>    particular "m" line allow it.
>>>>>
>>>>>    It is assumed that the application uses only one codec at a time t=
o
>>>>>    encode the media produced.  This codec MAY change dynamically duri=
ng
>>>>>    the session, but at any particular moment only one codec is in use=
.
>>>>>
>>>>>    The application encodes the media using the current codec and chec=
ks
>>>>>    one by one all the "m" lines that are part of the flow.  If a
>>>>>    particular "m" line contains the codec being used and the directio=
n
>>>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded media
>>>>> is
>>>>>    sent to the address/port specified in that particular media stream=
.
>>>>>    If either the "m" line does not contain the codec being used or th=
e
>>>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing =
is
>>>>>    sent over this media stream.
>>>>> ----------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>>
>>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>>
>>>>> How should the receiver interpret it?
>>>>>
>>>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>>>> the above SDP?
>>>>>
>>>>> Does not the above SDP look a complete mixture of hacks and
>>>>> workarounds?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> I=C3=B1aki Baz Castillo
>>>>> <ibc@aliax.net>
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>
>>>>
>>  _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e013c6b68004253050607ca79
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Just thinking loud=C2=A0<div>=C2=A0As long as there is eno=
ugh information to unabigously map the incoming streams to the SDP, i dont =
see we need any more information to be added.</div><div><br></div><div>I fe=
el the combination of SSRC and Payload Type in the RTP packet is good enoug=
h to find the mapping to the appropriate media line (if the PTs are not rep=
eated) when needed.</div><div><br></div><div>So in the initial examples abo=
ve, ssrc-group communicated in SDP along with semantics (FEC,FID) when appl=
ied to incoming SSRCs and the associated payload type has all the needed in=
fo for the receiver to demux at the rtp level as well as map at the SDP lev=
el. Thus i dont see a need for specifying PT association explicitly in SDP =
for the ssrc-groups.</div><div><br></div><div>Cheers</div><div>Suhas</div><=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Oct =
22, 2014 at 12:01 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto=
:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"auto"><div>SDP=
 specifies payload types for RED/ULPFEC/RTX so as to allow differentiation =
within an ssrc-group. So while an fmtp: attribute could be used to denote a=
 media stream on an a:ssrc line, it is not necessary.</div><div><br></div><=
div>RFC 5176 Example 3 gives an example with two cameras in which there are=
 two ssrc-group lines (one for each camera), with payload type used to diff=
erentiate between H.264 and RTX in each ssrc-group:</div><div><br></div><di=
v><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTex=
tStyleBody"><span style=3D"white-space:normal;background-color:rgba(255,255=
,255,0)">
   m=3Dvideo 49174 RTP/AVPF 96 98=C2=A0</span></font></pre><pre style=3D"ma=
rgin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span =
style=3D"white-space:normal;background-color:rgba(255,255,255,0)">=C2=A0a=
=3Drtpmap:96 H.264/90000=C2=A0</span></font></pre><pre style=3D"margin-top:=
0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"=
white-space:normal;background-color:rgba(255,255,255,0)">=C2=A0a=3Drtpmap:9=
8 rtx/90000=C2=A0</span></font></pre><pre style=3D"margin-top:0px;margin-bo=
ttom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:n=
ormal;background-color:rgba(255,255,255,0)">=C2=A0a=3Dfmtp:98 apt=3D96;rtx-=
time=3D3000=C2=A0</span></font></pre><pre style=3D"margin-top:0px;margin-bo=
ttom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:n=
ormal;background-color:rgba(255,255,255,0)">=C2=A0a=3Dssrc-group:FID 11111 =
22222=C2=A0</span></font></pre><pre style=3D"margin-top:0px;margin-bottom:0=
px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;=
background-color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:11111 cname:user3@<a h=
ref=3D"http://example.com" target=3D"_blank">example.com</a>=C2=A0</span></=
font></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UI=
CTFontTextStyleBody"><span style=3D"white-space:normal;background-color:rgb=
a(255,255,255,0)">=C2=A0a=3Dssrc:22222 cname:user3@<a href=3D"http://exampl=
e.com" target=3D"_blank">example.com</a>=C2=A0</span></font></pre><pre styl=
e=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"=
><span style=3D"white-space:normal;background-color:rgba(255,255,255,0)">=
=C2=A0a=3Dssrc-group:FID 33333 44444=C2=A0</span></font></pre><pre style=3D=
"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><sp=
an style=3D"white-space:normal;background-color:rgba(255,255,255,0)">=C2=A0=
a=3Dssrc:33333 cname:user3@<a href=3D"http://example.com" target=3D"_blank"=
>example.com</a>=C2=A0</span></font></pre><pre style=3D"margin-top:0px;marg=
in-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-sp=
ace:normal;background-color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:44444 cname=
:user3@<a href=3D"http://example.com" target=3D"_blank">example.com</a></sp=
an></font><span style=3D"font-size:1em">
</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"=
UICTFontTextStyleBody"><span style=3D"white-space:normal;background-color:r=
gba(255,255,255,0)"><br></span></font></pre><pre style=3D"margin-top:0px;ma=
rgin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-=
space:normal;background-color:rgba(255,255,255,0)">I believe that RFC 5176 =
includes the two groups to make clear that there are two sources so that th=
e Answerer might choose to reject the m-line if it cannot handle that.=C2=
=A0</span></font><span style=3D"white-space:normal;background-color:rgba(25=
5,255,255,0);font-family:UICTFontTextStyleBody">From RFC 5176 Section 8:</s=
pan></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UIC=
TFontTextStyleBody"><span style=3D"white-space:normal;background-color:rgba=
(255,255,255,0)"><br></span></font></pre><pre style=3D"margin-top:0px;margi=
n-bottom:0px"><span style=3D"white-space:normal;background-color:rgba(255,2=
55,255,0);font-family:UICTFontTextStyleBody">When used with the SDP Offer/A=
nswer Model [</span><a href=3D"http://tools.ietf.org/html/rfc3264" title=3D=
"&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;"=
 style=3D"white-space:normal;background-color:rgba(255,255,255,0);font-fami=
ly:UICTFontTextStyleBody" target=3D"_blank">RFC3264</a><span style=3D"white=
-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontText=
StyleBody">], SDP source-specific attributes describe only the sources that=
 each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer&#39;s only optio=
n
   is to reject the media stream or the entire multimedia session.</span></=
pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontT=
extStyleBody"><span style=3D"white-space:normal;background-color:rgba(255,2=
55,255,0)"><br></span></font></pre><pre style=3D"margin-top:0px;margin-bott=
om:0px"><span style=3D"white-space:normal;background-color:rgba(255,255,255=
,0);font-family:UICTFontTextStyleBody">Note that in the example in RFC 4588=
 Section 8.8 where there is only one source, there are no ssrc-group lines =
at all:</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody"><br></span></pre><pre style=3D"margin-top:0px;margi=
n-bottom:0px"><span style=3D"white-space:normal;background-color:rgba(255,2=
55,255,0);font-family:UICTFontTextStyleBody">=C2=A0v=3D0=C2=A0</span></pre>=
<pre style=3D"margin-top:0px;margin-bottom:0px"><span style=3D"white-space:=
normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBo=
dy">=C2=A0o=3Dmascha 2980675221 2980675778 IN IP4 <a href=3D"http://host.ex=
ample.net" target=3D"_blank">host.example.net</a>=C2=A0</span></pre><pre st=
yle=3D"margin-top:0px;margin-bottom:0px"><span style=3D"white-space:normal;=
background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">=C2=
=A0c=3DIN IP4 192.0.2.0=C2=A0</span></pre><pre style=3D"margin-top:0px;marg=
in-bottom:0px"><span style=3D"white-space:normal;background-color:rgba(255,=
255,255,0);font-family:UICTFontTextStyleBody">=C2=A0m=3Dvideo 49170 RTP/AVP=
F 96 97=C2=A0</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><=
span style=3D"white-space:normal;background-color:rgba(255,255,255,0);font-=
family:UICTFontTextStyleBody">=C2=A0a=3Drtpmap:96 MP4V-ES/90000=C2=A0</span=
></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span style=3D"white=
-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontText=
StyleBody">=C2=A0a=3Drtcp-fb:96 nack=C2=A0</span></pre><pre style=3D"margin=
-top:0px;margin-bottom:0px"><span style=3D"white-space:normal;background-co=
lor:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">=C2=A0a=3Dfmtp:9=
6 profile-level-id=3D8;config=3D01010000012000884006682C2090A21F=C2=A0</spa=
n></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span style=3D"whit=
e-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTex=
tStyleBody">=C2=A0a=3Drtpmap:97 rtx/90000=C2=A0</span></pre><pre style=3D"m=
argin-top:0px;margin-bottom:0px"><span style=3D"white-space:normal;backgrou=
nd-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">=C2=A0a=3Df=
mtp:97 apt=3D96;rtx-time=3D3000</span></pre></div><div><br></div><div>The s=
ituation would be the same for FEC groups created via ssrc-group:FEC. From=
=C2=A0</div><div><a href=3D"https://tools.ietf.org/html/draft-lennox-payloa=
d-ulp-ssrc-mux" target=3D"_blank">https://tools.ietf.org/html/draft-lennox-=
payload-ulp-ssrc-mux</a> =C2=A0:</div><div><br></div><div><pre style=3D"mar=
gin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0)">      The =
FEC and the payload MAY also be multiplexed by SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a href=3D"https://tools.ietf.or=
g/html/rfc5576" title=3D"&quot;Source-Specific Media Attributes in the Sess=
ion Description Protocol (SDP)&quot;" target=3D"_blank">RFC5576</a>]; other=
 mechanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style=3D"f=
ont-size:1em">
</span></pre></div><div><div class=3D"h5"><div><br></div><div><br></div><di=
v><br>On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"=
mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt; wrote:<br><br=
></div><blockquote type=3D"cite"><div><p dir=3D"ltr">I hope I can rule my S=
DP logic based on standards rather than on how Chrome implements some featu=
res.</p>
<p dir=3D"ltr">My question was generic: if I receive the above SDP, how do =
I know which payloads each ssrc is supposed to transport?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:57, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@go=
ogle.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif">I&#39;m not sure where it&#39;s specified, but here=
&#39;s where it&#39;s implemented:</div><div class=3D"gmail_default" style=
=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_d=
efault"><font face=3D"arial, helvetica, sans-serif"><a href=3D"https://code=
.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengin=
e.cc#4108" target=3D"_blank">https://code.google.com/p/webrtc/source/browse=
/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108</a></font><br></div><div=
 class=3D"gmail_default"><font face=3D"arial, helvetica, sans-serif"><br></=
font></div><div class=3D"gmail_default"><font face=3D"arial, helvetica, san=
s-serif">It only supports 2 SSRCs for a FID group.=C2=A0 It would ignore an=
y more than that.</font></div></div><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote">On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castill=
o <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">=
ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p di=
r=3D"ltr">How is that? Where is that specified? What about if I include 3 s=
src values in the ssrc-group? What is each one for?</p><div><div>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, &quot;Peter Thatcher&quot;=
 &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@go=
ogle.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif"><span style=3D"font-family:arial,sans-serif;font-si=
ze:12.7272720336914px">345259865 is &quot;real&quot;</span><br></div><div c=
lass=3D"gmail_default"><font face=3D"arial, sans-serif"><a href=3D"tel:2693=
756249" value=3D"+12693756249" target=3D"_blank">2693756249</a> is rtx</fon=
t><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+126937562=
49" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
=C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the specificatio=
n of the<br>
=C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initial seman=
tic values defined for<br>
=C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Identificat=
ion) [RFC3388]<br>
=C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In each ca=
se, the<br>
=C2=A0 =C2=A0relationship among the grouped sources is the same as the<br>
=C2=A0 =C2=A0relationship among corresponding sources in media streams grou=
ped<br>
=C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID semanti=
cs form a media<br>
=C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that comprises=
 several &quot;m&quot;<br>
=C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&quot; lin=
e part of the<br>
=C2=A0 =C2=A0flow as long as the codecs and the direction attribute present=
 in a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0It is assumed that the application uses only one codec at a ti=
me to<br>
=C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change dynamic=
ally during<br>
=C2=A0 =C2=A0the session, but at any particular moment only one codec is in=
 use.<br>
<br>
=C2=A0 =C2=A0The application encodes the media using the current codec and =
checks<br>
=C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of the fl=
ow.=C2=A0 If a<br>
=C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being used an=
d the direction<br>
=C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quot;, a c=
opy of the encoded media is<br>
=C2=A0 =C2=A0sent to the address/port specified in that particular media st=
ream.<br>
=C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the codec be=
ing used or the<br>
=C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor &quot;=
sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<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>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></div></blockquote></div><br></div>
</blockquote></div>
</div></blockquote><blockquote type=3D"cite"><div><span>___________________=
____________________________</span><br><span>rtcweb mailing list</span><br>=
<span><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org<=
/a></span><br><span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span>=
<br></div></blockquote></div></div></div><br>______________________________=
_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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Subject: Re: [rtcweb] SDP and ssrc-group,
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I agree. The SSRC-groups are included only to make it clear which FEC/RED/RT=
X streams go with which media streams in the multiple source case. As long a=
s payload types are distinct we have all the information we need.

> On Oct 22, 2014, at 12:16 PM, Suhas Nandakumar <suhasietf@gmail.com> wrote=
:
>=20
> Just thinking loud=20
>  As long as there is enough information to unabigously map the incoming st=
reams to the SDP, i dont see we need any more information to be added.
>=20
> I feel the combination of SSRC and Payload Type in the RTP packet is good e=
nough to find the mapping to the appropriate media line (if the PTs are not r=
epeated) when needed.
>=20
> So in the initial examples above, ssrc-group communicated in SDP along wit=
h semantics (FEC,FID) when applied to incoming SSRCs and the associated payl=
oad type has all the needed info for the receiver to demux at the rtp level a=
s well as map at the SDP level. Thus i dont see a need for specifying PT ass=
ociation explicitly in SDP for the ssrc-groups.
>=20
> Cheers
> Suhas
>=20
>> On Wed, Oct 22, 2014 at 12:01 PM, Bernard Aboba <bernard.aboba@gmail.com>=
 wrote:
>> SDP specifies payload types for RED/ULPFEC/RTX so as to allow differentia=
tion within an ssrc-group. So while an fmtp: attribute could be used to deno=
te a media stream on an a:ssrc line, it is not necessary.
>>=20
>> RFC 5176 Example 3 gives an example with two cameras in which there are t=
wo ssrc-group lines (one for each camera), with payload type used to differe=
ntiate between H.264 and RTX in each ssrc-group:
>>=20
>> m=3Dvideo 49174 RTP/AVPF 96 98=20
>>  a=3Drtpmap:96 H.264/90000=20
>>  a=3Drtpmap:98 rtx/90000=20
>>  a=3Dfmtp:98 apt=3D96;rtx-time=3D3000=20
>>  a=3Dssrc-group:FID 11111 22222=20
>>  a=3Dssrc:11111 cname:user3@example.com=20
>>  a=3Dssrc:22222 cname:user3@example.com=20
>>  a=3Dssrc-group:FID 33333 44444=20
>>  a=3Dssrc:33333 cname:user3@example.com=20
>>  a=3Dssrc:44444 cname:user3@example.com
>>=20
>> I believe that RFC 5176 includes the two groups to make clear that there a=
re two sources so that the Answerer might choose to reject the m-line if it c=
annot handle that. =46rom RFC 5176 Section 8:
>>=20
>> When used with the SDP Offer/Answer Model [RFC3264], SDP source-specific a=
ttributes describe only the sources that each party is willing to send (whet=
her it is sending RTP data or RTCP report blocks). No mechanism is provided b=
y which an answer can accept or reject individual sources within a media str=
eam; if the set of sources in a media stream is unacceptable, the answerer's=
 only option is to reject the media stream or the entire multimedia session.=

>>=20
>> Note that in the example in RFC 4588 Section 8.8 where there is only one s=
ource, there are no ssrc-group lines at all:
>>=20
>>  v=3D0=20
>>  o=3Dmascha 2980675221 2980675778 IN IP4 host.example.net=20
>>  c=3DIN IP4 192.0.2.0=20
>>  m=3Dvideo 49170 RTP/AVPF 96 97=20
>>  a=3Drtpmap:96 MP4V-ES/90000=20
>>  a=3Drtcp-fb:96 nack=20
>>  a=3Dfmtp:96 profile-level-id=3D8;config=3D01010000012000884006682C2090A2=
1F=20
>>  a=3Drtpmap:97 rtx/90000=20
>>  a=3Dfmtp:97 apt=3D96;rtx-time=3D3000
>>=20
>> The situation would be the same for FEC groups created via ssrc-group:FEC=
. =46rom=20
>> https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
>>=20
>> The FEC and the payload MAY also be multiplexed by SSRC into one single R=
TP session, with separate SSRC values, if the association between FEC and pa=
yload streams are communicated to all members of the session. If SDP is used=
, this association MAY be communicated through the FEC ssrc-group semantic [=
RFC5576]; other mechanisms are also possible. Receivers MUST NOT attempt to i=
nterpret FEC streams for which they do not have information to associate the=
m with the corresponding payload streams.
>>=20
>>=20
>>=20
>>> On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:
>>>=20
>>> I hope I can rule my SDP logic based on standards rather than on how Chr=
ome implements some features.
>>>=20
>>> My question was generic: if I receive the above SDP, how do I know which=
 payloads each ssrc is supposed to transport?
>>>=20
>>>> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:
>>>> I'm not sure where it's specified, but here's where it's implemented:
>>>>=20
>>>> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/=
webrtcvideoengine.cc#4108
>>>>=20
>>>> It only supports 2 SSRCs for a FID group.  It would ignore any more tha=
n that.
>>>>=20
>>>>> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.n=
et> wrote:
>>>>> How is that? Where is that specified? What about if I include 3 ssrc v=
alues in the ssrc-group? What is each one for?
>>>>>=20
>>>>>> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>>>>>> 345259865 is "real"
>>>>>> 2693756249 is rtx
>>>>>>=20
>>>>>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.=
net> wrote:
>>>>>>> Hi,
>>>>>>>=20
>>>>>>> May I know which SSRC (345259865 or 2693756249) will be used for the=

>>>>>>> real media stream (plus RED and FEC) and which SSRC will be used for=

>>>>>>> RTX?
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> --------------------------
>>>>>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>>>>>> a=3Dmid:video
>>>>>>> a=3Drtpmap:100 VP8/90000
>>>>>>> a=3Drtpmap:116 red/90000
>>>>>>> a=3Drtpmap:117 ulpfec/90000
>>>>>>> a=3Drtpmap:96 rtx/90000
>>>>>>> a=3Dfmtp:96 apt=3D100
>>>>>>> a=3Dssrc-group:FID 345259865 2693756249
>>>>>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>>>>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>>>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>>> -------------------------------
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> RFC 5576 does not clarify it at all:
>>>>>>>=20
>>>>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>>>>=20
>>>>>>> --------------------------------------------------
>>>>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>>>>=20
>>>>>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>>>>>=20
>>>>>>>    [..]
>>>>>>>=20
>>>>>>>    The <semantics> parameter is taken from the specification of the
>>>>>>>    "group" attribute [RFC3388].  The initial semantic values defined=
 for
>>>>>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388=
]
>>>>>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>>>>    relationship among the grouped sources is the same as the
>>>>>>>    relationship among corresponding sources in media streams grouped=

>>>>>>>    using the SDP "group" attribute.
>>>>>>> --------------------------------------------------
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> The referenced RFC 3388 neither clarifies it:
>>>>>>>=20
>>>>>>> ---------------------------------------------------
>>>>>>> 7.4 FID Semantics
>>>>>>>=20
>>>>>>>    Several "m" lines grouped together using FID semantics form a med=
ia
>>>>>>>    flow.  A media agent handling a media flow that comprises several=
 "m"
>>>>>>>    lines MUST send a copy of the media to every "m" line part of the=

>>>>>>>    flow as long as the codecs and the direction attribute present in=
 a
>>>>>>>    particular "m" line allow it.
>>>>>>>=20
>>>>>>>    It is assumed that the application uses only one codec at a time t=
o
>>>>>>>    encode the media produced.  This codec MAY change dynamically dur=
ing
>>>>>>>    the session, but at any particular moment only one codec is in us=
e.
>>>>>>>=20
>>>>>>>    The application encodes the media using the current codec and che=
cks
>>>>>>>    one by one all the "m" lines that are part of the flow.  If a
>>>>>>>    particular "m" line contains the codec being used and the directi=
on
>>>>>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded medi=
a is
>>>>>>>    sent to the address/port specified in that particular media strea=
m.
>>>>>>>    If either the "m" line does not contain the codec being used or t=
he
>>>>>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing=
 is
>>>>>>>    sent over this media stream.
>>>>>>> ----------------------------------------------------
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>>>>=20
>>>>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>>>>=20
>>>>>>> How should the receiver interpret it?
>>>>>>>=20
>>>>>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>>>>>> the above SDP?
>>>>>>>=20
>>>>>>> Does not the above SDP look a complete mixture of hacks and workarou=
nds?
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> --
>>>>>>> I=C3=B1aki Baz Castillo
>>>>>>> <ibc@aliax.net>
>>>>>>>=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
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

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	charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>I agree. The SSRC-groups are included o=
nly to make it clear which FEC/RED/RTX streams go with which media streams i=
n the multiple source case. As long as payload types are distinct we have al=
l the information we need.<br><br></div><div>On Oct 22, 2014, at 12:16 PM, S=
uhas Nandakumar &lt;<a href=3D"mailto:suhasietf@gmail.com">suhasietf@gmail.c=
om</a>&gt; wrote:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"lt=
r">Just thinking loud&nbsp;<div>&nbsp;As long as there is enough information=
 to unabigously map the incoming streams to the SDP, i dont see we need any m=
ore information to be added.</div><div><br></div><div>I feel the combination=
 of SSRC and Payload Type in the RTP packet is good enough to find the mappi=
ng to the appropriate media line (if the PTs are not repeated) when needed.<=
/div><div><br></div><div>So in the initial examples above, ssrc-group commun=
icated in SDP along with semantics (FEC,FID) when applied to incoming SSRCs a=
nd the associated payload type has all the needed info for the receiver to d=
emux at the rtp level as well as map at the SDP level. Thus i dont see a nee=
d for specifying PT association explicitly in SDP for the ssrc-groups.</div>=
<div><br></div><div>Cheers</div><div>Suhas</div></div><div class=3D"gmail_ex=
tra"><br><div class=3D"gmail_quote">On Wed, Oct 22, 2014 at 12:01 PM, Bernar=
d Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" tar=
get=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><div dir=3D"auto"><div>SDP specifies payload types for RED/=
ULPFEC/RTX so as to allow differentiation within an ssrc-group. So while an f=
mtp: attribute could be used to denote a media stream on an a:ssrc line, it i=
s not necessary.</div><div><br></div><div>RFC 5176 Example 3 gives an exampl=
e with two cameras in which there are two ssrc-group lines (one for each cam=
era), with payload type used to differentiate between H.264 and RTX in each s=
src-group:</div><div><br></div><div><pre style=3D"margin-top:0px;margin-bott=
om:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:norm=
al;background-color:rgba(255,255,255,0)">
   m=3Dvideo 49174 RTP/AVPF 96 98&nbsp;</span></font></pre><pre style=3D"mar=
gin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span st=
yle=3D"white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=3Drt=
pmap:96 H.264/90000&nbsp;</span></font></pre><pre style=3D"margin-top:0px;ma=
rgin-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-s=
pace:normal;background-color:rgba(255,255,255,0)">&nbsp;a=3Drtpmap:98 rtx/90=
000&nbsp;</span></font></pre><pre style=3D"margin-top:0px;margin-bottom:0px"=
><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;back=
ground-color:rgba(255,255,255,0)">&nbsp;a=3Dfmtp:98 apt=3D96;rtx-time=3D3000=
&nbsp;</span></font></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><f=
ont face=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;backgro=
und-color:rgba(255,255,255,0)">&nbsp;a=3Dssrc-group:FID 11111 22222&nbsp;</s=
pan></font></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D=
"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-color:r=
gba(255,255,255,0)">&nbsp;a=3Dssrc:11111 cname:user3@<a href=3D"http://examp=
le.com" target=3D"_blank">example.com</a>&nbsp;</span></font></pre><pre styl=
e=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextStyleBody">=
<span style=3D"white-space:normal;background-color:rgba(255,255,255,0)">&nbs=
p;a=3Dssrc:22222 cname:user3@<a href=3D"http://example.com" target=3D"_blank=
">example.com</a>&nbsp;</span></font></pre><pre style=3D"margin-top:0px;marg=
in-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-spa=
ce:normal;background-color:rgba(255,255,255,0)">&nbsp;a=3Dssrc-group:FID 333=
33 44444&nbsp;</span></font></pre><pre style=3D"margin-top:0px;margin-bottom=
:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal=
;background-color:rgba(255,255,255,0)">&nbsp;a=3Dssrc:33333 cname:user3@<a h=
ref=3D"http://example.com" target=3D"_blank">example.com</a>&nbsp;</span></f=
ont></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICT=
FontTextStyleBody"><span style=3D"white-space:normal;background-color:rgba(2=
55,255,255,0)">&nbsp;a=3Dssrc:44444 cname:user3@<a href=3D"http://example.co=
m" target=3D"_blank">example.com</a></span></font><span style=3D"font-size:1=
em">
</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"U=
ICTFontTextStyleBody"><span style=3D"white-space:normal;background-color:rgb=
a(255,255,255,0)"><br></span></font></pre><pre style=3D"margin-top:0px;margi=
n-bottom:0px"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-spac=
e:normal;background-color:rgba(255,255,255,0)">I believe that RFC 5176 inclu=
des the two groups to make clear that there are two sources so that the Answ=
erer might choose to reject the m-line if it cannot handle that.&nbsp;</span=
></font><span style=3D"white-space:normal;background-color:rgba(255,255,255,=
0);font-family:UICTFontTextStyleBody">=46rom RFC 5176 Section 8:</span></pre=
><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTextS=
tyleBody"><span style=3D"white-space:normal;background-color:rgba(255,255,25=
5,0)"><br></span></font></pre><pre style=3D"margin-top:0px;margin-bottom:0px=
"><span style=3D"white-space:normal;background-color:rgba(255,255,255,0);fon=
t-family:UICTFontTextStyleBody">When used with the SDP Offer/Answer Model [<=
/span><a href=3D"http://tools.ietf.org/html/rfc3264" title=3D"&quot;An Offer=
/Answer Model with Session Description Protocol (SDP)&quot;" style=3D"white-=
space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextSt=
yleBody" target=3D"_blank">RFC3264</a><span style=3D"white-space:normal;back=
ground-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">], SDP s=
ource-specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer's only option
   is to reject the media stream or the entire multimedia session.</span></p=
re><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"UICTFontTex=
tStyleBody"><span style=3D"white-space:normal;background-color:rgba(255,255,=
255,0)"><br></span></font></pre><pre style=3D"margin-top:0px;margin-bottom:0=
px"><span style=3D"white-space:normal;background-color:rgba(255,255,255,0);f=
ont-family:UICTFontTextStyleBody">Note that in the example in RFC 4588 Secti=
on 8.8 where there is only one source, there are no ssrc-group lines at all:=
</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span style=3D"=
white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFont=
TextStyleBody"><br></span></pre><pre style=3D"margin-top:0px;margin-bottom:0=
px"><span style=3D"white-space:normal;background-color:rgba(255,255,255,0);f=
ont-family:UICTFontTextStyleBody">&nbsp;v=3D0&nbsp;</span></pre><pre style=3D=
"margin-top:0px;margin-bottom:0px"><span style=3D"white-space:normal;backgro=
und-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;o=3Dm=
ascha 2980675221 2980675778 IN IP4 <a href=3D"http://host.example.net" targe=
t=3D"_blank">host.example.net</a>&nbsp;</span></pre><pre style=3D"margin-top=
:0px;margin-bottom:0px"><span style=3D"white-space:normal;background-color:r=
gba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;c=3DIN IP4 192.0=
.2.0&nbsp;</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span=
 style=3D"white-space:normal;background-color:rgba(255,255,255,0);font-famil=
y:UICTFontTextStyleBody">&nbsp;m=3Dvideo 49170 RTP/AVPF 96 97&nbsp;</span></=
pre><pre style=3D"margin-top:0px;margin-bottom:0px"><span style=3D"white-spa=
ce:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyle=
Body">&nbsp;a=3Drtpmap:96 MP4V-ES/90000&nbsp;</span></pre><pre style=3D"marg=
in-top:0px;margin-bottom:0px"><span style=3D"white-space:normal;background-c=
olor:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=3Drtcp-f=
b:96 nack&nbsp;</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px">=
<span style=3D"white-space:normal;background-color:rgba(255,255,255,0);font-=
family:UICTFontTextStyleBody">&nbsp;a=3Dfmtp:96 profile-level-id=3D8;config=3D=
01010000012000884006682C2090A21F&nbsp;</span></pre><pre style=3D"margin-top:=
0px;margin-bottom:0px"><span style=3D"white-space:normal;background-color:rg=
ba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=3Drtpmap:97 rtx=
/90000&nbsp;</span></pre><pre style=3D"margin-top:0px;margin-bottom:0px"><sp=
an style=3D"white-space:normal;background-color:rgba(255,255,255,0);font-fam=
ily:UICTFontTextStyleBody">&nbsp;a=3Dfmtp:97 apt=3D96;rtx-time=3D3000</span>=
</pre></div><div><br></div><div>The situation would be the same for FEC grou=
ps created via ssrc-group:FEC. From&nbsp;</div><div><a href=3D"https://tools=
.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux" target=3D"_blank">https://=
tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux</a> &nbsp;:</div><div>=
<br></div><div><pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D=
"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-color:r=
gba(255,255,255,0)">      The FEC and the payload MAY also be multiplexed by=
 SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a href=3D"https://tools.ietf.org=
/html/rfc5576" title=3D"&quot;Source-Specific Media Attributes in the Sessio=
n Description Protocol (SDP)&quot;" target=3D"_blank">RFC5576</a>]; other me=
chanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style=3D"fo=
nt-size:1em">
</span></pre></div><div><div class=3D"h5"><div><br></div><div><br></div><div=
><br>On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"ma=
ilto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt; wrote:<br><br></=
div><blockquote type=3D"cite"><div><p dir=3D"ltr">I hope I can rule my SDP l=
ogic based on standards rather than on how Chrome implements some features.<=
/p>
<p dir=3D"ltr">My question was generic: if I receive the above SDP, how do I=
 know which payloads each ssrc is supposed to transport?</p>
<div class=3D"gmail_quote">On 21 Oct 2014 19:57, "Peter Thatcher" &lt;<a hre=
f=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>=
&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"=
ltr"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-=
serif">I'm not sure where it's specified, but here's where it's implemented:=
</div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans=
-serif"><br></div><div class=3D"gmail_default"><font face=3D"arial, helvetic=
a, sans-serif"><a href=3D"https://code.google.com/p/webrtc/source/browse/tru=
nk/talk/media/webrtc/webrtcvideoengine.cc#4108" target=3D"_blank">https://co=
de.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengi=
ne.cc#4108</a></font><br></div><div class=3D"gmail_default"><font face=3D"ar=
ial, helvetica, sans-serif"><br></font></div><div class=3D"gmail_default"><f=
ont face=3D"arial, helvetica, sans-serif">It only supports 2 SSRCs for a FID=
 group.&nbsp; It would ignore any more than that.</font></div></div><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 10=
:40 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@=
aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><p dir=3D"ltr">How is that? Where is that specified? W=
hat about if I include 3 ssrc values in the ssrc-group? What is each one for=
?</p><div><div>
<div class=3D"gmail_quote">On 21 Oct 2014 19:31, "Peter Thatcher" &lt;<a hre=
f=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>=
&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"=
ltr"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-=
serif"><span style=3D"font-family:arial,sans-serif;font-size:12.727272033691=
4px">345259865 is "real"</span><br></div><div class=3D"gmail_default"><font f=
ace=3D"arial, sans-serif"><a href=3D"tel:2693756249" value=3D"+12693756249" t=
arget=3D"_blank">2693756249</a> is rtx</font><br></div></div><div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Tue, Oct 21, 2014 at 9:25 AM, I=
=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net=
" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;paddi=
ng-left:1ex">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+126=
93756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+1269375624=
9" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank"=
>2693756249</a> label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank"=
>http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.&nbsp; The "ssrc-group" Media Attribute<br>
<br>
&nbsp; &nbsp;a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
&nbsp; &nbsp;[..]<br>
<br>
&nbsp; &nbsp;The &lt;semantics&gt; parameter is taken from the specification=
 of the<br>
&nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The initial semantic values d=
efined for<br>
&nbsp; &nbsp;the "ssrc-group" attribute are FID (Flow Identification) [RFC33=
88]<br>
&nbsp; &nbsp;and FEC (Forward Error Correction) [RFC4756].&nbsp; In each cas=
e, the<br>
&nbsp; &nbsp;relationship among the grouped sources is the same as the<br>
&nbsp; &nbsp;relationship among corresponding sources in media streams group=
ed<br>
&nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
&nbsp; &nbsp;Several "m" lines grouped together using FID semantics form a m=
edia<br>
&nbsp; &nbsp;flow.&nbsp; A media agent handling a media flow that comprises s=
everal "m"<br>
&nbsp; &nbsp;lines MUST send a copy of the media to every "m" line part of t=
he<br>
&nbsp; &nbsp;flow as long as the codecs and the direction attribute present i=
n a<br>
&nbsp; &nbsp;particular "m" line allow it.<br>
<br>
&nbsp; &nbsp;It is assumed that the application uses only one codec at a tim=
e to<br>
&nbsp; &nbsp;encode the media produced.&nbsp; This codec MAY change dynamica=
lly during<br>
&nbsp; &nbsp;the session, but at any particular moment only one codec is in u=
se.<br>
<br>
&nbsp; &nbsp;The application encodes the media using the current codec and c=
hecks<br>
&nbsp; &nbsp;one by one all the "m" lines that are part of the flow.&nbsp; I=
f a<br>
&nbsp; &nbsp;particular "m" line contains the codec being used and the direc=
tion<br>
&nbsp; &nbsp;attribute is "sendonly" or "sendrecv", a copy of the encoded me=
dia is<br>
&nbsp; &nbsp;sent to the address/port specified in that particular media str=
eam.<br>
&nbsp; &nbsp;If either the "m" line does not contain the codec being used or=
 the<br>
&nbsp; &nbsp;direction attribute is neither "sendonly" nor "sendrecv", nothi=
ng is<br>
&nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br>=

<span><font color=3D"#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
<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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>
</blockquote></div>
</div></div></blockquote></div><br></div>
</blockquote></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>=
</span><br><span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br><=
/div></blockquote></div></div></div><br>____________________________________=
___________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></blockquote></body></html>=

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From nobody Wed Oct 22 12:34:47 2014
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Date: Wed, 22 Oct 2014 21:34:41 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Hi,

I disagree, It is not possible just by looking at the SDP to know which 
ssrc is from the media and which is from rtx. While for WebRTC-endpoints 
it may be irrelevant because, as you say, you can wait for the RTP data 
to check the payload type and match the ssrc, we have seen that it is a 
must have for interoperability with ORTC.

Best regards
Sergio
On 22/10/2014 21:28, Bernard Aboba wrote:
> I agree. The SSRC-groups are included only to make it clear which 
> FEC/RED/RTX streams go with which media streams in the multiple source 
> case. As long as payload types are distinct we have all the 
> information we need.
>
> On Oct 22, 2014, at 12:16 PM, Suhas Nandakumar <suhasietf@gmail.com 
> <mailto:suhasietf@gmail.com>> wrote:
>
>> Just thinking loud
>>  As long as there is enough information to unabigously map the 
>> incoming streams to the SDP, i dont see we need any more information 
>> to be added.
>>
>> I feel the combination of SSRC and Payload Type in the RTP packet is 
>> good enough to find the mapping to the appropriate media line (if the 
>> PTs are not repeated) when needed.
>>
>> So in the initial examples above, ssrc-group communicated in SDP 
>> along with semantics (FEC,FID) when applied to incoming SSRCs and the 
>> associated payload type has all the needed info for the receiver to 
>> demux at the rtp level as well as map at the SDP level. Thus i dont 
>> see a need for specifying PT association explicitly in SDP for the 
>> ssrc-groups.
>>
>> Cheers
>> Suhas
>>
>> On Wed, Oct 22, 2014 at 12:01 PM, Bernard Aboba 
>> <bernard.aboba@gmail.com <mailto:bernard.aboba@gmail.com>> wrote:
>>
>>     SDP specifies payload types for RED/ULPFEC/RTX so as to allow
>>     differentiation within an ssrc-group. So while an fmtp: attribute
>>     could be used to denote a media stream on an a:ssrc line, it is
>>     not necessary.
>>
>>     RFC 5176 Example 3 gives an example with two cameras in which
>>     there are two ssrc-group lines (one for each camera), with
>>     payload type used to differentiate between H.264 and RTX in each
>>     ssrc-group:
>>
>>
>>         m=video 49174 RTP/AVPF 96 98
>>
>>       a=rtpmap:96 H.264/90000
>>
>>       a=rtpmap:98 rtx/90000
>>
>>       a=fmtp:98 apt=96;rtx-time=3000
>>
>>       a=ssrc-group:FID 11111 22222
>>
>>       a=ssrc:11111 cname:user3@example.com  <http://example.com>  
>>
>>       a=ssrc:22222 cname:user3@example.com  <http://example.com>  
>>
>>       a=ssrc-group:FID 33333 44444
>>
>>       a=ssrc:33333 cname:user3@example.com  <http://example.com>  
>>
>>       a=ssrc:44444 cname:user3@example.com  <http://example.com>
>>
>>
>>     I believe that RFC 5176 includes the two groups to make clear that there are two sources so that the Answerer might choose to reject the m-line if it cannot handle that. From RFC 5176 Section 8:
>>
>>
>>     When used with the SDP Offer/Answer Model [RFC3264  <http://tools.ietf.org/html/rfc3264>], SDP source-specific attributes describe only the sources that each party is
>>         willing to send (whether it is sending RTP data or RTCP report
>>         blocks).  No mechanism is provided by which an answer can accept or
>>         reject individual sources within a media stream; if the set of
>>         sources in a media stream is unacceptable, the answerer's only option
>>         is to reject the media stream or the entire multimedia session.
>>
>>
>>     Note that in the example in RFC 4588 Section 8.8 where there is only one source, there are no ssrc-group lines at all:
>>
>>
>>       v=0
>>
>>       o=mascha 2980675221 2980675778 IN IP4host.example.net  <http://host.example.net>  
>>
>>       c=IN IP4 192.0.2.0
>>
>>       m=video 49170 RTP/AVPF 96 97
>>
>>       a=rtpmap:96 MP4V-ES/90000
>>
>>       a=rtcp-fb:96 nack
>>
>>       a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F
>>
>>       a=rtpmap:97 rtx/90000
>>
>>       a=fmtp:97 apt=96;rtx-time=3000
>>
>>
>>     The situation would be the same for FEC groups created via
>>     ssrc-group:FEC. From
>>     https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
>>
>>            The FEC and the payload MAY also be multiplexed by SSRC into one
>>            single RTP session, with separate SSRC values, if the association
>>            between FEC and payload streams are communicated to all members of
>>            the session.  If SDP is used, this association MAY be communicated
>>            through the FEC ssrc-group semantic [RFC5576  <https://tools.ietf.org/html/rfc5576>]; other mechanisms
>>            are also possible.  Receivers MUST NOT attempt to interpret FEC
>>            streams for which they do not have information to associate them
>>            with the corresponding payload streams.
>>
>>
>>
>>
>>     On Oct 21, 2014, at 11:36 AM, Iñaki Baz Castillo <ibc@aliax.net
>>     <mailto:ibc@aliax.net>> wrote:
>>
>>>     I hope I can rule my SDP logic based on standards rather than on
>>>     how Chrome implements some features.
>>>
>>>     My question was generic: if I receive the above SDP, how do I
>>>     know which payloads each ssrc is supposed to transport?
>>>
>>>     On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com
>>>     <mailto:pthatcher@google.com>> wrote:
>>>
>>>         I'm not sure where it's specified, but here's where it's
>>>         implemented:
>>>
>>>         https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108
>>>
>>>         It only supports 2 SSRCs for a FID group.  It would ignore
>>>         any more than that.
>>>
>>>         On Tue, Oct 21, 2014 at 10:40 AM, Iñaki Baz Castillo
>>>         <ibc@aliax.net <mailto:ibc@aliax.net>> wrote:
>>>
>>>             How is that? Where is that specified? What about if I
>>>             include 3 ssrc values in the ssrc-group? What is each
>>>             one for?
>>>
>>>             On 21 Oct 2014 19:31, "Peter Thatcher"
>>>             <pthatcher@google.com <mailto:pthatcher@google.com>> wrote:
>>>
>>>                 345259865 is "real"
>>>                 2693756249 <tel:2693756249> is rtx
>>>
>>>                 On Tue, Oct 21, 2014 at 9:25 AM, Iñaki Baz Castillo
>>>                 <ibc@aliax.net <mailto:ibc@aliax.net>> wrote:
>>>
>>>                     Hi,
>>>
>>>                     May I know which SSRC (345259865 or 2693756249
>>>                     <tel:2693756249>) will be used for the
>>>                     real media stream (plus RED and FEC) and which
>>>                     SSRC will be used for
>>>                     RTX?
>>>
>>>
>>>
>>>                     --------------------------
>>>                     m=video 62164 RTP/SAVPF 100 116 117 96
>>>                     a=mid:video
>>>                     a=rtpmap:100 VP8/90000
>>>                     a=rtpmap:116 red/90000
>>>                     a=rtpmap:117 ulpfec/90000
>>>                     a=rtpmap:96 rtx/90000
>>>                     a=fmtp:96 apt=100
>>>                     a=ssrc-group:FID 345259865 2693756249
>>>                     <tel:2693756249>
>>>                     a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>>>                     a=ssrc:345259865
>>>                     msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>                     c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>                     a=ssrc:345259865
>>>                     mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>                     a=ssrc:345259865
>>>                     label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>                     a=ssrc:2693756249 <tel:2693756249>
>>>                     cname:erS7E/KHLYKTejNs
>>>                     a=ssrc:2693756249 <tel:2693756249>
>>>                     msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>                     c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>                     a=ssrc:2693756249 <tel:2693756249>
>>>                     mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>                     a=ssrc:2693756249 <tel:2693756249>
>>>                     label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>                     -------------------------------
>>>
>>>
>>>
>>>
>>>                     RFC 5576 does not clarify it at all:
>>>
>>>                     http://tools.ietf.org/html/rfc5576#section-4.2
>>>
>>>                     --------------------------------------------------
>>>                     4.2.  The "ssrc-group" Media Attribute
>>>
>>>                      a=ssrc-group:<semantics> <ssrc-id> ...
>>>
>>>                        [..]
>>>
>>>                        The <semantics> parameter is taken from the
>>>                     specification of the
>>>                        "group" attribute [RFC3388].  The initial
>>>                     semantic values defined for
>>>                        the "ssrc-group" attribute are FID (Flow
>>>                     Identification) [RFC3388]
>>>                        and FEC (Forward Error Correction)
>>>                     [RFC4756].  In each case, the
>>>                        relationship among the grouped sources is the
>>>                     same as the
>>>                        relationship among corresponding sources in
>>>                     media streams grouped
>>>                        using the SDP "group" attribute.
>>>                     --------------------------------------------------
>>>
>>>
>>>
>>>                     The referenced RFC 3388 neither clarifies it:
>>>
>>>                     ---------------------------------------------------
>>>                     7.4 FID Semantics
>>>
>>>                        Several "m" lines grouped together using FID
>>>                     semantics form a media
>>>                        flow.  A media agent handling a media flow
>>>                     that comprises several "m"
>>>                        lines MUST send a copy of the media to every
>>>                     "m" line part of the
>>>                        flow as long as the codecs and the direction
>>>                     attribute present in a
>>>                        particular "m" line allow it.
>>>
>>>                        It is assumed that the application uses only
>>>                     one codec at a time to
>>>                        encode the media produced.  This codec MAY
>>>                     change dynamically during
>>>                        the session, but at any particular moment
>>>                     only one codec is in use.
>>>
>>>                        The application encodes the media using the
>>>                     current codec and checks
>>>                        one by one all the "m" lines that are part of
>>>                     the flow.  If a
>>>                        particular "m" line contains the codec being
>>>                     used and the direction
>>>                        attribute is "sendonly" or "sendrecv", a copy
>>>                     of the encoded media is
>>>                        sent to the address/port specified in that
>>>                     particular media stream.
>>>                        If either the "m" line does not contain the
>>>                     codec being used or the
>>>                        direction attribute is neither "sendonly" nor
>>>                     "sendrecv", nothing is
>>>                        sent over this media stream.
>>>                     ----------------------------------------------------
>>>
>>>
>>>
>>>
>>>                     So, how is the usage of ssrc-group? Where is it
>>>                     really defined?
>>>
>>>                     Can I put more than 2 ssrc together in the same
>>>                     ssrc-group line?
>>>
>>>                     How should the receiver interpret it?
>>>
>>>                     Does a ssrc-group always mean RTX usage? Where
>>>                     is that specified in
>>>                     the above SDP?
>>>
>>>                     Does not the above SDP look a complete mixture
>>>                     of hacks and workarounds?
>>>
>>>
>>>
>>>
>>>                     --
>>>                     Iñaki Baz Castillo
>>>                     <ibc@aliax.net <mailto:ibc@aliax.net>>
>>>
>>>                     _______________________________________________
>>>                     rtcweb mailing list
>>>                     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>                     https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>>
>>>     _______________________________________________
>>>     rtcweb mailing list
>>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">Hi,<br>
      <br>
      I disagree, It is not possible just by looking at the SDP to know
      which ssrc is from the media and which is from rtx. While for
      WebRTC-endpoints it may be irrelevant because, as you say, you can
      wait for the RTP data to check the payload type and match the
      ssrc, we have seen that it is a must have for interoperability
      with ORTC.<br>
      <br>
      Best regards<br>
      Sergio<br>
      On 22/10/2014 21:28, Bernard Aboba wrote:<br>
    </div>
    <blockquote
      cite="mid:5A936EB2-EC94-49BD-AF11-E5A5141D77BE@gmail.com"
      type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <div>I agree. The SSRC-groups are included only to make it clear
        which FEC/RED/RTX streams go with which media streams in the
        multiple source case. As long as payload types are distinct we
        have all the information we need.<br>
        <br>
      </div>
      <div>On Oct 22, 2014, at 12:16 PM, Suhas Nandakumar &lt;<a
          moz-do-not-send="true" href="mailto:suhasietf@gmail.com">suhasietf@gmail.com</a>&gt;
        wrote:<br>
        <br>
      </div>
      <blockquote type="cite">
        <div>
          <div dir="ltr">Just thinking loud&nbsp;
            <div>&nbsp;As long as there is enough information to unabigously
              map the incoming streams to the SDP, i dont see we need
              any more information to be added.</div>
            <div><br>
            </div>
            <div>I feel the combination of SSRC and Payload Type in the
              RTP packet is good enough to find the mapping to the
              appropriate media line (if the PTs are not repeated) when
              needed.</div>
            <div><br>
            </div>
            <div>So in the initial examples above, ssrc-group
              communicated in SDP along with semantics (FEC,FID) when
              applied to incoming SSRCs and the associated payload type
              has all the needed info for the receiver to demux at the
              rtp level as well as map at the SDP level. Thus i dont see
              a need for specifying PT association explicitly in SDP for
              the ssrc-groups.</div>
            <div><br>
            </div>
            <div>Cheers</div>
            <div>Suhas</div>
          </div>
          <div class="gmail_extra"><br>
            <div class="gmail_quote">On Wed, Oct 22, 2014 at 12:01 PM,
              Bernard Aboba <span dir="ltr">&lt;<a
                  moz-do-not-send="true"
                  href="mailto:bernard.aboba@gmail.com" target="_blank">bernard.aboba@gmail.com</a>&gt;</span>
              wrote:<br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex">
                <div dir="auto">
                  <div>SDP specifies payload types for RED/ULPFEC/RTX so
                    as to allow differentiation within an ssrc-group. So
                    while an fmtp: attribute could be used to denote a
                    media stream on an a:ssrc line, it is not necessary.</div>
                  <div><br>
                  </div>
                  <div>RFC 5176 Example 3 gives an example with two
                    cameras in which there are two ssrc-group lines (one
                    for each camera), with payload type used to
                    differentiate between H.264 and RTX in each
                    ssrc-group:</div>
                  <div><br>
                  </div>
                  <div>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">
   m=video 49174 RTP/AVPF 96 98&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=rtpmap:96 H.264/90000&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=rtpmap:98 rtx/90000&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=fmtp:98 apt=96;rtx-time=3000&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc-group:FID 11111 22222&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc:11111 cname:user3@<a moz-do-not-send="true" href="http://example.com" target="_blank">example.com</a>&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc:22222 cname:user3@<a moz-do-not-send="true" href="http://example.com" target="_blank">example.com</a>&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc-group:FID 33333 44444&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc:33333 cname:user3@<a moz-do-not-send="true" href="http://example.com" target="_blank">example.com</a>&nbsp;</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">&nbsp;a=ssrc:44444 cname:user3@<a moz-do-not-send="true" href="http://example.com" target="_blank">example.com</a></span></font><span style="font-size:1em">
</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">I believe that RFC 5176 includes the two groups to make clear that there are two sources so that the Answerer might choose to reject the m-line if it cannot handle that.&nbsp;</span></font><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">From RFC 5176 Section 8:</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">When used with the SDP Offer/Answer Model [</span><a moz-do-not-send="true" href="http://tools.ietf.org/html/rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;" style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody" target="_blank">RFC3264</a><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">], SDP source-specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer's only option
   is to reject the media stream or the entire multimedia session.</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">Note that in the example in RFC 4588 Section 8.8 where there is only one source, there are no ssrc-group lines at all:</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">
</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;v=0&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;o=mascha 2980675221 2980675778 IN IP4 <a moz-do-not-send="true" href="http://host.example.net" target="_blank">host.example.net</a>&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;c=IN IP4 192.0.2.0&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;m=video 49170 RTP/AVPF 96 97&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=rtpmap:96 MP4V-ES/90000&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=rtcp-fb:96 nack&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=rtpmap:97 rtx/90000&nbsp;</span></pre>
                    <pre style="margin-top:0px;margin-bottom:0px"><span style="white-space:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">&nbsp;a=fmtp:97 apt=96;rtx-time=3000</span></pre>
                  </div>
                  <div><br>
                  </div>
                  <div>The situation would be the same for FEC groups
                    created via ssrc-group:FEC. From&nbsp;</div>
                  <div><a moz-do-not-send="true"
                      href="https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux"
                      target="_blank">https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux</a>
                    &nbsp;:</div>
                  <div><br>
                  </div>
                  <div>
                    <pre style="margin-top:0px;margin-bottom:0px"><font face="UICTFontTextStyleBody"><span style="white-space:normal;background-color:rgba(255,255,255,0)">      The FEC and the payload MAY also be multiplexed by SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a moz-do-not-send="true" href="https://tools.ietf.org/html/rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;" target="_blank">RFC5576</a>]; other mechanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style="font-size:1em">
</span></pre>
                  </div>
                  <div>
                    <div class="h5">
                      <div><br>
                      </div>
                      <div><br>
                      </div>
                      <div><br>
                        On Oct 21, 2014, at 11:36 AM, I&ntilde;aki Baz Castillo
                        &lt;<a moz-do-not-send="true"
                          href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;
                        wrote:<br>
                        <br>
                      </div>
                      <blockquote type="cite">
                        <div>
                          <p dir="ltr">I hope I can rule my SDP logic
                            based on standards rather than on how Chrome
                            implements some features.</p>
                          <p dir="ltr">My question was generic: if I
                            receive the above SDP, how do I know which
                            payloads each ssrc is supposed to transport?</p>
                          <div class="gmail_quote">On 21 Oct 2014 19:57,
                            "Peter Thatcher" &lt;<a
                              moz-do-not-send="true"
                              href="mailto:pthatcher@google.com"
                              target="_blank">pthatcher@google.com</a>&gt;
                            wrote:<br type="attribution">
                            <blockquote class="gmail_quote"
                              style="margin:0 0 0 .8ex;border-left:1px
                              #ccc solid;padding-left:1ex">
                              <div dir="ltr">
                                <div class="gmail_default"
                                  style="font-family:arial,helvetica,sans-serif">I'm
                                  not sure where it's specified, but
                                  here's where it's implemented:</div>
                                <div class="gmail_default"
                                  style="font-family:arial,helvetica,sans-serif"><br>
                                </div>
                                <div class="gmail_default"><font
                                    face="arial, helvetica, sans-serif"><a
                                      moz-do-not-send="true"
href="https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108"
                                      target="_blank">https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108</a></font><br>
                                </div>
                                <div class="gmail_default"><font
                                    face="arial, helvetica, sans-serif"><br>
                                  </font></div>
                                <div class="gmail_default"><font
                                    face="arial, helvetica, sans-serif">It
                                    only supports 2 SSRCs for a FID
                                    group.&nbsp; It would ignore any more
                                    than that.</font></div>
                              </div>
                              <div class="gmail_extra"><br>
                                <div class="gmail_quote">On Tue, Oct 21,
                                  2014 at 10:40 AM, I&ntilde;aki Baz Castillo <span
                                    dir="ltr">&lt;<a
                                      moz-do-not-send="true"
                                      href="mailto:ibc@aliax.net"
                                      target="_blank">ibc@aliax.net</a>&gt;</span>
                                  wrote:<br>
                                  <blockquote class="gmail_quote"
                                    style="margin:0 0 0
                                    .8ex;border-left:1px #ccc
                                    solid;padding-left:1ex">
                                    <p dir="ltr">How is that? Where is
                                      that specified? What about if I
                                      include 3 ssrc values in the
                                      ssrc-group? What is each one for?</p>
                                    <div>
                                      <div>
                                        <div class="gmail_quote">On 21
                                          Oct 2014 19:31, "Peter
                                          Thatcher" &lt;<a
                                            moz-do-not-send="true"
                                            href="mailto:pthatcher@google.com"
                                            target="_blank">pthatcher@google.com</a>&gt;
                                          wrote:<br type="attribution">
                                          <blockquote
                                            class="gmail_quote"
                                            style="margin:0 0 0
                                            .8ex;border-left:1px #ccc
                                            solid;padding-left:1ex">
                                            <div dir="ltr">
                                              <div class="gmail_default"
style="font-family:arial,helvetica,sans-serif"><span
                                                  style="font-family:arial,sans-serif;font-size:12.7272720336914px">345259865
                                                  is "real"</span><br>
                                              </div>
                                              <div class="gmail_default"><font
                                                  face="arial,
                                                  sans-serif"><a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>
                                                  is rtx</font><br>
                                              </div>
                                            </div>
                                            <div class="gmail_extra"><br>
                                              <div class="gmail_quote">On
                                                Tue, Oct 21, 2014 at
                                                9:25 AM, I&ntilde;aki Baz
                                                Castillo <span
                                                  dir="ltr">&lt;<a
                                                    moz-do-not-send="true"
href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;</span>
                                                wrote:<br>
                                                <blockquote
                                                  class="gmail_quote"
                                                  style="margin:0 0 0
                                                  .8ex;border-left:1px
                                                  #ccc
                                                  solid;padding-left:1ex">Hi,<br>
                                                  <br>
                                                  May I know which SSRC
                                                  (345259865 or <a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>)
                                                  will be used for the<br>
                                                  real media stream
                                                  (plus RED and FEC) and
                                                  which SSRC will be
                                                  used for<br>
                                                  RTX?<br>
                                                  <br>
                                                  <br>
                                                  <br>
--------------------------<br>
                                                  m=video 62164
                                                  RTP/SAVPF 100 116 117
                                                  96<br>
                                                  a=mid:video<br>
                                                  a=rtpmap:100 VP8/90000<br>
                                                  a=rtpmap:116 red/90000<br>
                                                  a=rtpmap:117
                                                  ulpfec/90000<br>
                                                  a=rtpmap:96 rtx/90000<br>
                                                  a=fmtp:96 apt=100<br>
                                                  a=ssrc-group:FID
                                                  345259865 <a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a><br>
                                                  a=ssrc:345259865
                                                  cname:erS7E/KHLYKTejNs<br>
                                                  a=ssrc:345259865
                                                  msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                                  a=ssrc:345259865
                                                  mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                                  a=ssrc:345259865
                                                  <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
                                                  a=ssrc:<a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>
                                                  cname:erS7E/KHLYKTejNs<br>
                                                  a=ssrc:<a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>
msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                                  a=ssrc:<a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>
mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                                  a=ssrc:<a
                                                    moz-do-not-send="true"
href="tel:2693756249" value="+12693756249" target="_blank">2693756249</a>
<a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
-------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  RFC 5576 does not
                                                  clarify it at all:<br>
                                                  <br>
                                                  <a
                                                    moz-do-not-send="true"
href="http://tools.ietf.org/html/rfc5576#section-4.2" target="_blank">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
                                                  <br>
--------------------------------------------------<br>
                                                  4.2.&nbsp; The "ssrc-group"
                                                  Media Attribute<br>
                                                  <br>
                                                  &nbsp;
                                                  &nbsp;a=ssrc-group:&lt;semantics&gt;
                                                  &lt;ssrc-id&gt; ...<br>
                                                  <br>
                                                  &nbsp; &nbsp;[..]<br>
                                                  <br>
                                                  &nbsp; &nbsp;The
                                                  &lt;semantics&gt;
                                                  parameter is taken
                                                  from the specification
                                                  of the<br>
                                                  &nbsp; &nbsp;"group" attribute
                                                  [RFC3388].&nbsp; The
                                                  initial semantic
                                                  values defined for<br>
                                                  &nbsp; &nbsp;the "ssrc-group"
                                                  attribute are FID
                                                  (Flow Identification)
                                                  [RFC3388]<br>
                                                  &nbsp; &nbsp;and FEC (Forward
                                                  Error Correction)
                                                  [RFC4756].&nbsp; In each
                                                  case, the<br>
                                                  &nbsp; &nbsp;relationship among
                                                  the grouped sources is
                                                  the same as the<br>
                                                  &nbsp; &nbsp;relationship among
                                                  corresponding sources
                                                  in media streams
                                                  grouped<br>
                                                  &nbsp; &nbsp;using the SDP
                                                  "group" attribute.<br>
--------------------------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  The referenced RFC
                                                  3388 neither clarifies
                                                  it:<br>
                                                  <br>
---------------------------------------------------<br>
                                                  7.4 FID Semantics<br>
                                                  <br>
                                                  &nbsp; &nbsp;Several "m" lines
                                                  grouped together using
                                                  FID semantics form a
                                                  media<br>
                                                  &nbsp; &nbsp;flow.&nbsp; A media
                                                  agent handling a media
                                                  flow that comprises
                                                  several "m"<br>
                                                  &nbsp; &nbsp;lines MUST send a
                                                  copy of the media to
                                                  every "m" line part of
                                                  the<br>
                                                  &nbsp; &nbsp;flow as long as the
                                                  codecs and the
                                                  direction attribute
                                                  present in a<br>
                                                  &nbsp; &nbsp;particular "m" line
                                                  allow it.<br>
                                                  <br>
                                                  &nbsp; &nbsp;It is assumed that
                                                  the application uses
                                                  only one codec at a
                                                  time to<br>
                                                  &nbsp; &nbsp;encode the media
                                                  produced.&nbsp; This codec
                                                  MAY change dynamically
                                                  during<br>
                                                  &nbsp; &nbsp;the session, but at
                                                  any particular moment
                                                  only one codec is in
                                                  use.<br>
                                                  <br>
                                                  &nbsp; &nbsp;The application
                                                  encodes the media
                                                  using the current
                                                  codec and checks<br>
                                                  &nbsp; &nbsp;one by one all the
                                                  "m" lines that are
                                                  part of the flow.&nbsp; If
                                                  a<br>
                                                  &nbsp; &nbsp;particular "m" line
                                                  contains the codec
                                                  being used and the
                                                  direction<br>
                                                  &nbsp; &nbsp;attribute is
                                                  "sendonly" or
                                                  "sendrecv", a copy of
                                                  the encoded media is<br>
                                                  &nbsp; &nbsp;sent to the
                                                  address/port specified
                                                  in that particular
                                                  media stream.<br>
                                                  &nbsp; &nbsp;If either the "m"
                                                  line does not contain
                                                  the codec being used
                                                  or the<br>
                                                  &nbsp; &nbsp;direction attribute
                                                  is neither "sendonly"
                                                  nor "sendrecv",
                                                  nothing is<br>
                                                  &nbsp; &nbsp;sent over this
                                                  media stream.<br>
----------------------------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  So, how is the usage
                                                  of ssrc-group? Where
                                                  is it really defined?<br>
                                                  <br>
                                                  Can I put more than 2
                                                  ssrc together in the
                                                  same ssrc-group line?<br>
                                                  <br>
                                                  How should the
                                                  receiver interpret it?<br>
                                                  <br>
                                                  Does a ssrc-group
                                                  always mean RTX usage?
                                                  Where is that
                                                  specified in<br>
                                                  the above SDP?<br>
                                                  <br>
                                                  Does not the above SDP
                                                  look a complete
                                                  mixture of hacks and
                                                  workarounds?<br>
                                                  <span><font
                                                      color="#888888"><br>
                                                      <br>
                                                      <br>
                                                      <br>
                                                      --<br>
                                                      I&ntilde;aki Baz Castillo<br>
                                                      &lt;<a
                                                        moz-do-not-send="true"
href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>&gt;<br>
                                                      <br>
_______________________________________________<br>
                                                      rtcweb mailing
                                                      list<br>
                                                      <a
                                                        moz-do-not-send="true"
href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
                                                      <a
                                                        moz-do-not-send="true"
href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                                                    </font></span></blockquote>
                                              </div>
                                              <br>
                                            </div>
                                          </blockquote>
                                        </div>
                                      </div>
                                    </div>
                                  </blockquote>
                                </div>
                                <br>
                              </div>
                            </blockquote>
                          </div>
                        </div>
                      </blockquote>
                      <blockquote type="cite">
                        <div><span>_______________________________________________</span><br>
                          <span>rtcweb mailing list</span><br>
                          <span><a moz-do-not-send="true"
                              href="mailto:rtcweb@ietf.org"
                              target="_blank">rtcweb@ietf.org</a></span><br>
                          <span><a moz-do-not-send="true"
                              href="https://www.ietf.org/mailman/listinfo/rtcweb"
                              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br>
                        </div>
                      </blockquote>
                    </div>
                  </div>
                </div>
                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                <br>
              </blockquote>
            </div>
            <br>
          </div>
        </div>
      </blockquote>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
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From: Peter Thatcher <pthatcher@google.com>
Date: Wed, 22 Oct 2014 12:39:50 -0700
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Most of the discussion here has been about having enough information to
receive packets.  But there's another side to it in the rtcweb case.  When
viewed as an API surface, having SDP that the JS gives to the browser which
means "use one of these for RTX, and one for the normal media flow, but you
pick which" is suboptimal.  As an API surface, it would be much better to
allow the application to be explicit about which SSRC it wants for which.



On Wed, Oct 22, 2014 at 12:34 PM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

>  Hi,
>
> I disagree, It is not possible just by looking at the SDP to know which
> ssrc is from the media and which is from rtx. While for WebRTC-endpoints =
it
> may be irrelevant because, as you say, you can wait for the RTP data to
> check the payload type and match the ssrc, we have seen that it is a must
> have for interoperability with ORTC.
>
> Best regards
> Sergio
>
> On 22/10/2014 21:28, Bernard Aboba wrote:
>
> I agree. The SSRC-groups are included only to make it clear which
> FEC/RED/RTX streams go with which media streams in the multiple source
> case. As long as payload types are distinct we have all the information w=
e
> need.
>
>  On Oct 22, 2014, at 12:16 PM, Suhas Nandakumar <suhasietf@gmail.com>
> wrote:
>
>   Just thinking loud
>  As long as there is enough information to unabigously map the incoming
> streams to the SDP, i dont see we need any more information to be added.
>
>  I feel the combination of SSRC and Payload Type in the RTP packet is
> good enough to find the mapping to the appropriate media line (if the PTs
> are not repeated) when needed.
>
>  So in the initial examples above, ssrc-group communicated in SDP along
> with semantics (FEC,FID) when applied to incoming SSRCs and the associate=
d
> payload type has all the needed info for the receiver to demux at the rtp
> level as well as map at the SDP level. Thus i dont see a need for
> specifying PT association explicitly in SDP for the ssrc-groups.
>
>  Cheers
> Suhas
>
> On Wed, Oct 22, 2014 at 12:01 PM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>>  SDP specifies payload types for RED/ULPFEC/RTX so as to allow
>> differentiation within an ssrc-group. So while an fmtp: attribute could =
be
>> used to denote a media stream on an a:ssrc line, it is not necessary.
>>
>>  RFC 5176 Example 3 gives an example with two cameras in which there are
>> two ssrc-group lines (one for each camera), with payload type used to
>> differentiate between H.264 and RTX in each ssrc-group:
>>
>>
>>    m=3Dvideo 49174 RTP/AVPF 96 98
>>
>>  a=3Drtpmap:96 H.264/90000
>>
>>  a=3Drtpmap:98 rtx/90000
>>
>>  a=3Dfmtp:98 apt=3D96;rtx-time=3D3000
>>
>>  a=3Dssrc-group:FID 11111 22222
>>
>>  a=3Dssrc:11111 cname:user3@example.com
>>
>>  a=3Dssrc:22222 cname:user3@example.com
>>
>>  a=3Dssrc-group:FID 33333 44444
>>
>>  a=3Dssrc:33333 cname:user3@example.com
>>
>>  a=3Dssrc:44444 cname:user3@example.com
>>
>>  I believe that RFC 5176 includes the two groups to make clear that ther=
e are two sources so that the Answerer might choose to reject the m-line if=
 it cannot handle that. From RFC 5176 Section 8:
>>
>>  When used with the SDP Offer/Answer Model [RFC3264 <http://tools.ietf.o=
rg/html/rfc3264>], SDP source-specific attributes describe only the sources=
 that each party is
>>    willing to send (whether it is sending RTP data or RTCP report
>>    blocks).  No mechanism is provided by which an answer can accept or
>>    reject individual sources within a media stream; if the set of
>>    sources in a media stream is unacceptable, the answerer's only option
>>    is to reject the media stream or the entire multimedia session.
>>
>>  Note that in the example in RFC 4588 Section 8.8 where there is only on=
e source, there are no ssrc-group lines at all:
>>
>>   v=3D0
>>
>>  o=3Dmascha 2980675221 2980675778 IN IP4 host.example.net
>>
>>  c=3DIN IP4 192.0.2.0
>>
>>  m=3Dvideo 49170 RTP/AVPF 96 97
>>
>>  a=3Drtpmap:96 MP4V-ES/90000
>>
>>  a=3Drtcp-fb:96 nack
>>
>>  a=3Dfmtp:96 profile-level-id=3D8;config=3D01010000012000884006682C2090A=
21F
>>
>>  a=3Drtpmap:97 rtx/90000
>>
>>  a=3Dfmtp:97 apt=3D96;rtx-time=3D3000
>>
>>
>>  The situation would be the same for FEC groups created via
>> ssrc-group:FEC. From
>> https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
>>
>>        The FEC and the payload MAY also be multiplexed by SSRC into one
>>       single RTP session, with separate SSRC values, if the association
>>       between FEC and payload streams are communicated to all members of
>>       the session.  If SDP is used, this association MAY be communicated
>>       through the FEC ssrc-group semantic [RFC5576 <https://tools.ietf.o=
rg/html/rfc5576>]; other mechanisms
>>       are also possible.  Receivers MUST NOT attempt to interpret FEC
>>       streams for which they do not have information to associate them
>>       with the corresponding payload streams.
>>
>>
>>
>>
>> On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:
>>
>>   I hope I can rule my SDP logic based on standards rather than on how
>> Chrome implements some features.
>>
>> My question was generic: if I receive the above SDP, how do I know which
>> payloads each ssrc is supposed to transport?
>> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:
>>
>>>  I'm not sure where it's specified, but here's where it's implemented:
>>>
>>>
>>> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/=
webrtcvideoengine.cc#4108
>>>
>>>  It only supports 2 SSRCs for a FID group.  It would ignore any more
>>> than that.
>>>
>>> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.ne=
t>
>>> wrote:
>>>
>>>> How is that? Where is that specified? What about if I include 3 ssrc
>>>> values in the ssrc-group? What is each one for?
>>>>  On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
>>>>
>>>>>  345259865 is "real"
>>>>>  2693756249 is rtx
>>>>>
>>>>> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.n=
et>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>>>>> RTX?
>>>>>>
>>>>>>
>>>>>>
>>>>>> --------------------------
>>>>>> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
>>>>>> a=3Dmid:video
>>>>>> a=3Drtpmap:100 VP8/90000
>>>>>> a=3Drtpmap:116 red/90000
>>>>>> a=3Drtpmap:117 ulpfec/90000
>>>>>> a=3Drtpmap:96 rtx/90000
>>>>>> a=3Dfmtp:96 apt=3D100
>>>>>> a=3Dssrc-group:FID 345259865 2693756249
>>>>>> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
>>>>>> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>>>> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>>> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>>> -------------------------------
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> RFC 5576 does not clarify it at all:
>>>>>>
>>>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>>>
>>>>>> --------------------------------------------------
>>>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>>>
>>>>>>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>>>>>>
>>>>>>    [..]
>>>>>>
>>>>>>    The <semantics> parameter is taken from the specification of the
>>>>>>    "group" attribute [RFC3388].  The initial semantic values defined
>>>>>> for
>>>>>>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388=
]
>>>>>>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>>>    relationship among the grouped sources is the same as the
>>>>>>    relationship among corresponding sources in media streams grouped
>>>>>>    using the SDP "group" attribute.
>>>>>> --------------------------------------------------
>>>>>>
>>>>>>
>>>>>>
>>>>>> The referenced RFC 3388 neither clarifies it:
>>>>>>
>>>>>> ---------------------------------------------------
>>>>>> 7.4 FID Semantics
>>>>>>
>>>>>>    Several "m" lines grouped together using FID semantics form a med=
ia
>>>>>>    flow.  A media agent handling a media flow that comprises several
>>>>>> "m"
>>>>>>    lines MUST send a copy of the media to every "m" line part of the
>>>>>>    flow as long as the codecs and the direction attribute present in=
 a
>>>>>>    particular "m" line allow it.
>>>>>>
>>>>>>    It is assumed that the application uses only one codec at a time =
to
>>>>>>    encode the media produced.  This codec MAY change dynamically
>>>>>> during
>>>>>>    the session, but at any particular moment only one codec is in us=
e.
>>>>>>
>>>>>>    The application encodes the media using the current codec and
>>>>>> checks
>>>>>>    one by one all the "m" lines that are part of the flow.  If a
>>>>>>    particular "m" line contains the codec being used and the directi=
on
>>>>>>    attribute is "sendonly" or "sendrecv", a copy of the encoded medi=
a
>>>>>> is
>>>>>>    sent to the address/port specified in that particular media strea=
m.
>>>>>>    If either the "m" line does not contain the codec being used or t=
he
>>>>>>    direction attribute is neither "sendonly" nor "sendrecv", nothing
>>>>>> is
>>>>>>    sent over this media stream.
>>>>>> ----------------------------------------------------
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>>>
>>>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>>>
>>>>>> How should the receiver interpret it?
>>>>>>
>>>>>> Does a ssrc-group always mean RTX usage? Where is that specified in
>>>>>> the above SDP?
>>>>>>
>>>>>> Does not the above SDP look a complete mixture of hacks and
>>>>>> workarounds?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> I=C3=B1aki Baz Castillo
>>>>>> <ibc@aliax.net>
>>>>>>
>>>>>> _______________________________________________
>>>>>> rtcweb mailing list
>>>>>> rtcweb@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>
>>>>>
>>>>>
>>>    _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/r=
tcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c3cc02fe3a280506081ea8
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">Most of the discussion here has been about having enoug=
h information to receive packets.=C2=A0 But there&#39;s another side to it =
in the rtcweb case.=C2=A0 When viewed as an API surface, having SDP that th=
e JS gives to the browser which means &quot;use one of these for RTX, and o=
ne for the normal media flow, but you pick which&quot; is suboptimal.=C2=A0=
 As an API surface, it would be much better to allow the application to be =
explicit about which SSRC it wants for which.</div><div class=3D"gmail_defa=
ult" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">=C2=A0=
=C2=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Wed, Oct 22, 2014 at 12:34 PM, Sergio Garcia Murillo <span dir=3D"ltr">=
&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">se=
rgio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>Hi,<br>
      <br>
      I disagree, It is not possible just by looking at the SDP to know
      which ssrc is from the media and which is from rtx. While for
      WebRTC-endpoints it may be irrelevant because, as you say, you can
      wait for the RTP data to check the payload type and match the
      ssrc, we have seen that it is a must have for interoperability
      with ORTC.<br>
      <br>
      Best regards<span class=3D"HOEnZb"><font color=3D"#888888"><br>
      Sergio</font></span><div><div class=3D"h5"><br>
      On 22/10/2014 21:28, Bernard Aboba wrote:<br>
    </div></div></div><div><div class=3D"h5">
    <blockquote type=3D"cite">
     =20
      <div>I agree. The SSRC-groups are included only to make it clear
        which FEC/RED/RTX streams go with which media streams in the
        multiple source case. As long as payload types are distinct we
        have all the information we need.<br>
        <br>
      </div>
      <div>On Oct 22, 2014, at 12:16 PM, Suhas Nandakumar &lt;<a href=3D"ma=
ilto:suhasietf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>&gt;
        wrote:<br>
        <br>
      </div>
      <blockquote type=3D"cite">
        <div>
          <div dir=3D"ltr">Just thinking loud=C2=A0
            <div>=C2=A0As long as there is enough information to unabigousl=
y
              map the incoming streams to the SDP, i dont see we need
              any more information to be added.</div>
            <div><br>
            </div>
            <div>I feel the combination of SSRC and Payload Type in the
              RTP packet is good enough to find the mapping to the
              appropriate media line (if the PTs are not repeated) when
              needed.</div>
            <div><br>
            </div>
            <div>So in the initial examples above, ssrc-group
              communicated in SDP along with semantics (FEC,FID) when
              applied to incoming SSRCs and the associated payload type
              has all the needed info for the receiver to demux at the
              rtp level as well as map at the SDP level. Thus i dont see
              a need for specifying PT association explicitly in SDP for
              the ssrc-groups.</div>
            <div><br>
            </div>
            <div>Cheers</div>
            <div>Suhas</div>
          </div>
          <div class=3D"gmail_extra"><br>
            <div class=3D"gmail_quote">On Wed, Oct 22, 2014 at 12:01 PM,
              Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard=
.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span>
              wrote:<br>
              <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
                <div dir=3D"auto">
                  <div>SDP specifies payload types for RED/ULPFEC/RTX so
                    as to allow differentiation within an ssrc-group. So
                    while an fmtp: attribute could be used to denote a
                    media stream on an a:ssrc line, it is not necessary.</d=
iv>
                  <div><br>
                  </div>
                  <div>RFC 5176 Example 3 gives an example with two
                    cameras in which there are two ssrc-group lines (one
                    for each camera), with payload type used to
                    differentiate between H.264 and RTX in each
                    ssrc-group:</div>
                  <div><br>
                  </div>
                  <div>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">
   m=3Dvideo 49174 RTP/AVPF 96 98=C2=A0</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Drtpmap:96 H.264/90000=C2=A0</span></fo=
nt></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Drtpmap:98 rtx/90000=C2=A0</span></font=
></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dfmtp:98 apt=3D96;rtx-time=3D3000=C2=A0=
</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc-group:FID 11111 22222=C2=A0</span=
></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:11111 cname:user3@<a href=3D"http=
://example.com" target=3D"_blank">example.com</a>=C2=A0</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:22222 cname:user3@<a href=3D"http=
://example.com" target=3D"_blank">example.com</a>=C2=A0</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc-group:FID 33333 44444=C2=A0</span=
></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:33333 cname:user3@<a href=3D"http=
://example.com" target=3D"_blank">example.com</a>=C2=A0</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">=C2=A0a=3Dssrc:44444 cname:user3@<a href=3D"http=
://example.com" target=3D"_blank">example.com</a></span></font><span style=
=3D"font-size:1em">
</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">I believe that RFC 5176 includes the two groups =
to make clear that there are two sources so that the Answerer might choose =
to reject the m-line if it cannot handle that.=C2=A0</span></font><span sty=
le=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family:U=
ICTFontTextStyleBody">From RFC 5176 Section 8:</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">When used with the SDP Offer/Answer Model [</span><=
a href=3D"http://tools.ietf.org/html/rfc3264" title=3D"&quot;An Offer/Answe=
r Model with Session Description Protocol (SDP)&quot;" style=3D"white-space=
:normal;background-color:rgba(255,255,255,0);font-family:UICTFontTextStyleB=
ody" target=3D"_blank">RFC3264</a><span style=3D"white-space:normal;backgro=
und-color:rgba(255,255,255,0);font-family:UICTFontTextStyleBody">], SDP sou=
rce-specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer&#39;s only optio=
n
   is to reject the media stream or the entire multimedia session.</span></=
pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">
</span></font></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">Note that in the example in RFC 4588 Section 8.8 wh=
ere there is only one source, there are no ssrc-group lines at all:</span><=
/pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">
</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0v=3D0=C2=A0</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0o=3Dmascha 2980675221 2980675778 IN IP4 <a hr=
ef=3D"http://host.example.net" target=3D"_blank">host.example.net</a>=C2=A0=
</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0c=3DIN IP4 192.0.2.0=C2=A0</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0m=3Dvideo 49170 RTP/AVPF 96 97=C2=A0</span></=
pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0a=3Drtpmap:96 MP4V-ES/90000=C2=A0</span></pre=
>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0a=3Drtcp-fb:96 nack=C2=A0</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0a=3Dfmtp:96 profile-level-id=3D8;config=3D010=
10000012000884006682C2090A21F=C2=A0</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0a=3Drtpmap:97 rtx/90000=C2=A0</span></pre>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><span s=
tyle=3D"white-space:normal;background-color:rgba(255,255,255,0);font-family=
:UICTFontTextStyleBody">=C2=A0a=3Dfmtp:97 apt=3D96;rtx-time=3D3000</span></=
pre>
                  </div>
                  <div><br>
                  </div>
                  <div>The situation would be the same for FEC groups
                    created via ssrc-group:FEC. From=C2=A0</div>
                  <div><a href=3D"https://tools.ietf.org/html/draft-lennox-=
payload-ulp-ssrc-mux" target=3D"_blank">https://tools.ietf.org/html/draft-l=
ennox-payload-ulp-ssrc-mux</a>
                    =C2=A0:</div>
                  <div><br>
                  </div>
                  <div>
                    <pre style=3D"margin-top:0px;margin-bottom:0px"><font f=
ace=3D"UICTFontTextStyleBody"><span style=3D"white-space:normal;background-=
color:rgba(255,255,255,0)">      The FEC and the payload MAY also be multip=
lexed by SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a href=3D"https://tools.ietf.or=
g/html/rfc5576" title=3D"&quot;Source-Specific Media Attributes in the Sess=
ion Description Protocol (SDP)&quot;" target=3D"_blank">RFC5576</a>]; other=
 mechanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style=3D"f=
ont-size:1em">
</span></pre>
                  </div>
                  <div>
                    <div>
                      <div><br>
                      </div>
                      <div><br>
                      </div>
                      <div><br>
                        On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castil=
lo
                        &lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_bla=
nk">ibc@aliax.net</a>&gt;
                        wrote:<br>
                        <br>
                      </div>
                      <blockquote type=3D"cite">
                        <div>
                          <p dir=3D"ltr">I hope I can rule my SDP logic
                            based on standards rather than on how Chrome
                            implements some features.</p>
                          <p dir=3D"ltr">My question was generic: if I
                            receive the above SDP, how do I know which
                            payloads each ssrc is supposed to transport?</p=
>
                          <div class=3D"gmail_quote">On 21 Oct 2014 19:57,
                            &quot;Peter Thatcher&quot; &lt;<a href=3D"mailt=
o:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt;
                            wrote:<br type=3D"attribution">
                            <blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                              <div dir=3D"ltr">
                                <div class=3D"gmail_default" style=3D"font-=
family:arial,helvetica,sans-serif">I&#39;m
                                  not sure where it&#39;s specified, but
                                  here&#39;s where it&#39;s implemented:</d=
iv>
                                <div class=3D"gmail_default" style=3D"font-=
family:arial,helvetica,sans-serif"><br>
                                </div>
                                <div class=3D"gmail_default"><font face=3D"=
arial, helvetica, sans-serif"><a href=3D"https://code.google.com/p/webrtc/s=
ource/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108" target=3D"_=
blank">https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webr=
tc/webrtcvideoengine.cc#4108</a></font><br>
                                </div>
                                <div class=3D"gmail_default"><font face=3D"=
arial, helvetica, sans-serif"><br>
                                  </font></div>
                                <div class=3D"gmail_default"><font face=3D"=
arial, helvetica, sans-serif">It
                                    only supports 2 SSRCs for a FID
                                    group.=C2=A0 It would ignore any more
                                    than that.</font></div>
                              </div>
                              <div class=3D"gmail_extra"><br>
                                <div class=3D"gmail_quote">On Tue, Oct 21,
                                  2014 at 10:40 AM, I=C3=B1aki Baz Castillo=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">i=
bc@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">
                                    <p dir=3D"ltr">How is that? Where is
                                      that specified? What about if I
                                      include 3 ssrc values in the
                                      ssrc-group? What is each one for?</p>
                                    <div>
                                      <div>
                                        <div class=3D"gmail_quote">On 21
                                          Oct 2014 19:31, &quot;Peter
                                          Thatcher&quot; &lt;<a href=3D"mai=
lto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt;
                                          wrote:<br type=3D"attribution">
                                          <blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                                            <div dir=3D"ltr">
                                              <div class=3D"gmail_default" =
style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-family=
:arial,sans-serif;font-size:12.7272720336914px">345259865
                                                  is &quot;real&quot;</span=
><br>
                                              </div>
                                              <div class=3D"gmail_default">=
<font face=3D"arial,
                                                  sans-serif"><a href=3D"te=
l:2693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>
                                                  is rtx</font><br>
                                              </div>
                                            </div>
                                            <div class=3D"gmail_extra"><br>
                                              <div class=3D"gmail_quote">On
                                                Tue, Oct 21, 2014 at
                                                9:25 AM, I=C3=B1aki Baz
                                                Castillo <span dir=3D"ltr">=
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;</span>
                                                wrote:<br>
                                                <blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">Hi,<br>
                                                  <br>
                                                  May I know which SSRC
                                                  (345259865 or <a href=3D"=
tel:2693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>)
                                                  will be used for the<br>
                                                  real media stream
                                                  (plus RED and FEC) and
                                                  which SSRC will be
                                                  used for<br>
                                                  RTX?<br>
                                                  <br>
                                                  <br>
                                                  <br>
--------------------------<br>
                                                  m=3Dvideo 62164
                                                  RTP/SAVPF 100 116 117
                                                  96<br>
                                                  a=3Dmid:video<br>
                                                  a=3Drtpmap:100 VP8/90000<=
br>
                                                  a=3Drtpmap:116 red/90000<=
br>
                                                  a=3Drtpmap:117
                                                  ulpfec/90000<br>
                                                  a=3Drtpmap:96 rtx/90000<b=
r>
                                                  a=3Dfmtp:96 apt=3D100<br>
                                                  a=3Dssrc-group:FID
                                                  345259865 <a href=3D"tel:=
2693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a><br>
                                                  a=3Dssrc:345259865
                                                  cname:erS7E/KHLYKTejNs<br=
>
                                                  a=3Dssrc:345259865
                                                  msid:DWpWct9bWKzTMNYZn5bK=
VgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                                  a=3Dssrc:345259865
                                                  mslabel:DWpWct9bWKzTMNYZn=
5bKVgwZ8Mfy2EtfqBY5<br>
                                                  a=3Dssrc:345259865
                                                  <a>label:c0134f05-e7c2-4a=
fd-a979-4e224de5eb91</a><br>
                                                  a=3Dssrc:<a href=3D"tel:2=
693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>
                                                  cname:erS7E/KHLYKTejNs<br=
>
                                                  a=3Dssrc:<a href=3D"tel:2=
693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>
msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                                  a=3Dssrc:<a href=3D"tel:2=
693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>
mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                                  a=3Dssrc:<a href=3D"tel:2=
693756249" value=3D"+12693756249" target=3D"_blank">2693756249</a>
<a>label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
-------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  RFC 5576 does not
                                                  clarify it at all:<br>
                                                  <br>
                                                  <a href=3D"http://tools.i=
etf.org/html/rfc5576#section-4.2" target=3D"_blank">http://tools.ietf.org/h=
tml/rfc5576#section-4.2</a><br>
                                                  <br>
--------------------------------------------------<br>
                                                  4.2.=C2=A0 The &quot;ssrc=
-group&quot;
                                                  Media Attribute<br>
                                                  <br>
                                                  =C2=A0
                                                  =C2=A0a=3Dssrc-group:&lt;=
semantics&gt;
                                                  &lt;ssrc-id&gt; ...<br>
                                                  <br>
                                                  =C2=A0 =C2=A0[..]<br>
                                                  <br>
                                                  =C2=A0 =C2=A0The
                                                  &lt;semantics&gt;
                                                  parameter is taken
                                                  from the specification
                                                  of the<br>
                                                  =C2=A0 =C2=A0&quot;group&=
quot; attribute
                                                  [RFC3388].=C2=A0 The
                                                  initial semantic
                                                  values defined for<br>
                                                  =C2=A0 =C2=A0the &quot;ss=
rc-group&quot;
                                                  attribute are FID
                                                  (Flow Identification)
                                                  [RFC3388]<br>
                                                  =C2=A0 =C2=A0and FEC (For=
ward
                                                  Error Correction)
                                                  [RFC4756].=C2=A0 In each
                                                  case, the<br>
                                                  =C2=A0 =C2=A0relationship=
 among
                                                  the grouped sources is
                                                  the same as the<br>
                                                  =C2=A0 =C2=A0relationship=
 among
                                                  corresponding sources
                                                  in media streams
                                                  grouped<br>
                                                  =C2=A0 =C2=A0using the SD=
P
                                                  &quot;group&quot; attribu=
te.<br>
--------------------------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  The referenced RFC
                                                  3388 neither clarifies
                                                  it:<br>
                                                  <br>
---------------------------------------------------<br>
                                                  7.4 FID Semantics<br>
                                                  <br>
                                                  =C2=A0 =C2=A0Several &quo=
t;m&quot; lines
                                                  grouped together using
                                                  FID semantics form a
                                                  media<br>
                                                  =C2=A0 =C2=A0flow.=C2=A0 =
A media
                                                  agent handling a media
                                                  flow that comprises
                                                  several &quot;m&quot;<br>
                                                  =C2=A0 =C2=A0lines MUST s=
end a
                                                  copy of the media to
                                                  every &quot;m&quot; line =
part of
                                                  the<br>
                                                  =C2=A0 =C2=A0flow as long=
 as the
                                                  codecs and the
                                                  direction attribute
                                                  present in a<br>
                                                  =C2=A0 =C2=A0particular &=
quot;m&quot; line
                                                  allow it.<br>
                                                  <br>
                                                  =C2=A0 =C2=A0It is assume=
d that
                                                  the application uses
                                                  only one codec at a
                                                  time to<br>
                                                  =C2=A0 =C2=A0encode the m=
edia
                                                  produced.=C2=A0 This code=
c
                                                  MAY change dynamically
                                                  during<br>
                                                  =C2=A0 =C2=A0the session,=
 but at
                                                  any particular moment
                                                  only one codec is in
                                                  use.<br>
                                                  <br>
                                                  =C2=A0 =C2=A0The applicat=
ion
                                                  encodes the media
                                                  using the current
                                                  codec and checks<br>
                                                  =C2=A0 =C2=A0one by one a=
ll the
                                                  &quot;m&quot; lines that =
are
                                                  part of the flow.=C2=A0 I=
f
                                                  a<br>
                                                  =C2=A0 =C2=A0particular &=
quot;m&quot; line
                                                  contains the codec
                                                  being used and the
                                                  direction<br>
                                                  =C2=A0 =C2=A0attribute is
                                                  &quot;sendonly&quot; or
                                                  &quot;sendrecv&quot;, a c=
opy of
                                                  the encoded media is<br>
                                                  =C2=A0 =C2=A0sent to the
                                                  address/port specified
                                                  in that particular
                                                  media stream.<br>
                                                  =C2=A0 =C2=A0If either th=
e &quot;m&quot;
                                                  line does not contain
                                                  the codec being used
                                                  or the<br>
                                                  =C2=A0 =C2=A0direction at=
tribute
                                                  is neither &quot;sendonly=
&quot;
                                                  nor &quot;sendrecv&quot;,
                                                  nothing is<br>
                                                  =C2=A0 =C2=A0sent over th=
is
                                                  media stream.<br>
----------------------------------------------------<br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                  So, how is the usage
                                                  of ssrc-group? Where
                                                  is it really defined?<br>
                                                  <br>
                                                  Can I put more than 2
                                                  ssrc together in the
                                                  same ssrc-group line?<br>
                                                  <br>
                                                  How should the
                                                  receiver interpret it?<br=
>
                                                  <br>
                                                  Does a ssrc-group
                                                  always mean RTX usage?
                                                  Where is that
                                                  specified in<br>
                                                  the above SDP?<br>
                                                  <br>
                                                  Does not the above SDP
                                                  look a complete
                                                  mixture of hacks and
                                                  workarounds?<br>
                                                  <span><font color=3D"#888=
888"><br>
                                                      <br>
                                                      <br>
                                                      <br>
                                                      --<br>
                                                      I=C3=B1aki Baz Castil=
lo<br>
                                                      &lt;<a href=3D"mailto=
:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;<br>
                                                      <br>
_______________________________________________<br>
                                                      rtcweb mailing
                                                      list<br>
                                                      <a href=3D"mailto:rtc=
web@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>
                                                      <a href=3D"https://ww=
w.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/=
mailman/listinfo/rtcweb</a><br>
                                                    </font></span></blockqu=
ote>
                                              </div>
                                              <br>
                                            </div>
                                          </blockquote>
                                        </div>
                                      </div>
                                    </div>
                                  </blockquote>
                                </div>
                                <br>
                              </div>
                            </blockquote>
                          </div>
                        </div>
                      </blockquote>
                      <blockquote type=3D"cite">
                        <div><span>________________________________________=
_______</span><br>
                          <span>rtcweb mailing list</span><br>
                          <span><a href=3D"mailto:rtcweb@ietf.org" target=
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tinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcwe=
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                        </div>
                      </blockquote>
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                  </div>
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                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb=
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rget=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
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              </blockquote>
            </div>
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      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
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</pre>
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    <br>
  </div></div></div>

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BTW, please, please, please, let's use the sssrc-group:FEC and send the 
FEC in its own ssrc stream, so we don't have to use RED anymore..

On 22/10/2014 21:01, Bernard Aboba wrote:
> SDP specifies payload types for RED/ULPFEC/RTX so as to allow 
> differentiation within an ssrc-group. So while an fmtp: attribute 
> could be used to denote a media stream on an a:ssrc line, it is not 
> necessary.
>
> RFC 5176 Example 3 gives an example with two cameras in which there 
> are two ssrc-group lines (one for each camera), with payload type used 
> to differentiate between H.264 and RTX in each ssrc-group:
>
>
>     m=video 49174 RTP/AVPF 96 98
>   a=rtpmap:96 H.264/90000
>   a=rtpmap:98 rtx/90000
>   a=fmtp:98 apt=96;rtx-time=3000
>   a=ssrc-group:FID 11111 22222
>   a=ssrc:11111 cname:user3@example.com  <http://example.com>  
>   a=ssrc:22222 cname:user3@example.com  <http://example.com>  
>   a=ssrc-group:FID 33333 44444
>   a=ssrc:33333 cname:user3@example.com  <http://example.com>  
>   a=ssrc:44444 cname:user3@example.com  <http://example.com>
>
> I believe that RFC 5176 includes the two groups to make clear that there are two sources so that the Answerer might choose to reject the m-line if it cannot handle that. From RFC 5176 Section 8:
>
> When used with the SDP Offer/Answer Model [RFC3264  <http://tools.ietf.org/html/rfc3264>], SDP source-specific attributes describe only the sources that each party is
>     willing to send (whether it is sending RTP data or RTCP report
>     blocks).  No mechanism is provided by which an answer can accept or
>     reject individual sources within a media stream; if the set of
>     sources in a media stream is unacceptable, the answerer's only option
>     is to reject the media stream or the entire multimedia session.
>
> Note that in the example in RFC 4588 Section 8.8 where there is only one source, there are no ssrc-group lines at all:
>
>   v=0
>   o=mascha 2980675221 2980675778 IN IP4host.example.net  <http://host.example.net>  
>   c=IN IP4 192.0.2.0
>   m=video 49170 RTP/AVPF 96 97
>   a=rtpmap:96 MP4V-ES/90000
>   a=rtcp-fb:96 nack
>   a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F
>   a=rtpmap:97 rtx/90000
>   a=fmtp:97 apt=96;rtx-time=3000
>
> The situation would be the same for FEC groups created via 
> ssrc-group:FEC. From
> https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
>
>        The FEC and the payload MAY also be multiplexed by SSRC into one
>        single RTP session, with separate SSRC values, if the association
>        between FEC and payload streams are communicated to all members of
>        the session.  If SDP is used, this association MAY be communicated
>        through the FEC ssrc-group semantic [RFC5576  <https://tools.ietf.org/html/rfc5576>]; other mechanisms
>        are also possible.  Receivers MUST NOT attempt to interpret FEC
>        streams for which they do not have information to associate them
>        with the corresponding payload streams.
>
>
>
> On Oct 21, 2014, at 11:36 AM, Iñaki Baz Castillo <ibc@aliax.net 
> <mailto:ibc@aliax.net>> wrote:
>
>> I hope I can rule my SDP logic based on standards rather than on how 
>> Chrome implements some features.
>>
>> My question was generic: if I receive the above SDP, how do I know 
>> which payloads each ssrc is supposed to transport?
>>
>> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com 
>> <mailto:pthatcher@google.com>> wrote:
>>
>>     I'm not sure where it's specified, but here's where it's implemented:
>>
>>     https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108
>>
>>     It only supports 2 SSRCs for a FID group.  It would ignore any
>>     more than that.
>>
>>     On Tue, Oct 21, 2014 at 10:40 AM, Iñaki Baz Castillo
>>     <ibc@aliax.net <mailto:ibc@aliax.net>> wrote:
>>
>>         How is that? Where is that specified? What about if I include
>>         3 ssrc values in the ssrc-group? What is each one for?
>>
>>         On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com
>>         <mailto:pthatcher@google.com>> wrote:
>>
>>             345259865 is "real"
>>             2693756249 <tel:2693756249> is rtx
>>
>>             On Tue, Oct 21, 2014 at 9:25 AM, Iñaki Baz Castillo
>>             <ibc@aliax.net <mailto:ibc@aliax.net>> wrote:
>>
>>                 Hi,
>>
>>                 May I know which SSRC (345259865 or 2693756249
>>                 <tel:2693756249>) will be used for the
>>                 real media stream (plus RED and FEC) and which SSRC
>>                 will be used for
>>                 RTX?
>>
>>
>>
>>                 --------------------------
>>                 m=video 62164 RTP/SAVPF 100 116 117 96
>>                 a=mid:video
>>                 a=rtpmap:100 VP8/90000
>>                 a=rtpmap:116 red/90000
>>                 a=rtpmap:117 ulpfec/90000
>>                 a=rtpmap:96 rtx/90000
>>                 a=fmtp:96 apt=100
>>                 a=ssrc-group:FID 345259865 2693756249 <tel:2693756249>
>>                 a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>>                 a=ssrc:345259865
>>                 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>                 c0134f05-e7c2-4afd-a979-4e224de5eb91
>>                 a=ssrc:345259865
>>                 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>                 a=ssrc:345259865
>>                 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>                 a=ssrc:2693756249 <tel:2693756249> cname:erS7E/KHLYKTejNs
>>                 a=ssrc:2693756249 <tel:2693756249>
>>                 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>                 c0134f05-e7c2-4afd-a979-4e224de5eb91
>>                 a=ssrc:2693756249 <tel:2693756249>
>>                 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>                 a=ssrc:2693756249 <tel:2693756249>
>>                 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>                 -------------------------------
>>
>>
>>
>>
>>                 RFC 5576 does not clarify it at all:
>>
>>                 http://tools.ietf.org/html/rfc5576#section-4.2
>>
>>                 --------------------------------------------------
>>                 4.2.  The "ssrc-group" Media Attribute
>>
>>                    a=ssrc-group:<semantics> <ssrc-id> ...
>>
>>                    [..]
>>
>>                    The <semantics> parameter is taken from the
>>                 specification of the
>>                    "group" attribute [RFC3388].  The initial semantic
>>                 values defined for
>>                    the "ssrc-group" attribute are FID (Flow
>>                 Identification) [RFC3388]
>>                    and FEC (Forward Error Correction) [RFC4756].  In
>>                 each case, the
>>                    relationship among the grouped sources is the same
>>                 as the
>>                    relationship among corresponding sources in media
>>                 streams grouped
>>                    using the SDP "group" attribute.
>>                 --------------------------------------------------
>>
>>
>>
>>                 The referenced RFC 3388 neither clarifies it:
>>
>>                 ---------------------------------------------------
>>                 7.4 FID Semantics
>>
>>                    Several "m" lines grouped together using FID
>>                 semantics form a media
>>                    flow.  A media agent handling a media flow that
>>                 comprises several "m"
>>                    lines MUST send a copy of the media to every "m"
>>                 line part of the
>>                    flow as long as the codecs and the direction
>>                 attribute present in a
>>                    particular "m" line allow it.
>>
>>                    It is assumed that the application uses only one
>>                 codec at a time to
>>                    encode the media produced.  This codec MAY change
>>                 dynamically during
>>                    the session, but at any particular moment only one
>>                 codec is in use.
>>
>>                    The application encodes the media using the
>>                 current codec and checks
>>                    one by one all the "m" lines that are part of the
>>                 flow.  If a
>>                    particular "m" line contains the codec being used
>>                 and the direction
>>                    attribute is "sendonly" or "sendrecv", a copy of
>>                 the encoded media is
>>                    sent to the address/port specified in that
>>                 particular media stream.
>>                    If either the "m" line does not contain the codec
>>                 being used or the
>>                    direction attribute is neither "sendonly" nor
>>                 "sendrecv", nothing is
>>                    sent over this media stream.
>>                 ----------------------------------------------------
>>
>>
>>
>>
>>                 So, how is the usage of ssrc-group? Where is it
>>                 really defined?
>>
>>                 Can I put more than 2 ssrc together in the same
>>                 ssrc-group line?
>>
>>                 How should the receiver interpret it?
>>
>>                 Does a ssrc-group always mean RTX usage? Where is
>>                 that specified in
>>                 the above SDP?
>>
>>                 Does not the above SDP look a complete mixture of
>>                 hacks and workarounds?
>>
>>
>>
>>
>>                 --
>>                 Iñaki Baz Castillo
>>                 <ibc@aliax.net <mailto:ibc@aliax.net>>
>>


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    <div class="moz-cite-prefix">BTW, please, please, please, let's use&nbsp;
      the sssrc-group:FEC and send the FEC in its own ssrc stream, so we
      don't have to use RED anymore.. <br>
      <br>
      On 22/10/2014 21:01, Bernard Aboba wrote:<br>
    </div>
    <blockquote
      cite="mid:5F93BF20-03B2-4D2D-BEFC-ED91250993BD@gmail.com"
      type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <div>SDP specifies payload types for RED/ULPFEC/RTX so as to allow
        differentiation within an ssrc-group. So while an fmtp:
        attribute could be used to denote a media stream on an a:ssrc
        line, it is not necessary.</div>
      <div><br>
      </div>
      <div>RFC 5176 Example 3 gives an example with two cameras in which
        there are two ssrc-group lines (one for each camera), with
        payload type used to differentiate between H.264 and RTX in each
        ssrc-group:</div>
      <div><br>
      </div>
      <div>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">
   m=video 49174 RTP/AVPF 96 98&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=rtpmap:96 H.264/90000&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=rtpmap:98 rtx/90000&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=fmtp:98 apt=96;rtx-time=3000&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc-group:FID 11111 22222&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc:11111 cname:user3@<a moz-do-not-send="true" href="http://example.com">example.com</a>&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc:22222 cname:user3@<a moz-do-not-send="true" href="http://example.com">example.com</a>&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc-group:FID 33333 44444&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc:33333 cname:user3@<a moz-do-not-send="true" href="http://example.com">example.com</a>&nbsp;</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">&nbsp;a=ssrc:44444 cname:user3@<a moz-do-not-send="true" href="http://example.com">example.com</a></span></font><span style="font-size: 1em; -webkit-text-size-adjust: auto;">
</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">
</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">I believe that RFC 5176 includes the two groups to make clear that there are two sources so that the Answerer might choose to reject the m-line if it cannot handle that.&nbsp;</span></font><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">From RFC 5176 Section 8:</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">
</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">When used with the SDP Offer/Answer Model [</span><a moz-do-not-send="true" href="http://tools.ietf.org/html/rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;" style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">RFC3264</a><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">], SDP source-specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer's only option
   is to reject the media stream or the entire multimedia session.</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">
</span></font></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">Note that in the example in RFC 4588 Section 8.8 where there is only one source, there are no ssrc-group lines at all:</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">
</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;v=0&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;o=mascha 2980675221 2980675778 IN IP4 <a moz-do-not-send="true" href="http://host.example.net">host.example.net</a>&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;c=IN IP4 192.0.2.0&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;m=video 49170 RTP/AVPF 96 97&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=rtpmap:96 MP4V-ES/90000&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=rtcp-fb:96 nack&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=rtpmap:97 rtx/90000&nbsp;</span></pre>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0); font-family: UICTFontTextStyleBody;">&nbsp;a=fmtp:97 apt=96;rtx-time=3000</span></pre>
      </div>
      <div><br>
      </div>
      <div>The situation would be the same for FEC groups created via
        ssrc-group:FEC. From&nbsp;</div>
      <div><a moz-do-not-send="true"
          href="https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux">https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux</a>
        &nbsp;:</div>
      <div><br>
      </div>
      <div>
        <pre class="newpage" style="margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font face="UICTFontTextStyleBody"><span style="white-space: normal; background-color: rgba(255, 255, 255, 0);">      The FEC and the payload MAY also be multiplexed by SSRC into one
      single RTP session, with separate SSRC values, if the association
      between FEC and payload streams are communicated to all members of
      the session.  If SDP is used, this association MAY be communicated
      through the FEC ssrc-group semantic [<a moz-do-not-send="true" href="https://tools.ietf.org/html/rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;">RFC5576</a>]; other mechanisms
      are also possible.  Receivers MUST NOT attempt to interpret FEC
      streams for which they do not have information to associate them
      with the corresponding payload streams.</span></font><span style="font-size: 1em; -webkit-text-size-adjust: auto;">
</span></pre>
      </div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
        On Oct 21, 2014, at 11:36 AM, I&ntilde;aki Baz Castillo &lt;<a
          moz-do-not-send="true" href="mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;
        wrote:<br>
        <br>
      </div>
      <blockquote type="cite">
        <div>
          <p dir="ltr">I hope I can rule my SDP logic based on standards
            rather than on how Chrome implements some features.</p>
          <p dir="ltr">My question was generic: if I receive the above
            SDP, how do I know which payloads each ssrc is supposed to
            transport?</p>
          <div class="gmail_quote">On 21 Oct 2014 19:57, "Peter
            Thatcher" &lt;<a moz-do-not-send="true"
              href="mailto:pthatcher@google.com">pthatcher@google.com</a>&gt;
            wrote:<br type="attribution">
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div dir="ltr">
                <div class="gmail_default"
                  style="font-family:arial,helvetica,sans-serif">I'm not
                  sure where it's specified, but here's where it's
                  implemented:</div>
                <div class="gmail_default"
                  style="font-family:arial,helvetica,sans-serif"><br>
                </div>
                <div class="gmail_default"><font face="arial, helvetica,
                    sans-serif"><a moz-do-not-send="true"
href="https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108"
                      target="_blank">https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoengine.cc#4108</a></font><br>
                </div>
                <div class="gmail_default"><font face="arial, helvetica,
                    sans-serif"><br>
                  </font></div>
                <div class="gmail_default"><font face="arial, helvetica,
                    sans-serif">It only supports 2 SSRCs for a FID
                    group.&nbsp; It would ignore any more than that.</font></div>
              </div>
              <div class="gmail_extra"><br>
                <div class="gmail_quote">On Tue, Oct 21, 2014 at 10:40
                  AM, I&ntilde;aki Baz Castillo <span dir="ltr">&lt;<a
                      moz-do-not-send="true" href="mailto:ibc@aliax.net"
                      target="_blank">ibc@aliax.net</a>&gt;</span>
                  wrote:<br>
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    <p dir="ltr">How is that? Where is that specified?
                      What about if I include 3 ssrc values in the
                      ssrc-group? What is each one for?</p>
                    <div>
                      <div>
                        <div class="gmail_quote">On 21 Oct 2014 19:31,
                          "Peter Thatcher" &lt;<a moz-do-not-send="true"
                            href="mailto:pthatcher@google.com"
                            target="_blank">pthatcher@google.com</a>&gt;
                          wrote:<br type="attribution">
                          <blockquote class="gmail_quote"
                            style="margin:0 0 0 .8ex;border-left:1px
                            #ccc solid;padding-left:1ex">
                            <div dir="ltr">
                              <div class="gmail_default"
                                style="font-family:arial,helvetica,sans-serif"><span
style="font-family:arial,sans-serif;font-size:12.7272720336914px">345259865
                                  is "real"</span><br>
                              </div>
                              <div class="gmail_default"><font
                                  face="arial, sans-serif"><a
                                    moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>
                                  is rtx</font><br>
                              </div>
                            </div>
                            <div class="gmail_extra"><br>
                              <div class="gmail_quote">On Tue, Oct 21,
                                2014 at 9:25 AM, I&ntilde;aki Baz Castillo <span
                                  dir="ltr">&lt;<a
                                    moz-do-not-send="true"
                                    href="mailto:ibc@aliax.net"
                                    target="_blank">ibc@aliax.net</a>&gt;</span>
                                wrote:<br>
                                <blockquote class="gmail_quote"
                                  style="margin:0 0 0
                                  .8ex;border-left:1px #ccc
                                  solid;padding-left:1ex">Hi,<br>
                                  <br>
                                  May I know which SSRC (345259865 or <a
                                    moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>)
                                  will be used for the<br>
                                  real media stream (plus RED and FEC)
                                  and which SSRC will be used for<br>
                                  RTX?<br>
                                  <br>
                                  <br>
                                  <br>
                                  --------------------------<br>
                                  m=video 62164 RTP/SAVPF 100 116 117 96<br>
                                  a=mid:video<br>
                                  a=rtpmap:100 VP8/90000<br>
                                  a=rtpmap:116 red/90000<br>
                                  a=rtpmap:117 ulpfec/90000<br>
                                  a=rtpmap:96 rtx/90000<br>
                                  a=fmtp:96 apt=100<br>
                                  a=ssrc-group:FID 345259865 <a
                                    moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a><br>
                                  a=ssrc:345259865
                                  cname:erS7E/KHLYKTejNs<br>
                                  a=ssrc:345259865
                                  msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                  c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                  a=ssrc:345259865
                                  mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                  a=ssrc:345259865
                                  <a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
                                  a=ssrc:<a moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>
                                  cname:erS7E/KHLYKTejNs<br>
                                  a=ssrc:<a moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>
msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                  c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
                                  a=ssrc:<a moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>
mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
                                  a=ssrc:<a moz-do-not-send="true"
                                    href="tel:2693756249"
                                    value="+12693756249" target="_blank">2693756249</a>
<a class="moz-txt-link-freetext" href="label:c0134f05-e7c2-4afd-a979-4e224de5eb91">label:c0134f05-e7c2-4afd-a979-4e224de5eb91</a><br>
                                  -------------------------------<br>
                                  <br>
                                  <br>
                                  <br>
                                  <br>
                                  RFC 5576 does not clarify it at all:<br>
                                  <br>
                                  <a moz-do-not-send="true"
                                    href="http://tools.ietf.org/html/rfc5576#section-4.2"
                                    target="_blank">http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
                                  <br>
--------------------------------------------------<br>
                                  4.2.&nbsp; The "ssrc-group" Media Attribute<br>
                                  <br>
                                  &nbsp; &nbsp;a=ssrc-group:&lt;semantics&gt;
                                  &lt;ssrc-id&gt; ...<br>
                                  <br>
                                  &nbsp; &nbsp;[..]<br>
                                  <br>
                                  &nbsp; &nbsp;The &lt;semantics&gt; parameter is
                                  taken from the specification of the<br>
                                  &nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The
                                  initial semantic values defined for<br>
                                  &nbsp; &nbsp;the "ssrc-group" attribute are FID
                                  (Flow Identification) [RFC3388]<br>
                                  &nbsp; &nbsp;and FEC (Forward Error Correction)
                                  [RFC4756].&nbsp; In each case, the<br>
                                  &nbsp; &nbsp;relationship among the grouped
                                  sources is the same as the<br>
                                  &nbsp; &nbsp;relationship among corresponding
                                  sources in media streams grouped<br>
                                  &nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
                                  <br>
                                  <br>
                                  <br>
                                  The referenced RFC 3388 neither
                                  clarifies it:<br>
                                  <br>
---------------------------------------------------<br>
                                  7.4 FID Semantics<br>
                                  <br>
                                  &nbsp; &nbsp;Several "m" lines grouped together
                                  using FID semantics form a media<br>
                                  &nbsp; &nbsp;flow.&nbsp; A media agent handling a
                                  media flow that comprises several "m"<br>
                                  &nbsp; &nbsp;lines MUST send a copy of the media
                                  to every "m" line part of the<br>
                                  &nbsp; &nbsp;flow as long as the codecs and the
                                  direction attribute present in a<br>
                                  &nbsp; &nbsp;particular "m" line allow it.<br>
                                  <br>
                                  &nbsp; &nbsp;It is assumed that the application
                                  uses only one codec at a time to<br>
                                  &nbsp; &nbsp;encode the media produced.&nbsp; This
                                  codec MAY change dynamically during<br>
                                  &nbsp; &nbsp;the session, but at any particular
                                  moment only one codec is in use.<br>
                                  <br>
                                  &nbsp; &nbsp;The application encodes the media
                                  using the current codec and checks<br>
                                  &nbsp; &nbsp;one by one all the "m" lines that
                                  are part of the flow.&nbsp; If a<br>
                                  &nbsp; &nbsp;particular "m" line contains the
                                  codec being used and the direction<br>
                                  &nbsp; &nbsp;attribute is "sendonly" or
                                  "sendrecv", a copy of the encoded
                                  media is<br>
                                  &nbsp; &nbsp;sent to the address/port specified
                                  in that particular media stream.<br>
                                  &nbsp; &nbsp;If either the "m" line does not
                                  contain the codec being used or the<br>
                                  &nbsp; &nbsp;direction attribute is neither
                                  "sendonly" nor "sendrecv", nothing is<br>
                                  &nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
                                  <br>
                                  <br>
                                  <br>
                                  <br>
                                  So, how is the usage of ssrc-group?
                                  Where is it really defined?<br>
                                  <br>
                                  Can I put more than 2 ssrc together in
                                  the same ssrc-group line?<br>
                                  <br>
                                  How should the receiver interpret it?<br>
                                  <br>
                                  Does a ssrc-group always mean RTX
                                  usage? Where is that specified in<br>
                                  the above SDP?<br>
                                  <br>
                                  Does not the above SDP look a complete
                                  mixture of hacks and workarounds?<br>
                                  <span><font color="#888888"><br>
                                      <br>
                                      <br>
                                      <br>
                                      --<br>
                                      I&ntilde;aki Baz Castillo<br>
                                      &lt;<a moz-do-not-send="true"
                                        href="mailto:ibc@aliax.net"
                                        target="_blank">ibc@aliax.net</a>&gt;<br>
                                    </font></span><br>
                                </blockquote>
                              </div>
                            </div>
                          </blockquote>
                        </div>
                      </div>
                    </div>
                  </blockquote>
                </div>
              </div>
            </blockquote>
          </div>
        </div>
      </blockquote>
    </blockquote>
    <br>
  </body>
</html>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Suhas Nandakumar <suhasietf@gmail.com>, Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] SDP and ssrc-group,
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Date: Wed, 22 Oct 2014 20:05:33 +0000
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] SDP and ssrc-group,
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Date: Wed, 22 Oct 2014 20:14:31 +0000
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From: Justin Uberti <juberti@google.com>
Date: Wed, 22 Oct 2014 13:36:05 -0700
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--001a1140a9beceb237050608e60e
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yes, they need to happen. Framing is 4571. See
http://tools.ietf.org/html/rfc6544#section-7.1.

On Wed, Oct 22, 2014 at 7:52 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>  On 08/25/2014 07:36 AM, Justin Uberti wrote:
>
> Yes, 4571 must be used for ICE connectivity checks, when using ICE-TCP.
>
>  However, TURN/TCP has its own framing for STUN messages.
>
>  I would simplify your sentence to:
>
>     If ICE-TCP connections are used, framing according to [RFC4571] MUST
>    be used for all packets, unless the content has its own framing
> mechanism.
>
>
> Checking the list for long-delayed updates....
>
> Checking back: The change from TCP to ICE-TCP is good (this excludes
> TURN-TCP from this paragraph), but the exchange leaves me uncertain about
> the status of STUN packets on ICE-TCP connections.
>
> Do they happen at all? If they happen, should they be RFC 4571
> encapsulated, or not?
>
>
>
>
> On Fri, Aug 22, 2014 at 8:38 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>
>> Hi,
>>
>> draft-ietf-rtcweb-transports-06 says:
>>
>>    If TCP connections are used, RTP framing according to [RFC4571] MUST
>>    be used, both for the RTP packets and for the DTLS packets used to
>>    carry data channels.
>>
>>
>> Not just RTP and DTLS, but also for STUN packets must be framed with
>> RFC4571. And I do not understand why "DTLS packets used to carry data
>> channels". In fact all the DTLS packets (also the ones for the
>> handshake) must be framed as per RFC 4571.
>>
>>
>> So I suggest to improve the above text as follows:
>>
>>    If TCP connections are used, RTP framing according to [RFC4571] MUST
>>    be used for the STUN packets, SRTP packets, SRTCP packets and DTLS
>>    packets.
>>
>>
>>
>>
>>
>>
>> --
>> I=C3=B1aki Baz Castillo
>> <ibc@aliax.net>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/r=
tcweb
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1140a9beceb237050608e60e
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<div dir=3D"ltr">yes, they need to happen. Framing is 4571. See=C2=A0<a hre=
f=3D"http://tools.ietf.org/html/rfc6544#section-7.1">http://tools.ietf.org/=
html/rfc6544#section-7.1</a>.</div><div class=3D"gmail_extra"><br><div clas=
s=3D"gmail_quote">On Wed, Oct 22, 2014 at 7:52 AM, Harald Alvestrand <span =
dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">h=
arald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
    <div>On 08/25/2014 07:36 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">Yes, 4571 must be used for ICE connectivity checks,
        when using ICE-TCP.
        <div><br>
        </div>
        <div>However, TURN/TCP has its own framing for STUN messages.</div>
        <div><br>
        </div>
        <div>I would simplify your sentence to:</div>
        <div><br>
        </div>
        <div><span style=3D"font-family:arial,sans-serif;font-size:13px">=
=C2=A0
            =C2=A0If ICE-TCP connections are used, framing according to
            [RFC4571] MUST</span><br style=3D"font-family:arial,sans-serif;=
font-size:13px">
          <span style=3D"font-family:arial,sans-serif;font-size:13px">=C2=
=A0
            =C2=A0be used for all packets, unless the content has its own
            framing mechanism.</span><br>
        </div>
      </div>
    </blockquote>
    <br></span>
    Checking the list for long-delayed updates....<br>
    <br>
    Checking back: The change from TCP to ICE-TCP is good (this excludes
    TURN-TCP from this paragraph), but the exchange leaves me uncertain
    about the status of STUN packets on ICE-TCP connections.<br>
    <br>
    Do they happen at all? If they happen, should they be RFC 4571
    encapsulated, or not?<div><div class=3D"h5"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
        </div>
      </div>
      <div class=3D"gmail_extra"><br>
        <br>
        <div class=3D"gmail_quote">On Fri, Aug 22, 2014 at 8:38 AM, I=C3=B1=
aki
          Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.ne=
t" target=3D"_blank">ibc@aliax.net</a>&gt;</span>
          wrote:<br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">Hi,<br>
            <br>
            draft-ietf-rtcweb-transports-06 says:<br>
            <br>
            =C2=A0 =C2=A0If TCP connections are used, RTP framing according=
 to
            [RFC4571] MUST<br>
            =C2=A0 =C2=A0be used, both for the RTP packets and for the DTLS
            packets used to<br>
            =C2=A0 =C2=A0carry data channels.<br>
            <br>
            <br>
            Not just RTP and DTLS, but also for STUN packets must be
            framed with<br>
            RFC4571. And I do not understand why &quot;DTLS packets used to
            carry data<br>
            channels&quot;. In fact all the DTLS packets (also the ones for
            the<br>
            handshake) must be framed as per RFC 4571.<br>
            <br>
            <br>
            So I suggest to improve the above text as follows:<br>
            <br>
            =C2=A0 =C2=A0If TCP connections are used, RTP framing according=
 to
            [RFC4571] MUST<br>
            =C2=A0 =C2=A0be used for the STUN packets, SRTP packets, SRTCP =
packets
            and DTLS<br>
            =C2=A0 =C2=A0packets.<br>
            <span><font color=3D"#888888"><br>
                <br>
                <br>
                <br>
                <br>
                <br>
                --<br>
                I=C3=B1aki Baz Castillo<br>
                &lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@=
aliax.net</a>&gt;<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" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </font></span></blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    </div></div><span class=3D"HOEnZb"><font color=3D"#888888"><pre cols=3D=
"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
  </font></span></div>

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

--001a1140a9beceb237050608e60e--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 22 Oct 2014 14:28:37 -0700
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RFC 5576 makes it clear that partial rejection is not supported.=20



> On Oct 22, 2014, at 1:14 PM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@a=
lcatel-lucent.com> wrote:
>=20
> SDP specifies payload types for RED/ULPFEC/RTX so as to allow differentiat=
ion within an ssrc-group. So while an fmtp: attribute could be used to denot=
e a media stream on an a:ssrc line, it is not necessary.
> <Raju>
> What if the answerer wants to reject a=3Dssrc-group:FID (and a=3Dfmtp:98 a=
pt=3D96;rtx-time=3D3000) then which of the two ssrcs it can expect to receiv=
e for the original transmission?
> </Raju>
> =20
> BR
> Raju
> =20
> RFC 5576 Example 3 gives an example with two cameras in which there are tw=
o ssrc-group lines (one for each camera), with payload type used to differen=
tiate between H.264 and RTX in each ssrc-group:
> =20
> m=3Dvideo 49174 RTP/AVPF 96 98=20
>  a=3Drtpmap:96 H.264/90000=20
>  a=3Drtpmap:98 rtx/90000=20
>  a=3Dfmtp:98 apt=3D96;rtx-time=3D3000=20
>  a=3Dssrc-group:FID 11111 22222=20
>  a=3Dssrc:11111 cname:user3@example.com=20
>  a=3Dssrc:22222 cname:user3@example.com=20
>  a=3Dssrc-group:FID 33333 44444=20
>  a=3Dssrc:33333 cname:user3@example.com=20
>  a=3Dssrc:44444 cname:user3@example.com
>=20
>  I believe that RFC 5577 includes the two groups to make clear that there a=
re two sources so that the Answerer might choose to reject the m-line if it c=
annot handle that. =46rom RFC 5576 Section 8:
>=20
>  When used with the SDP Offer/Answer Model [RFC3264], SDP source-specific a=
ttributes describe only the sources that each party is willing to send (whet=
her it is sending RTP data or RTCP report blocks). No mechanism is provided b=
y which an answer can accept or reject individual sources within a media str=
eam; if the set of sources in a media stream is unacceptable, the answerer's=
 only option is to reject the media stream or the entire multimedia session.=

>=20
>  Note that in the example in RFC 4588 Section 8.8 where there is only one s=
ource, there are no ssrc-group lines at all:
>=20
>   v=3D0=20
>  o=3Dmascha 2980675221 2980675778 IN IP4 host.example.net=20
>  c=3DIN IP4 192.0.2.0=20
>  m=3Dvideo 49170 RTP/AVPF 96 97=20
>  a=3Drtpmap:96 MP4V-ES/90000=20
>  a=3Drtcp-fb:96 nack=20
>  a=3Dfmtp:96 profile-level-id=3D8;config=3D01010000012000884006682C2090A21=
F=20
>  a=3Drtpmap:97 rtx/90000=20
>  a=3Dfmtp:97 apt=3D96;rtx-time=3D3000
> =20
> The situation would be the same for FEC groups created via ssrc-group:FEC.=
 =46rom=20
> https://tools.ietf.org/html/draft-lennox-payload-ulp-ssrc-mux  :
> =20
> The FEC and the payload MAY also be multiplexed by SSRC into one single RT=
P session, with separate SSRC values, if the association between FEC and pay=
load streams are communicated to all members of the session. If SDP is used,=
 this association MAY be communicated through the FEC ssrc-group semantic [R=
FC5576]; other mechanisms are also possible. Receivers MUST NOT attempt to i=
nterpret FEC streams for which they do not have information to associate the=
m with the corresponding payload streams.
> =20
> =20
>=20
> On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrot=
e:
>=20
> I hope I can rule my SDP logic based on standards rather than on how Chrom=
e implements some features.
>=20
> My question was generic: if I receive the above SDP, how do I know which p=
ayloads each ssrc is supposed to transport?
>=20
> On 21 Oct 2014 19:57, "Peter Thatcher" <pthatcher@google.com> wrote:
> I'm not sure where it's specified, but here's where it's implemented:
> =20
> https://code.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/web=
rtcvideoengine.cc#4108
> =20
> It only supports 2 SSRCs for a FID group.  It would ignore any more than t=
hat.
> =20
> On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
> How is that? Where is that specified? What about if I include 3 ssrc value=
s in the ssrc-group? What is each one for?
>=20
> On 21 Oct 2014 19:31, "Peter Thatcher" <pthatcher@google.com> wrote:
> 345259865 is "real"
> 2693756249 is rtx
> =20
> On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
> Hi,
>=20
> May I know which SSRC (345259865 or 2693756249) will be used for the
> real media stream (plus RED and FEC) and which SSRC will be used for
> RTX?
>=20
>=20
>=20
> --------------------------
> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
> a=3Dmid:video
> a=3Drtpmap:100 VP8/90000
> a=3Drtpmap:116 red/90000
> a=3Drtpmap:117 ulpfec/90000
> a=3Drtpmap:96 rtx/90000
> a=3Dfmtp:96 apt=3D100
> a=3Dssrc-group:FID 345259865 2693756249
> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> -------------------------------
>=20
>=20
>=20
>=20
> RFC 5576 does not clarify it at all:
>=20
> http://tools.ietf.org/html/rfc5576#section-4.2
>=20
> --------------------------------------------------
> 4.2.  The "ssrc-group" Media Attribute
>=20
>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>=20
>    [..]
>=20
>    The <semantics> parameter is taken from the specification of the
>    "group" attribute [RFC3388].  The initial semantic values defined for
>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>    relationship among the grouped sources is the same as the
>    relationship among corresponding sources in media streams grouped
>    using the SDP "group" attribute.
> --------------------------------------------------
>=20
>=20
>=20
> The referenced RFC 3388 neither clarifies it:
>=20
> ---------------------------------------------------
> 7.4 FID Semantics
>=20
>    Several "m" lines grouped together using FID semantics form a media
>    flow.  A media agent handling a media flow that comprises several "m"
>    lines MUST send a copy of the media to every "m" line part of the
>    flow as long as the codecs and the direction attribute present in a
>    particular "m" line allow it.
>=20
>    It is assumed that the application uses only one codec at a time to
>    encode the media produced.  This codec MAY change dynamically during
>    the session, but at any particular moment only one codec is in use.
>=20
>    The application encodes the media using the current codec and checks
>    one by one all the "m" lines that are part of the flow.  If a
>    particular "m" line contains the codec being used and the direction
>    attribute is "sendonly" or "sendrecv", a copy of the encoded media is
>    sent to the address/port specified in that particular media stream.
>    If either the "m" line does not contain the codec being used or the
>    direction attribute is neither "sendonly" nor "sendrecv", nothing is
>    sent over this media stream.
> ----------------------------------------------------
>=20
>=20
>=20
>=20
> So, how is the usage of ssrc-group? Where is it really defined?
>=20
> Can I put more than 2 ssrc together in the same ssrc-group line?
>=20
> How should the receiver interpret it?
>=20
> Does a ssrc-group always mean RTX usage? Where is that specified in
> the above SDP?
>=20
> Does not the above SDP look a complete mixture of hacks and workarounds?
>=20
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =20
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>RFC 5576 makes it clear that partial r=
ejection is not supported.&nbsp;<br><br><br></div><div><br>On Oct 22, 2014, a=
t 1:14 PM, Makaraju, Maridi Raju (Raju) &lt;<a href=3D"mailto:Raju.Makaraju@=
alcatel-lucent.com">Raju.Makaraju@alcatel-lucent.com</a>&gt; wrote:<br><br><=
/div><blockquote type=3D"cite"><div>

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<div class=3D"WordSection1">
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4=
.0pt">
<div>
<p class=3D"MsoNormal">SDP specifies payload types for RED/ULPFEC/RTX so as t=
o allow differentiation within an ssrc-group. So while an fmtp: attribute co=
uld be used to denote a media stream on an a:ssrc line, it is not necessary.=
<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">What if the answerer w=
ants to reject a=3Dssrc-group:FID (and a=3Dfmtp:98 apt=3D96;rtx-time=3D3000)=
 then which of the two ssrcs it can expect to receive for the
 original transmission? <o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<o:p></o=
:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></sp=
an></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span><=
/i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju</span></i></b><s=
pan style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-ser=
if&quot;;color:#1F497D"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">RFC 5576 Example 3 gives an example with two cameras i=
n which there are two ssrc-group lines (one for each camera), with payload t=
ype used to differentiate between H.264 and RTX in each ssrc-group:<o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">m=3Dvideo 49174 RTP/AVPF 96 98&n=
bsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Drtpmap:96 H.264/90000&=
nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Drtpmap:98 rtx/90000&nb=
sp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dfmtp:98 apt=3D96;rtx-t=
ime=3D3000&nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc-group:FID 11111 2=
2222&nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc:11111 cname:user3=
@<a href=3D"http://example.com">example.com</a>&nbsp;</span><o:p></o:p></pre=
>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc:22222 cname:user3=
@<a href=3D"http://example.com">example.com</a>&nbsp;</span><o:p></o:p></pre=
>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc-group:FID 33333 4=
4444&nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc:33333 cname:user3=
@<a href=3D"http://example.com">example.com</a>&nbsp;</span><o:p></o:p></pre=
>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dssrc:44444 cname:user3=
@<a href=3D"http://example.com">example.com</a></span><o:p></o:p></pre>
<span style=3D"font-size:10.0pt;font-family:&quot;UICTFontTextStyleBody&quot=
;,&quot;serif&quot;"><br clear=3D"all" style=3D"page-break-before:always">
</span>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">I believe that RFC 5577 includes=
 the two groups to make clear that there are two sources so that the Answere=
r might choose to reject the m-line if it cannot handle that.&nbsp;=46rom RFC=
 5576 Section 8:</span><o:p></o:p></pre>
<span style=3D"font-size:10.0pt;font-family:&quot;UICTFontTextStyleBody&quot=
;,&quot;serif&quot;"><br clear=3D"all" style=3D"page-break-before:always">
</span>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">When used with the SDP Offer/Ans=
wer Model [</span><a href=3D"http://tools.ietf.org/html/rfc3264" title=3D"&q=
uot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;"><sp=
an style=3D"font-family:&quot;UICTFontTextStyleBody&quot;,&quot;serif&quot;"=
>RFC3264</span></a><span style=3D"font-family:&quot;UICTFontTextStyleBody&qu=
ot;,&quot;serif&quot;">], SDP source-specific attributes describe only the s=
ources that each party is willing to send (whether it is sending RTP data or=
 RTCP report blocks). No mechanism is provided by which an answer can accept=
 or reject individual sources within a media stream; if the set of sources i=
n a media stream is unacceptable, the answerer's only option is to reject th=
e media stream or the entire multimedia session.</span><o:p></o:p></pre>
<span style=3D"font-size:10.0pt;font-family:&quot;UICTFontTextStyleBody&quot=
;,&quot;serif&quot;"><br clear=3D"all" style=3D"page-break-before:always">
</span>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">Note that in the example in RFC 4=
588 Section 8.8 where there is only one source, there are no ssrc-group line=
s at all:</span><o:p></o:p></pre>
<span style=3D"font-size:10.0pt;font-family:&quot;UICTFontTextStyleBody&quot=
;,&quot;serif&quot;"><br clear=3D"all" style=3D"page-break-before:always">
</span>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;v=3D0&nbsp;</span><o:p></o=
:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;o=3Dmascha 2980675221 2980=
675778 IN IP4 <a href=3D"http://host.example.net">host.example.net</a>&nbsp;=
</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;c=3DIN IP4 192.0.2.0&nbsp;=
</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;m=3Dvideo 49170 RTP/AVPF 9=
6 97&nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Drtpmap:96 MP4V-ES/9000=
0&nbsp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Drtcp-fb:96 nack&nbsp;<=
/span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dfmtp:96 profile-level-=
id=3D8;config=3D01010000012000884006682C2090A21F&nbsp;</span><o:p></o:p></pr=
e>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Drtpmap:97 rtx/90000&nb=
sp;</span><o:p></o:p></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">&nbsp;a=3Dfmtp:97 apt=3D96;rtx-t=
ime=3D3000</span><o:p></o:p></pre>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The situation would be the same for FEC groups create=
d via ssrc-group:FEC. From&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><a href=3D"https://tools.ietf.org/html/draft-lennox-p=
ayload-ulp-ssrc-mux">https://tools.ietf.org/html/draft-lennox-payload-ulp-ss=
rc-mux</a> &nbsp;:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<pre style=3D"page-break-before:always"><span style=3D"font-family:&quot;UIC=
TFontTextStyleBody&quot;,&quot;serif&quot;">The FEC and the payload MAY also=
 be multiplexed by SSRC into one single RTP session, with separate SSRC valu=
es, if the association between FEC and payload streams are communicated to a=
ll members of the session. If SDP is used, this association MAY be communica=
ted through the FEC ssrc-group semantic [<a href=3D"https://tools.ietf.org/h=
tml/rfc5576" title=3D"&quot;Source-Specific Media Attributes in the Session D=
escription Protocol (SDP)&quot;">RFC5576</a>]; other mechanisms are also pos=
sible. Receivers MUST NOT attempt to interpret FEC streams for which they do=
 not have information to associate them with the corresponding payload strea=
ms.</span><o:p></o:p></pre>
</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" style=3D"margin-bottom:12.0pt"><br>
On Oct 21, 2014, at 11:36 AM, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:=
ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p>I hope I can rule my SDP logic based on standards rather than on how Chro=
me implements some features.<o:p></o:p></p>
<p>My question was generic: if I receive the above SDP, how do I know which p=
ayloads each ssrc is supposed to transport?<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 21 Oct 2014 19:57, "Peter Thatcher" &lt;<a href=3D=
"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; wrote:<o:p></o:p>=
</p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;sa=
ns-serif&quot;">I'm not sure where it's specified, but here's where it's imp=
lemented:<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;sa=
ns-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;sa=
ns-serif&quot;"><a href=3D"https://code.google.com/p/webrtc/source/browse/tr=
unk/talk/media/webrtc/webrtcvideoengine.cc#4108" target=3D"_blank">https://c=
ode.google.com/p/webrtc/source/browse/trunk/talk/media/webrtc/webrtcvideoeng=
ine.cc#4108</a></span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;sa=
ns-serif&quot;">It only supports 2 SSRCs for a FID group.&nbsp; It would ign=
ore any more than that.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Oct 21, 2014 at 10:40 AM, I=C3=B1aki Baz Cast=
illo &lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a=
>&gt; wrote:<o:p></o:p></p>
<p>How is that? Where is that specified? What about if I include 3 ssrc valu=
es in the ssrc-group? What is each one for?<o:p></o:p></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On 21 Oct 2014 19:31, "Peter Thatcher" &lt;<a href=3D=
"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt;=
 wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:9.5pt;font-family:&quot;Aria=
l&quot;,&quot;sans-serif&quot;">345259865 is "real"</span><span style=3D"fon=
t-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;sa=
ns-serif&quot;"><a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> i=
s rtx</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Oct 21, 2014 at 9:25 AM, I=C3=B1aki Baz Casti=
llo &lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>=
&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" target=3D"_bl=
ank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for<br>
RTX?<br>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96<br>
a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" target=3D"_blank">26=
93756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> cname:e=
rS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> msid:DW=
pWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
c0134f05-e7c2-4afd-a979-4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> mslabel=
:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5<br>
a=3Dssrc:<a href=3D"tel:2693756249" target=3D"_blank">2693756249</a> label:c=
0134f05-e7c2-4afd-a979-4e224de5eb91<br>
-------------------------------<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank"=
>http://tools.ietf.org/html/rfc5576#section-4.2</a><br>
<br>
--------------------------------------------------<br>
4.2.&nbsp; The "ssrc-group" Media Attribute<br>
<br>
&nbsp; &nbsp;a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br>
<br>
&nbsp; &nbsp;[..]<br>
<br>
&nbsp; &nbsp;The &lt;semantics&gt; parameter is taken from the specification=
 of the<br>
&nbsp; &nbsp;"group" attribute [RFC3388].&nbsp; The initial semantic values d=
efined for<br>
&nbsp; &nbsp;the "ssrc-group" attribute are FID (Flow Identification) [RFC33=
88]<br>
&nbsp; &nbsp;and FEC (Forward Error Correction) [RFC4756].&nbsp; In each cas=
e, the<br>
&nbsp; &nbsp;relationship among the grouped sources is the same as the<br>
&nbsp; &nbsp;relationship among corresponding sources in media streams group=
ed<br>
&nbsp; &nbsp;using the SDP "group" attribute.<br>
--------------------------------------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
---------------------------------------------------<br>
7.4 FID Semantics<br>
<br>
&nbsp; &nbsp;Several "m" lines grouped together using FID semantics form a m=
edia<br>
&nbsp; &nbsp;flow.&nbsp; A media agent handling a media flow that comprises s=
everal "m"<br>
&nbsp; &nbsp;lines MUST send a copy of the media to every "m" line part of t=
he<br>
&nbsp; &nbsp;flow as long as the codecs and the direction attribute present i=
n a<br>
&nbsp; &nbsp;particular "m" line allow it.<br>
<br>
&nbsp; &nbsp;It is assumed that the application uses only one codec at a tim=
e to<br>
&nbsp; &nbsp;encode the media produced.&nbsp; This codec MAY change dynamica=
lly during<br>
&nbsp; &nbsp;the session, but at any particular moment only one codec is in u=
se.<br>
<br>
&nbsp; &nbsp;The application encodes the media using the current codec and c=
hecks<br>
&nbsp; &nbsp;one by one all the "m" lines that are part of the flow.&nbsp; I=
f a<br>
&nbsp; &nbsp;particular "m" line contains the codec being used and the direc=
tion<br>
&nbsp; &nbsp;attribute is "sendonly" or "sendrecv", a copy of the encoded me=
dia is<br>
&nbsp; &nbsp;sent to the address/port specified in that particular media str=
eam.<br>
&nbsp; &nbsp;If either the "m" line does not contain the codec being used or=
 the<br>
&nbsp; &nbsp;direction attribute is neither "sendonly" nor "sendrecv", nothi=
ng is<br>
&nbsp; &nbsp;sent over this media stream.<br>
----------------------------------------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in<br>
the above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br>=

<span style=3D"color:#888888"><br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
<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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a></span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</blockquote>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">_______________________________________________<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">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
</blockquote>
</div>
</div>


</div></blockquote></body></html>=

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--_000_E1FE4C082A89A246A11D7F32A95A17828E5ECD15US70UWXCHMBA02z_--


From nobody Wed Oct 22 18:22:09 2014
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Sergio Garcia Murillo wrote:
> BTW, please, please, please, let's use the sssrc-group:FEC and send the
> FEC in its own ssrc stream, so we don't have to use RED anymore..

That may be fine for video, but the added header overhead for audio is 
substantial (and there are a number of other problems that RED solves 
there, as discussed at the last meeting).


From nobody Thu Oct 23 00:41:33 2014
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Date: Thu, 23 Oct 2014 09:41:23 +0200
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From: Mohammed Raad <mohammedsraad@raadtech.com>
To: Stephan Wenger <stewe@stewe.org>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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Hi Stephan,

The relevant section in the ISO/IEC directives
<http://www.iec.ch/members_experts/refdocs/iec/isoiecdir-1%7Bed11.0%7Den.pd=
f>
is:

2.14.3 Should it be revealed after publication of a document that licences
under patent
rights, which appear to cover items included in the document, cannot be
obtained under
reasonable and non-discriminatory terms and conditions, the document shall
be referred back
to the relevant committee for further consideration.

It should be clear to participants in this discussion that not listing any
patents in a type 3 declaration makes it difficult to claim that it has
been "revealed after publication of a document that licences under patent
rights, which appear to cover items included in the document,  cannot be
obtained".

So, yes, ISO does have a way of dealing with such situations. MPEG,
specifically, has dealt with such a situation during the development of the
CDVS standard, which is a recent standard. The path chosen in that instance
was to work around the known unlicensable patent.

If one was to accept the idea that a type-3 declaration (i.e. there will be
no licensing) without being told as to what patent the claim refers to
should stop the progress of a standard, then it would be impossible to
develop any standard.

Regards,
Mohammed

On Wed, Oct 22, 2014 at 9:45 PM, Stephan Wenger <stewe@stewe.org> wrote:

> Hi,
> I have to make one correction in the light of information that has
> surfaced at the MPEG meeting currently ongoing in Strasbourg.  (I=E2=80=
=99m not at
> that meeting this time, but a colleague is and she briefed me.)
> Nokia has made MPEG and ISO/IEC officially aware that they are not willin=
g
> to license essential patents under RAND terms.  For those with MPEG
> document access, please see M34917.  The official declaration is dated
> 9/19/2014, and is not yet available from the respective databases, as ISO
> is apparently changing its recordation infrastructure.
> My understanding of the joint ITU/ISO/IEC patent policy is that no
> standard can be issued that has a type 3 declaration against it.  To the
> best of my knowledge, ISO has no established procedure how to deal with
> type 3 (non-RAND) declarations and still keep the standard project going.
> Unlike, for example, W3C and its Patent Advisory Groups.
> The declaration does not list specific patents. To the best of my
> knowledge, such info is not required (only desired) for ISO and IEC
> standardization work--one of the few differences in patent policy
> guidelines between ITU and ISO/IEC.
> Therefore, I have to row back on my previous statement of likeliness of
> having an ISO number for VP8 anytime soon.  At this point, I just don=E2=
=80=99t
> know whether, if ever, that will happen.
> Regards,
> Stephan
>
>
> On 10/19/14, 6:10 PM, "Stephan Wenger" <stewe@stewe.org> wrote:
>
> >Hi,
> >
> >On 10/19/14, 9:15 AM, "Watson Ladd" <watsonbladd@gmail.com> wrote:
> >
> >>On Sun, Oct 19, 2014 at 9:13 AM, Alexandre GOUAILLARD
> >><agouaillard@gmail.com> wrote:
> >>> @jonathan,
> >>>
> >>> while you are right and availability of 264 implementation or hardwar=
e
> >>> acceleration has improved, it has never been reported as a problem in
> >>>the
> >>> previous pool by anyone. The 264 royalties, and the VP8 IP risks were=
,
> >>> AFAIR, the main reasons used by individuals to justify their position=
s.
> >>> Today, nothing has changed with respect to those two items (even thou=
gh
> >>> providing open264 royalties and absorbing the license cost for some
> >>> platforms might have been a set in the right direction). This is why =
I
> >>>think
> >>> the conditions are not met for a consensus to be reached.
> >>
> >>But now VP8 is going through ISO,
> >
> >... and is is DIS ballot.  Few projects in ISO get stopped at that stage=
.
> >To me, it=C2=B9s pretty clear that VP8 will have an ISO/IEC blessing wit=
hin a
> >year or two.  Without substantial technical changes.  Given the very
> >limited participation in the relevant subgroup in MPEG, it=C2=B9s unclea=
r to me
> >what good that will do, though.
> >
> >>and the people claiming patents on
> >>VP8 have had time to sue, and haven't.
> >
> >That=C2=B9s factually incorrect.  To the best of my knowledge, what woul=
d be
> >factually correct is this: in two cases, companies have been sued over
> >patents allegedly reading on VP8 in the context of the wider =C2=B3smart=
phone
> >wars=C2=B2 lawsuits, and the defendants have won non-infringement ruling=
s in
> >the first instance (though, last I looked, appeals were pending in both
> >cases).  At least one other case was settled on undisclosed terms.  Some
> >of these cases were widely reported in the press, others are a little bi=
t
> >harder to find without access to legal search tools.
> >
> >>That's evidence that some
> >>concerns are overblown.
> >
> >And that depends on your viewpoint.
> >
> >Stephan
> >
> >>
> >>>
> >>> Alex.
> >>>
> >>>
> >>> On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba
> >>><bernard.aboba@gmail.com>
> >>> wrote:
> >>>>
> >>>> "And its one of the issues holding up wider adoption of the
> >>>>technology"
> >>>>
> >>>> [BA] Specifying an MTI encoder/decoder is not sufficient for
> >>>> interoperability.  It is also necessary to specify the transport in
> >>>>enough
> >>>> detail to allow independent implementations that interoperate well
> >>>>enough to
> >>>> be usable in a wide variety of scenarios, including wireless network=
s
> >>>>where
> >>>> loss is commonly experienced.
> >>>>
> >>>> We made the mistake of having an MTI discussion previously with not
> >>>>enough
> >>>> details on that subject, particularly with respect to H.264.
> >>>> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
> >>>>
> >>>> So if we are to have the discussion again, it should occur in the
> >>>>context
> >>>> of detailed specifications and interoperability reports.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg
> >>>><jdrosen@jdrosen.net>
> >>>> wrote:
> >>>>>
> >>>>> I'm in favor of taking another run at this.
> >>>>>
> >>>>> The working group has repeatedly said that an MTI codec is somethin=
g
> >>>>>we
> >>>>> need to produce. And its one of the issues holding up wider adoptio=
n
> >>>>>of the
> >>>>> technology (not the only one for sure).
> >>>>>
> >>>>> And things have changed since the last meeting, a year ago now
> >>>>>(November
> >>>>> in Vancouver). Cisco's open264 plugin shipped and now just recently
> >>>>>is
> >>>>> integrated into Firefox. iOS8 shipped with APIs for H264. There are
> >>>>>other
> >>>>> things worth noting. Will this change the minds of everyone? Surely
> >>>>>not.
> >>>>> Will it sway enough for us to achieve rough consensus? Maybe. IMHO =
-
> >>>>>worth
> >>>>> finding out.
> >>>>>
> >>>>> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD
> >>>>> <agouaillard@gmail.com> wrote:
> >>>>>>
> >>>>>> +1 to not having MTI codec discussion unless some progress has bee=
n
> >>>>>>made
> >>>>>> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling th=
at
> >>>>>>there
> >>>>>> was no change on the members position.
> >>>>>>
> >>>>>> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand
> >>>>>> <harald@alvestrand.no> wrote:
> >>>>>>>
> >>>>>>> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
> >>>>>>>>
> >>>>>>>> One thing we could do instead of wasting time on MTI is to
> >>>>>>>>actually
> >>>>>>>> make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, =
so
> >>>>>>>>we could
> >>>>>>>> actually interoperate regardless of the codec.
> >>>>>>>
> >>>>>>>
> >>>>>>> The big argument for an MTI is actually the one stated in
> >>>>>>>-overview: It
> >>>>>>> guards against interoperability failure.
> >>>>>>>
> >>>>>>> I would like to have an ecosystem where one can field a box,
> >>>>>>>connect it
> >>>>>>> to everything else, and run well for *some* level of "well" - and=
 I
> >>>>>>>would
> >>>>>>> prefer that ecosystem to be one where it's possible to field the
> >>>>>>>box without
> >>>>>>> making prior arrangements with anyone about licenses.
> >>>>>>>
> >>>>>>> This argument hasn't changed one whit since last time we discusse=
d
> >>>>>>>it.
> >>>>>>> And I don't see much movement on the specifics of the proposals
> >>>>>>>either.
> >>>>>>>
> >>>>>>> We'll have to interoperate well with the codecs we field. So usin=
g
> >>>>>>>some
> >>>>>>> time to discuss draft-ietf-rtcweb-video seems to make sense. (And
> >>>>>>>4.1 isn't
> >>>>>>> finished either. There's one sentence that needs to be removed.)
> >>>>>>>
> >>>>>>> I wouldn't say I'd be happy to not discuss this in Honolulu. But
> >>>>>>>I'd
> >>>>>>> prefer to re-discuss based on the knowledge that some significant
> >>>>>>>players
> >>>>>>> have actually changed their minds.
> >>>>>>>
> >>>>>>> At the moment, I don't see signs that any of the poll respondents
> >>>>>>>have
> >>>>>>> said "I have changed my mind".
> >>>>>>>
> >>>>>>> Harald
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>> On Oct 16, 2014, at 2:28 PM, Martin Thomson
> >>>>>>>>> <martin.thomson@gmail.com> wrote:
> >>>>>>>>>
> >>>>>>>>>> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at>
> >>>>>>>>>> wrote:
> >>>>>>>>>> And that's because something substantive has changed, or simpl=
y
> >>>>>>>>>> because
> >>>>>>>>>> wasting the WG time on this again is more entertaining than
> >>>>>>>>>>actually
> >>>>>>>>>> finishing a specification that can be independently implemente=
d
> >>>>>>>>>>by
> >>>>>>>>>> all
> >>>>>>>>>> browser vendors? (A specification that we are nowhere near
> >>>>>>>>>>having,
> >>>>>>>>>> as far as
> >>>>>>>>>> I can tell)
> >>>>>>>>>
> >>>>>>>>> Personally, I've found the reprieve from this fight refreshing.
> >>>>>>>>>And
> >>>>>>>>> it would appear that we've made some real progress as a result.
> >>>>>>>>>I'd
> >>>>>>>>> suggest that if we don't have new information, we continue to
> >>>>>>>>>spend
> >>>>>>>>> our time productively.  If we can't find topics to occupy our
> >>>>>>>>>meeting
> >>>>>>>>> agenda time, then maybe we can free an agenda slot.
> >>>>>>>>>
> >>>>>>>>> _______________________________________________
> >>>>>>>>> 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
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> --
> >>>>>> Alex. Gouaillard, PhD, PhD, MBA
> >>>>>>
> >>>>>>
> >>>>>>-------------------------------------------------------------------=
--
> >>>>>>-
> >>>>>>--------------
> >>>>>> CTO - Temasys Communications, S'pore / Mountain View
> >>>>>> President - CoSMo Software, Cambridge, MA
> >>>>>>
> >>>>>>
> >>>>>>-------------------------------------------------------------------=
--
> >>>>>>-
> >>>>>>--------------
> >>>>>> sg.linkedin.com/agouaillard
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> rtcweb mailing list
> >>>>>> rtcweb@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> --
> >>>>> Jonathan Rosenberg, Ph.D.
> >>>>> jdrosen@jdrosen.net
> >>>>> http://www.jdrosen.net
> >>>>>
> >>>>> _______________________________________________
> >>>>> rtcweb mailing list
> >>>>> rtcweb@ietf.org
> >>>>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>>>>
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>> Alex. Gouaillard, PhD, PhD, MBA
> >>>
> >>>----------------------------------------------------------------------=
--
> >>>-
> >>>-----------
> >>> CTO - Temasys Communications, S'pore / Mountain View
> >>> President - CoSMo Software, Cambridge, MA
> >>>
> >>>----------------------------------------------------------------------=
--
> >>>-
> >>>-----------
> >>> sg.linkedin.com/agouaillard
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>>
> >>
> >>
> >>
> >>--
> >>"Those who would give up Essential Liberty to purchase a little
> >>Temporary Safety deserve neither  Liberty nor Safety."
> >>-- Benjamin Franklin
> >>
> >>_______________________________________________
> >>rtcweb mailing list
> >>rtcweb@ietf.org
> >>https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >_______________________________________________
> >rtcweb mailing list
> >rtcweb@ietf.org
> >https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



--=20
Mohammed Raad, PhD.
Partner
RAADTECH CONSULTING
P.O. Box 113
Warrawong
NSW 2502 Australia
Phone: +61 414451478
Email: mohammedsraad@raadtech.com

--047d7b62533ce786f90506123088
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<div dir=3D"ltr">Hi Stephan,<div><br></div><div>The relevant section in the=
 ISO/IEC <a href=3D"http://www.iec.ch/members_experts/refdocs/iec/isoiecdir=
-1%7Bed11.0%7Den.pdf">directives</a> is:</div><div><br></div><div><div>2.14=
.3 Should it be revealed after publication of a document that licences unde=
r patent=C2=A0</div><div>rights, which appear to cover items included in th=
e document, cannot be obtained under=C2=A0</div><div>reasonable and non-dis=
criminatory terms and conditions, the document shall be referred back=C2=A0=
</div><div>to the relevant committee for further consideration.</div></div>=
<div><br></div><div>It should be clear to participants in this discussion t=
hat not listing any patents in a type 3 declaration makes it difficult to c=
laim that it has been &quot;revealed after publication of a document that l=
icences under patent=C2=A0</div><div>rights, which appear to cover items in=
cluded in the document,=C2=A0=C2=A0cannot be obtained&quot;.</div><div><br>=
</div><div>So, yes, ISO does have a way of dealing with such situations. MP=
EG, specifically, has dealt with such a situation during the development of=
 the CDVS standard, which is a recent standard. The path chosen in that ins=
tance was to work around the known unlicensable patent.</div><div><br></div=
><div>If one was to accept the idea that a type-3 declaration (i.e. there w=
ill be no licensing) without being told as to what patent the claim refers =
to should stop the progress of a standard, then it would be impossible to d=
evelop any standard.</div><div><br></div><div>Regards,</div><div>Mohammed</=
div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed,=
 Oct 22, 2014 at 9:45 PM, Stephan Wenger <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:stewe@stewe.org" target=3D"_blank">stewe@stewe.org</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex">Hi,<br>
I have to make one correction in the light of information that has<br>
surfaced at the MPEG meeting currently ongoing in Strasbourg.=C2=A0 (I=E2=
=80=99m not at<br>
that meeting this time, but a colleague is and she briefed me.)<br>
Nokia has made MPEG and ISO/IEC officially aware that they are not willing<=
br>
to license essential patents under RAND terms.=C2=A0 For those with MPEG<br=
>
document access, please see M34917.=C2=A0 The official declaration is dated=
<br>
9/19/2014, and is not yet available from the respective databases, as ISO<b=
r>
is apparently changing its recordation infrastructure.<br>
My understanding of the joint ITU/ISO/IEC patent policy is that no<br>
standard can be issued that has a type 3 declaration against it.=C2=A0 To t=
he<br>
best of my knowledge, ISO has no established procedure how to deal with<br>
type 3 (non-RAND) declarations and still keep the standard project going.<b=
r>
Unlike, for example, W3C and its Patent Advisory Groups.<br>
The declaration does not list specific patents. To the best of my<br>
knowledge, such info is not required (only desired) for ISO and IEC<br>
standardization work--one of the few differences in patent policy<br>
guidelines between ITU and ISO/IEC.<br>
Therefore, I have to row back on my previous statement of likeliness of<br>
having an ISO number for VP8 anytime soon.=C2=A0 At this point, I just don=
=E2=80=99t<br>
know whether, if ever, that will happen.<br>
Regards,<br>
Stephan<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 10/19/14, 6:10 PM, &quot;Stephan Wenger&quot; &lt;<a href=3D"mailto:stew=
e@stewe.org">stewe@stewe.org</a>&gt; wrote:<br>
<br>
&gt;Hi,<br>
&gt;<br>
&gt;On 10/19/14, 9:15 AM, &quot;Watson Ladd&quot; &lt;<a href=3D"mailto:wat=
sonbladd@gmail.com">watsonbladd@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt;On Sun, Oct 19, 2014 at 9:13 AM, Alexandre GOUAILLARD<br>
&gt;&gt;&lt;<a href=3D"mailto:agouaillard@gmail.com">agouaillard@gmail.com<=
/a>&gt; wrote:<br>
&gt;&gt;&gt; @jonathan,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; while you are right and availability of 264 implementation or =
hardware<br>
&gt;&gt;&gt; acceleration has improved, it has never been reported as a pro=
blem in<br>
&gt;&gt;&gt;the<br>
&gt;&gt;&gt; previous pool by anyone. The 264 royalties, and the VP8 IP ris=
ks were,<br>
&gt;&gt;&gt; AFAIR, the main reasons used by individuals to justify their p=
ositions.<br>
&gt;&gt;&gt; Today, nothing has changed with respect to those two items (ev=
en though<br>
&gt;&gt;&gt; providing open264 royalties and absorbing the license cost for=
 some<br>
&gt;&gt;&gt; platforms might have been a set in the right direction). This =
is why I<br>
&gt;&gt;&gt;think<br>
&gt;&gt;&gt; the conditions are not met for a consensus to be reached.<br>
&gt;&gt;<br>
&gt;&gt;But now VP8 is going through ISO,<br>
&gt;<br>
&gt;... and is is DIS ballot.=C2=A0 Few projects in ISO get stopped at that=
 stage.<br>
&gt;To me, it=C2=B9s pretty clear that VP8 will have an ISO/IEC blessing wi=
thin a<br>
&gt;year or two.=C2=A0 Without substantial technical changes.=C2=A0 Given t=
he very<br>
&gt;limited participation in the relevant subgroup in MPEG, it=C2=B9s uncle=
ar to me<br>
&gt;what good that will do, though.<br>
&gt;<br>
&gt;&gt;and the people claiming patents on<br>
&gt;&gt;VP8 have had time to sue, and haven&#39;t.<br>
&gt;<br>
&gt;That=C2=B9s factually incorrect.=C2=A0 To the best of my knowledge, wha=
t would be<br>
&gt;factually correct is this: in two cases, companies have been sued over<=
br>
&gt;patents allegedly reading on VP8 in the context of the wider =C2=B3smar=
tphone<br>
&gt;wars=C2=B2 lawsuits, and the defendants have won non-infringement rulin=
gs in<br>
&gt;the first instance (though, last I looked, appeals were pending in both=
<br>
&gt;cases).=C2=A0 At least one other case was settled on undisclosed terms.=
=C2=A0 Some<br>
&gt;of these cases were widely reported in the press, others are a little b=
it<br>
&gt;harder to find without access to legal search tools.<br>
&gt;<br>
&gt;&gt;That&#39;s evidence that some<br>
&gt;&gt;concerns are overblown.<br>
&gt;<br>
&gt;And that depends on your viewpoint.<br>
&gt;<br>
&gt;Stephan<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Alex.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Sun, Oct 19, 2014 at 11:43 PM, Bernard Aboba<br>
&gt;&gt;&gt;&lt;<a href=3D"mailto:bernard.aboba@gmail.com">bernard.aboba@gm=
ail.com</a>&gt;<br>
&gt;&gt;&gt; wrote:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; &quot;And its one of the issues holding up wider adoption =
of the<br>
&gt;&gt;&gt;&gt;technology&quot;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; [BA] Specifying an MTI encoder/decoder is not sufficient f=
or<br>
&gt;&gt;&gt;&gt; interoperability.=C2=A0 It is also necessary to specify th=
e transport in<br>
&gt;&gt;&gt;&gt;enough<br>
&gt;&gt;&gt;&gt; detail to allow independent implementations that interoper=
ate well<br>
&gt;&gt;&gt;&gt;enough to<br>
&gt;&gt;&gt;&gt; be usable in a wide variety of scenarios, including wirele=
ss networks<br>
&gt;&gt;&gt;&gt;where<br>
&gt;&gt;&gt;&gt; loss is commonly experienced.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; We made the mistake of having an MTI discussion previously=
 with not<br>
&gt;&gt;&gt;&gt;enough<br>
&gt;&gt;&gt;&gt; details on that subject, particularly with respect to H.26=
4.<br>
&gt;&gt;&gt;&gt; draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy =
at best.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; So if we are to have the discussion again, it should occur=
 in the<br>
&gt;&gt;&gt;&gt;context<br>
&gt;&gt;&gt;&gt; of detailed specifications and interoperability reports.<b=
r>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg<br>
&gt;&gt;&gt;&gt;&lt;<a href=3D"mailto:jdrosen@jdrosen.net">jdrosen@jdrosen.=
net</a>&gt;<br>
&gt;&gt;&gt;&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; I&#39;m in favor of taking another run at this.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; The working group has repeatedly said that an MTI code=
c is something<br>
&gt;&gt;&gt;&gt;&gt;we<br>
&gt;&gt;&gt;&gt;&gt; need to produce. And its one of the issues holding up =
wider adoption<br>
&gt;&gt;&gt;&gt;&gt;of the<br>
&gt;&gt;&gt;&gt;&gt; technology (not the only one for sure).<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; And things have changed since the last meeting, a year=
 ago now<br>
&gt;&gt;&gt;&gt;&gt;(November<br>
&gt;&gt;&gt;&gt;&gt; in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently<br>
&gt;&gt;&gt;&gt;&gt;is<br>
&gt;&gt;&gt;&gt;&gt; integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are<br>
&gt;&gt;&gt;&gt;&gt;other<br>
&gt;&gt;&gt;&gt;&gt; things worth noting. Will this change the minds of eve=
ryone? Surely<br>
&gt;&gt;&gt;&gt;&gt;not.<br>
&gt;&gt;&gt;&gt;&gt; Will it sway enough for us to achieve rough consensus?=
 Maybe. IMHO -<br>
&gt;&gt;&gt;&gt;&gt;worth<br>
&gt;&gt;&gt;&gt;&gt; finding out.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD<=
br>
&gt;&gt;&gt;&gt;&gt; &lt;<a href=3D"mailto:agouaillard@gmail.com">agouailla=
rd@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; +1 to not having MTI codec discussion unless some =
progress has been<br>
&gt;&gt;&gt;&gt;&gt;&gt;made<br>
&gt;&gt;&gt;&gt;&gt;&gt; on VP8 at MPEG. Any news on that? I&#39;m sharing =
harald&#39;s=C2=A0 feeling that<br>
&gt;&gt;&gt;&gt;&gt;&gt;there<br>
&gt;&gt;&gt;&gt;&gt;&gt; was no change on the members position.<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand=
<br>
&gt;&gt;&gt;&gt;&gt;&gt; &lt;<a href=3D"mailto:harald@alvestrand.no">harald=
@alvestrand.no</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; On 10/17/2014 12:02 AM, Bernard Aboba wrote:<b=
r>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; One thing we could do instead of wasting t=
ime on MTI is to<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;actually<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; make progress on Sections 4.2 - 4.4 of dra=
ft-IETF-RTCWEB-video, so<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;we could<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; actually interoperate regardless of the co=
dec.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; The big argument for an MTI is actually the on=
e stated in<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;-overview: It<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; guards against interoperability failure.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; I would like to have an ecosystem where one ca=
n field a box,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;connect it<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; to everything else, and run well for *some* le=
vel of &quot;well&quot; - and I<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;would<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; prefer that ecosystem to be one where it&#39;s=
 possible to field the<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;box without<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; making prior arrangements with anyone about li=
censes.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; This argument hasn&#39;t changed one whit sinc=
e last time we discussed<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;it.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; And I don&#39;t see much movement on the speci=
fics of the proposals<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;either.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; We&#39;ll have to interoperate well with the c=
odecs we field. So using<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;some<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; time to discuss draft-ietf-rtcweb-video seems =
to make sense. (And<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;4.1 isn&#39;t<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; finished either. There&#39;s one sentence that=
 needs to be removed.)<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; I wouldn&#39;t say I&#39;d be happy to not dis=
cuss this in Honolulu. But<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;I&#39;d<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; prefer to re-discuss based on the knowledge th=
at some significant<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;players<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; have actually changed their minds.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; At the moment, I don&#39;t see signs that any =
of the poll respondents<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;have<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; said &quot;I have changed my mind&quot;.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Harald<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; On Oct 16, 2014, at 2:28 PM, Martin Th=
omson<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;<a href=3D"mailto:martin.thomson@g=
mail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; On 16 October 2014 14:17, Matthew =
Kaufman &lt;<a href=3D"mailto:matthew@matthew.at">matthew@matthew.at</a>&gt=
;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; And that&#39;s because something s=
ubstantive has changed, or simply<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; because<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; wasting the WG time on this again =
is more entertaining than<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;actually<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; finishing a specification that can=
 be independently implemented<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;by<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; all<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; browser vendors? (A specification =
that we are nowhere near<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;having,<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; as far as<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I can tell)<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Personally, I&#39;ve found the repriev=
e from this fight refreshing.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;And<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; it would appear that we&#39;ve made so=
me real progress as a result.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;I&#39;d<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; suggest that if we don&#39;t have new =
information, we continue to<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;spend<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; our time productively.=C2=A0 If we can=
&#39;t find topics to occupy our<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;meeting<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; agenda time, then maybe we can free an=
 agenda slot.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; ______________________________________=
_________<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtc=
web@ietf.org</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailma=
n/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/=
rtcweb</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; __________________________________________=
_____<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@=
ietf.org</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/li=
stinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcw=
eb</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; ______________________________________________=
_<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listin=
fo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</=
a><br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; --<br>
&gt;&gt;&gt;&gt;&gt;&gt; Alex. Gouaillard, PhD, PhD, MBA<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;---------------------------------------------------=
------------------<br>
&gt;&gt;&gt;&gt;&gt;&gt;-<br>
&gt;&gt;&gt;&gt;&gt;&gt;--------------<br>
&gt;&gt;&gt;&gt;&gt;&gt; CTO - Temasys Communications, S&#39;pore / Mountai=
n View<br>
&gt;&gt;&gt;&gt;&gt;&gt; President - CoSMo Software, Cambridge, MA<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;---------------------------------------------------=
------------------<br>
&gt;&gt;&gt;&gt;&gt;&gt;-<br>
&gt;&gt;&gt;&gt;&gt;&gt;--------------<br>
&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"http://sg.linkedin.com/agouaillard" tar=
get=3D"_blank">sg.linkedin.com/agouaillard</a><br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; _______________________________________________<br=
>
&gt;&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt;&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><b=
r>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; --<br>
&gt;&gt;&gt;&gt;&gt; Jonathan Rosenberg, Ph.D.<br>
&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:jdrosen@jdrosen.net">jdrosen@jdrosen=
.net</a><br>
&gt;&gt;&gt;&gt;&gt; <a href=3D"http://www.jdrosen.net" target=3D"_blank">h=
ttp://www.jdrosen.net</a><br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>=
<br>
&gt;&gt;&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcwe=
b" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; --<br>
&gt;&gt;&gt; Alex. Gouaillard, PhD, PhD, MBA<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;---------------------------------------------------------------=
---------<br>
&gt;&gt;&gt;-<br>
&gt;&gt;&gt;-----------<br>
&gt;&gt;&gt; CTO - Temasys Communications, S&#39;pore / Mountain View<br>
&gt;&gt;&gt; President - CoSMo Software, Cambridge, MA<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;---------------------------------------------------------------=
---------<br>
&gt;&gt;&gt;-<br>
&gt;&gt;&gt;-----------<br>
&gt;&gt;&gt; <a href=3D"http://sg.linkedin.com/agouaillard" target=3D"_blan=
k">sg.linkedin.com/agouaillard</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" targe=
t=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;--<br>
&gt;&gt;&quot;Those who would give up Essential Liberty to purchase a littl=
e<br>
&gt;&gt;Temporary Safety deserve neither=C2=A0 Liberty nor Safety.&quot;<br=
>
&gt;&gt;-- Benjamin Franklin<br>
&gt;&gt;<br>
&gt;&gt;_______________________________________________<br>
&gt;&gt;rtcweb mailing list<br>
&gt;&gt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&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"_bla=
nk">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
Mohammed Raad, PhD.<br>Partner<br>RAADTECH CONSULTING<br>P.O. Box 113<br>Wa=
rrawong<br>NSW 2502 Australia<br>Phone: +61 414451478<br>Email: <a href=3D"=
mailto:mohammedsraad@raadtech.com" target=3D"_blank">mohammedsraad@raadtech=
.com</a>
</div>

--047d7b62533ce786f90506123088--


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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: Harald Alvestrand <harald@alvestrand.no>, Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
Thread-Index: AQHP69ySvH0vdbAWv0eULyOYAeS/S5w4ZQqA//+vamCABWG9QA==
Date: Thu, 23 Oct 2014 10:15:56 +0000
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Subject: Re: [rtcweb] Plan for MTI video codec?
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From nobody Thu Oct 23 05:27:16 2014
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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Thu, 23 Oct 2014 14:26:59 +0200
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To: Benjamin Schwartz <bemasc@webrtc.org>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-07.txt
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On 22 Oct 2014, at 16:57, Benjamin Schwartz <bemasc@webrtc.org> wrote:

> FYI, the purpose of the RETURN draft =
(https://tools.ietf.org/html/draft-schwartz-rtcweb-return-03) is to try =
to specify the precise interaction and precedence when both the browser =
and the application provide TURN servers.  (It's not as simple as "one =
or the other".)
>=20
> I haven't considered the issues related to browser-provided STUN =
servers, though.
Cool! I'll take a look at that. Maybe Haralds draft need to refer to =
that.

/O
>=20
> On Wed, Oct 22, 2014 at 10:51 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> The advantage of an application configuration is that it can be =
dynamically provided/updated, based on location based etc - assuming the =
webpage provider has knowledge about STUN/TURN servers, of course.
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Olle E. =
Johansson
> Sent: 22 October 2014 17:44
> To: rtcweb@ietf.org
> Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt
>=20
> Section 3.4
>=20
> " WebRTC browsers MUST support configuration of STUN and TURN servers,
>    both from browser configuration and from an application."
>=20
>=20
> Should we mention which takes precedence? If configured in the =
browser, start with that server.
> If not use the one provided by the application.
>=20
> Maybe we should clarify that turn DNS discovery MUST be used to =
provide failover and load balancing.
>=20
> /O
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


--Apple-Mail=_E7B0BEFE-F6BC-48A3-B724-BB993AF9C57C
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On 22 Oct 2014, at 16:57, Benjamin =
Schwartz &lt;<a =
href=3D"mailto:bemasc@webrtc.org">bemasc@webrtc.org</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div dir=3D"ltr">FYI, the purpose of the RETURN draft (<a =
href=3D"https://tools.ietf.org/html/draft-schwartz-rtcweb-return-03">https=
://tools.ietf.org/html/draft-schwartz-rtcweb-return-03</a>) is to try to =
specify the precise interaction and precedence when both the browser and =
the application provide TURN servers. &nbsp;(It's not as simple as "one =
or the other".)<div><br></div><div>I haven't considered the issues =
related to browser-provided STUN servers, =
though.</div></div></blockquote>Cool! I'll take a look at that. Maybe =
Haralds draft need to refer to =
that.</div><div><br></div><div>/O<br><blockquote type=3D"cite"><div =
class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Oct 22, =
2014 at 10:51 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
The advantage of an application configuration is that it can be =
dynamically provided/updated, based on location based etc - assuming the =
webpage provider has knowledge about STUN/TURN servers, of course.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] On =
Behalf Of Olle E. Johansson<br>
Sent: 22 October 2014 17:44<br>
To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt<br>
<br>
Section 3.4<br>
<br>
" WebRTC browsers MUST support configuration of STUN and TURN =
servers,<br>
&nbsp; &nbsp;both from browser configuration and from an =
application."<br>
<br>
<br>
Should we mention which takes precedence? If configured in the browser, =
start with that server.<br>
If not use the one provided by the application.<br>
<br>
Maybe we should clarify that turn DNS discovery MUST be used to provide =
failover and load balancing.<br>
<br>
/O<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
</blockquote></div><br></body></html>=

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References: <CAGTXFp-HVJDwd86207PNM2QVYO4Z_K4WF-KarnRs1fb7nvy4zA@mail.gmail.com> <CA+9kkMDfES8gpi0-PTXpCnQHjFYUSF2r44TNzH5B4UfDGo8PtA@mail.gmail.com> <CAGTXFp8O-7ACksk3v3f=KjCkcDb4e8G=t-e=EJ1503vt7TkpCQ@mail.gmail.com> <CAGTXFp867AMUZ_fEKxG9uAoR1H1AirVHi3-ayJ=KTQk9L+C7+g@mail.gmail.com> <CA+9kkMAZufR7gUrwkS7Tf5GOfg+ZtsZWGcn-8YLCvnmYnTgfFw@mail.gmail.com> <544035DE.8000606@matthew.at> <CABkgnnUNgWaauS6-nZ5fcExjsMPy4ZGPXaahduzA39=iqh9+fQ@mail.gmail.com> <D5D11F2B-9E32-4932-A601-F1D7FD50C706@gmail.com> <544117FB.6050706@alvestrand.no> <CAHgZEq6GTk5ei+LLpWPM5povpieompD66VU9F+u7--WJVgapaQ@mail.gmail.com> <CA+23+fGWnWd0QEeCmZ=6BmJkPrUVW6cZ0jwmXA+fM88=_+_NWw@mail.gmail.com> <CAOW+2dugTtfLhk0VuJOk7OPEonGBApMjY93EZocH90RbX6X22w@mail.gmail.com> <54442128.6070009@alvestrand.no> <CAOW+2dt8j2VwmpeQ3qaCNOKNgrGz95Sp_ROq=FO9sNm7M2EX2w@mail.gmail.com> <E36D1A4AE0B6AA4091F1728D584A6AD2400622F1@ORSMSX158.amr.corp.intel.com> <E36D1A4AE0B6AA4091F1728D584A6AD24007E125@fmsmsx118.amr.corp.intel.com>
Date: Thu, 23 Oct 2014 16:57:35 +0200
Message-ID: <CAGgHUiRFeN6Kb44U-OCo9Y1=yMH9U8-tXxLTqaS0S_wwmxxaDA@mail.gmail.com>
From: Leon Geyser <lgeyser@gmail.com>
To: "Cavigioli, Chris" <chris.cavigioli@intel.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Plan for MTI video codec?
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In section '6 Use Cases' it seems like there isn't any transcoding required
between AMR-WB <-> EVS
Can an AMR-WB decoder decode an EVS stream?

On 23 October 2014 12:15, Cavigioli, Chris <chris.cavigioli@intel.com>
wrote:

>  The GSMA white paper has been published.  Download a copy here:
> http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/
>
>
>
> Note: Transcoding can be avoided by negotiating optional codecs and
> finding a matching pair.  The GSMA white paper doesn=E2=80=99t focus on t=
hat case,
> but instead specifically highlights the scenario where a pure WebRTC
> endpoint is communicating with a pure 3GPP LTE VoLTE/ViLTE IMS endpoint,
> assuming each endpoint ONLY has access to the MTI codecs in its respectiv=
e
> domain.  In that case, the network is responsible to transcode between
> (Opus or G.711, <WebRTC MTI video codec TBD>) and (AMR or AMR-WB, H.264).
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Cavigioli,
> Chris
> *Sent:* Sunday, October 19, 2014 6:20 PM
> *To:* Harald Alvestrand; Bernard Aboba
>
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org <rtcweb-bounces@ietf.org>]=
 *On
> Behalf Of *Bernard Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> 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
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> 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
>
>

--047d7bfd08baf1b1590506184886
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>In section &#39;6 Use Cases&#39; it seems like there =
isn&#39;t any transcoding required between AMR-WB &lt;-&gt; EVS<br></div>Ca=
n an AMR-WB decoder decode an EVS stream?<br></div><div class=3D"gmail_extr=
a"><br><div class=3D"gmail_quote">On 23 October 2014 12:15, Cavigioli, Chri=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:chris.cavigioli@intel.com" target=
=3D"_blank">chris.cavigioli@intel.com</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">





<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">The GSMA white paper has be=
en published.=C2=A0 Download a copy here:=C2=A0
<a href=3D"http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/" target=
=3D"_blank">http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/</a>
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">Note: Transcoding can be av=
oided by negotiating optional codecs and finding a matching pair.=C2=A0 The=
 GSMA white paper doesn=E2=80=99t focus on that case, but instead specifica=
lly
 highlights the scenario where a pure WebRTC endpoint is communicating with=
 a pure 3GPP LTE VoLTE/ViLTE IMS endpoint, assuming each endpoint ONLY has =
access to the MTI codecs in its respective domain.=C2=A0 In that case, the =
network is responsible to transcode between
 (Opus or G.711, &lt;WebRTC MTI video codec TBD&gt;) and (AMR or AMR-WB, H.=
264).=C2=A0 <u></u>
<u></u></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"><u></u>=C2=A0<u></u></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 [=
mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Cavigioli, Chris<br>
<b>Sent:</b> Sunday, October 19, 2014 6:20 PM<br>
<b>To:</b> Harald Alvestrand; Bernard Aboba</span></p><div><div class=3D"h5=
"><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></div><=
/div><p></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-seri=
f&quot;;color:#1f497d"><u></u><u></u></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"><u></u>=C2=A0<u></u></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">We didn=E2=80=99t have quan=
titative data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=
=93 LTE interop is important.=C2=A0 A group of us in GSMA have been looking=
 at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">WebRTC =E2=80=93 LTE transc=
oding is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processin=
g Requirements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a significant degradation
 imposed on Web =E2=80=93 LTE links if only MTI codecs are used.=C2=A0 The =
only systems to avoid this are those which implement OPTIONAL audio codecs =
to be transcoder-free with LTE.=C2=A0 The same can apply for video codecs.<=
u></u><u></u></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"><u></u>=C2=A0<u></u></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">The GSMA whitepaper refers =
to:<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Quantitative voice-ove=
r-LTE (VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0=
 To paraphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">OPUS =E2=80=93 AMR tra=
nscoding adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub>=
 + T<sub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>3.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">To put these numbers i=
n perspective, it is in general desirable to minimize UE delays to ensure l=
ow enough end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">This is a complex topic wit=
h many network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays =
(T<sub>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS =E2=80=93
 AMR transcoding in the network imposes an additional 211.5 ms on top of th=
at, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 211.5 ms ca=
n be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">CONCLUSION: =C2=A0regardles=
s of MTI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE in=
terop experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">POSITION:=C2=A0 To be clear=
, several Intel SoCs launched at MWC this year for smartphones and tablets =
support both VP8 and H.264 hardware encode and decode =E2=80=93 so Intel
 is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has high-pe=
rformance solutions for transcoding.=C2=A0 However, we wish to inform the i=
ndustry, with quantitative data, about impact to end-user experience of man=
dating such transcoding so IETF can make a
 data-driven decision on video codecs as well as reflect on the impact of a=
lready having mandated transcoding on the audio side.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Lucida Console&quot=
;;color:#1f497d">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">Intel Corp,
</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot=
;sans-serif&quot;;color:#0070c0">System Architecture and Planning</span><sp=
an style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans-seri=
f&quot;;color:#1f497d">, Video/Multimedia, Mobile and Communications Group =
(MCG)<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
15%29%20254-4545" value=3D"+14152544545" target=3D"_blank">+1 (415) 254-454=
5</a> mobile<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20653-8348" value=3D"+14086538348" target=3D"_blank">+1 (408) 653-834=
8</a> desk<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20884-2400" value=3D"+14088842400" target=3D"_blank">+1 (408) 884-240=
0</a> fax<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:gray">This e-mail may contain confi=
dential and privileged material for the sole use of the intended recipient(=
s).=C2=A0 Any review or distribution by others is strictly prohibited.
 If you are not the intended recipient, please contact the sender and delet=
e all copies.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">mailto:rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"font-size:10.0pt;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">So it can&#39;t be all bad.....</span>&qu=
ot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u></u>
<u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"color:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On Oct 16, 2014, at 2=
:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA <u></u>
<u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-left:0in;line-height:14.4pt;vertical=
-align:baseline">
<u></u><span style=3D"font-size:10.0pt;font-family:Symbol;color:#333333"><s=
pan>=C2=B7<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:8.5pt;font-family:inhe=
rit;color:#333333"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
</div>
</div>
<pre><span style=3D"color:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"color:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>

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

--047d7bfd08baf1b1590506184886--


From nobody Thu Oct 23 08:20:40 2014
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Date: Thu, 23 Oct 2014 17:20:43 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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On 23/10/2014 3:22, Timothy B. Terriberry wrote:
> Sergio Garcia Murillo wrote:
>> BTW, please, please, please, let's use the sssrc-group:FEC and send the
>> FEC in its own ssrc stream, so we don't have to use RED anymore..
>
> That may be fine for video, but the added header overhead for audio is 
> substantial (and there are a number of other problems that RED solves 
> there, as discussed at the last meeting).
>
Could you elaborate please?

Best regards
Sergio


From nobody Thu Oct 23 08:51:16 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 23 Oct 2014 17:50:50 +0200
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Subject: Re: [rtcweb] Plan for MTI video codec?
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2014-10-23 12:15 GMT+02:00 Cavigioli, Chris <chris.cavigioli@intel.com>:
> The GSMA white paper has been published.  Download a copy here:
> http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/

Great. And now... can we go on with WebRTC and ignore anything 3GPP/GSMA sa=
ys?


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


From nobody Thu Oct 23 08:56:55 2014
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On Thu, Oct 23, 2014 at 3:15 AM, Cavigioli, Chris <chris.cavigioli@intel.co=
m
> wrote:

>  The GSMA white paper has been published.  Download a copy here:
> http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/
>

=E2=80=8BNote that this document claims to be confidential.  It's not clear=
 to me
to whom, but IETF working groups are =E2=80=8Bnot limited in membership, so=
 sharing
it with a working group is similar to making a public statement of its
availability.  Are you sure that this is the intent of the GSMA?


>
> Note: Transcoding can be avoided by negotiating optional codecs and
> finding a matching pair.  The GSMA white paper doesn=E2=80=99t focus on t=
hat case,
> but instead specifically highlights the scenario where a pure WebRTC
> endpoint is communicating with a pure 3GPP LTE VoLTE/ViLTE IMS endpoint,
> assuming each endpoint ONLY has access to the MTI codecs in its respectiv=
e
> domain.
>

=E2=80=8BWhile WebRTC has re-affirmed multiple times that a mandatory-to-im=
plement
codec is an important strategy for avoiding interoperability failure, the
negotiation of mutually acceptable codecs is far more fundamental.  You
don't get a lot of list traffic on it because it is a given.

For clients which expect to interoperate with telephony systems, the
inclusion of other codecs that assist with that interoperation would be a
far more natural strategy than relying on transcoding.  There's been
discussion of which those might be already, and we can have more.

But please realize that there is a broad variety of client use cases here
and that not all of them are in the realm of telephony.  For those, the
handheld device is not really an "IMS endpoint", it is a mobile
platform--and the choice of what to include there is not limited to the
mandatory to implement IMS codecs in any way of which I am aware.

=E2=80=8Bregards,

Ted Hardie
/wearing no hat at the moment.=E2=80=8B





> In that case, the network is responsible to transcode between (Opus or
> G.711, <WebRTC MTI video codec TBD>) and (AMR or AMR-WB, H.264).
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Cavigioli,
> Chris
> *Sent:* Sunday, October 19, 2014 6:20 PM
> *To:* Harald Alvestrand; Bernard Aboba
>
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
> =E2=80=9CWe made the mistake of having an MTI discussion previously with =
not
> enough details on that subject=E2=80=A6=E2=80=9D
>
>
>
> We didn=E2=80=99t have quantitative data earlier for a data-driven decisi=
on.
> Future Web =E2=80=93 LTE interop is important.  A group of us in GSMA hav=
e been
> looking at WebRTC interop with LTE (where 3GPP / GSMA defined IMS-based
> voice/video/messaging), specifically looking at user experience impact of
> latency introduced by added in-network transcoding.  GSMA has approved th=
e
> whitepaper and it is being prepared for public distribution.
>
>
>
> WebRTC =E2=80=93 LTE transcoding is now MANDATORY because of =E2=80=9CWeb=
RTC Audio Codec
> and Processing Requirements=E2=80=9D.
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05, where MTI audio
> codecs in WebRTC and in LTE are dissimilar.  Thus, REGARDLESS of the vide=
o
> codecs, there is already a significant degradation imposed on Web =E2=80=
=93 LTE
> links if only MTI codecs are used.  The only systems to avoid this are
> those which implement OPTIONAL audio codecs to be transcoder-free with
> LTE.  The same can apply for video codecs.
>
>
>
> The GSMA whitepaper refers to:
>
> 1.     Quantitative voice-over-LTE (VoLTE) delay limits were agreed-to by
> 3GPP SA4 in August 2014.  To paraphrase, performance objectives for maxim=
um
> VoLTE device delays in error and jitter free conditions and AMR speech
> codec operation, the sum of the UE delays in sending and receiving
> directions (TS + TR) should be =E2=89=A4 150ms. If this performance objec=
tive
> cannot be met, the sum of the UE delays in sending and receiving directio=
ns
> (TS + TR) shall in any case be =E2=89=A4 190ms.
>
> 2.     OPUS =E2=80=93 AMR transcoding adds an additional 211.5 ms
> implementation-agnostic (TS + TR) delay, including 40 ms for jitter
> buffer and 40 ms for discontinuous reception (DRX).
>
> 3.     To put these numbers in perspective, it is in general desirable to
> minimize UE delays to ensure low enough end-to-end delays and hence a goo=
d
> conversational experience.  Increasing delay causes people to double-talk
> making conversations increasingly frustrating.  Guidance to stay below 40=
0
> ms maximum one-way delay is found in ITU-T Recommendation G.114 and the
> computational E-model is in ITU-T Recommendation G.107.
>
>
>
> This is a complex topic with many network factors in play.  LTE operators
> in 3GPP seek UE delays (TS + TR) around 150 ms.  Adding OPUS =E2=80=93 AM=
R
> transcoding in the network imposes an additional 211.5 ms on top of that,
> just for audio.  Optionally using AMR in WebRTC, those 211.5 ms can be
> eliminated, delivering a superior end-user experience.
>
>
>
> CONCLUSION:  regardless of MTI codec choices in WebRTC, to provide a good
> WebRTC =E2=80=93 LTE interop experience, emerging WebRTC systems must acc=
ommodate
> MTI codecs already deployed in the LTE domain and in mobile operating
> systems, namely AMR/AMR-WB and H.264.
>
>
>
> POSITION:  To be clear, several Intel SoCs launched at MWC this year for
> smartphones and tablets support both VP8 and H.264 hardware encode and
> decode =E2=80=93 so Intel is codec-neutral in this MTI debate.  Additiona=
lly, Intel
> has high-performance solutions for transcoding.  However, we wish to info=
rm
> the industry, with quantitative data, about impact to end-user experience
> of mandating such transcoding so IETF can make a data-driven decision on
> video codecs as well as reflect on the impact of already having mandated
> transcoding on the audio side.
>
>
>
> Chris Cavigioli
>
> Intel Corp, System Architecture and Planning, Video/Multimedia, Mobile
> and Communications Group (MCG)
>
> +1 (415) 254-4545 mobile
>
> +1 (408) 653-8348 desk
>
> +1 (408) 884-2400 fax
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient(s).  Any review or distribution by others i=
s
> strictly prohibited. If you are not the intended recipient, please contac=
t
> the sender and delete all copies.
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org <rtcweb-bounces@ietf.org>]=
 *On
> Behalf Of *Bernard Aboba
> *Sent:* Sunday, October 19, 2014 2:29 PM
> *To:* Harald Alvestrand
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Plan for MTI video codec?
>
>
>
> Harald said:
>
>
>
> "Bernard,
>
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad....."
>
>
>
> [BA]  I agree that the lack of interoperability is largely "codec
> independent":   that is, implementations based on different mechanisms fo=
r
> congestion control, repair, etc. will exhibit interoperability problems,
> regardless of the codec chosen.   The real test for RTCWEB will be to mov=
e
> beyond "interoperability of implementations sharing source code"  to
> "interoperability of independent implementations, based on open standards=
".
>
>
>
>
> On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/19/2014 05:43 PM, Bernard Aboba wrote:
>
>  "And its one of the issues holding up wider adoption of the technology"
>
>
>
> [BA] Specifying an MTI encoder/decoder is not sufficient for
> interoperability.  It is also necessary to specify the transport in enoug=
h
> detail to allow independent implementations that interoperate well enough
> to be usable in a wide variety of scenarios, including wireless networks
> where loss is commonly experienced.
>
>
> Bernard,
>
> I think this is, to a large degree, codec independent.
>
> We have demonstrated interoperability on VP8 between Firefox and Chrome,
> and usage of various mechanisms for congestion control and repair of pack=
et
> loss being applied in live services.
>
> So it can't be all bad.....
>
>
>
>
>
> We made the mistake of having an MTI discussion previously with not enoug=
h
> details on that subject, particularly with respect to H.264.
> draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.
>
>
>
> So if we are to have the discussion again, it should occur in the context
> of detailed specifications and interoperability reports.
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg <jdrosen@jdrosen.net>
> wrote:
>
> I'm in favor of taking another run at this.
>
>
>
> The working group has repeatedly said that an MTI codec is something we
> need to produce. And its one of the issues holding up wider adoption of t=
he
> technology (not the only one for sure).
>
>
>
> And things have changed since the last meeting, a year ago now (November
> in Vancouver). Cisco's open264 plugin shipped and now just recently is
> integrated into Firefox. iOS8 shipped with APIs for H264. There are other
> things worth noting. Will this change the minds of everyone? Surely not.
> Will it sway enough for us to achieve rough consensus? Maybe. IMHO - wort=
h
> finding out.
>
>
>
> On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLARD <
> agouaillard@gmail.com> wrote:
>
> +1 to not having MTI codec discussion unless some progress has been made
> on VP8 at MPEG. Any news on that? I'm sharing harald's  feeling that ther=
e
> was no change on the members position.
>
>
>
> On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 10/17/2014 12:02 AM, Bernard Aboba wrote:
>
> One thing we could do instead of wasting time on MTI is to actually make
> progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-video, so we could
> actually interoperate regardless of the codec.
>
>
> The big argument for an MTI is actually the one stated in -overview: It
> guards against interoperability failure.
>
> I would like to have an ecosystem where one can field a box, connect it t=
o
> everything else, and run well for *some* level of "well" - and I would
> prefer that ecosystem to be one where it's possible to field the box
> without making prior arrangements with anyone about licenses.
>
> This argument hasn't changed one whit since last time we discussed it. An=
d
> I don't see much movement on the specifics of the proposals either.
>
> We'll have to interoperate well with the codecs we field. So using some
> time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn=
't
> finished either. There's one sentence that needs to be removed.)
>
> I wouldn't say I'd be happy to not discuss this in Honolulu. But I'd
> prefer to re-discuss based on the knowledge that some significant players
> have actually changed their minds.
>
> At the moment, I don't see signs that any of the poll respondents have
> said "I have changed my mind".
>
> Harald
>
>
>
>
>
> On Oct 16, 2014, at 2:28 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
> On 16 October 2014 14:17, Matthew Kaufman <matthew@matthew.at> wrote:
> And that's because something substantive has changed, or simply because
> wasting the WG time on this again is more entertaining than actually
> finishing a specification that can be independently implemented by all
> browser vendors? (A specification that we are nowhere near having, as far
> as
> I can tell)
>
> Personally, I've found the reprieve from this fight refreshing.  And
> it would appear that we've made some real progress as a result.  I'd
> suggest that if we don't have new information, we continue to spend
> our time productively.  If we can't find topics to occupy our meeting
> agenda time, then maybe we can free an agenda slot.
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Alex. Gouaillard, PhD, PhD, MBA
>
>
> -------------------------------------------------------------------------=
-----------
>
> CTO - Temasys Communications, S'pore / Mountain View
>
> President - CoSMo Software, Cambridge, MA
>
>
> -------------------------------------------------------------------------=
-----------
>
> sg.linkedin.com/agouaillard
>
> =C2=B7
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
> --
>
> Jonathan Rosenberg, Ph.D.
> jdrosen@jdrosen.net
> 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
>
>
>
> --
>
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> 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
>
>

--bcaec5196bddf291130506191a6f
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Oct 23, 2014 at 3:15 AM=
, Cavigioli, Chris <span dir=3D"ltr">&lt;<a href=3D"mailto:chris.cavigioli@=
intel.com" target=3D"_blank">chris.cavigioli@intel.com</a>&gt;</span> wrote=
:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">The GSMA white paper has be=
en published.=C2=A0 Download a copy here:=C2=A0
<a href=3D"http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/" target=
=3D"_blank">http://www.gsma.com/newsroom/webrtc-codecs-draft-v1-3/</a></spa=
n></p></div></div></blockquote><div><br><div class=3D"gmail_default" style=
=3D"font-family:tahoma,sans-serif;font-size:small;display:inline">=E2=80=8B=
Note that this document claims to be confidential.=C2=A0 It&#39;s not clear=
 to me to whom, but IETF working groups are =E2=80=8Bnot limited in members=
hip, so sharing it with a working group is similar to making a public state=
ment of its availability.=C2=A0 Are you sure that this is the intent of the=
 GSMA?<br><br></div></div><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link=3D"blue"=
 vlink=3D"purple" lang=3D"EN-US"><div><p class=3D"MsoNormal"><span style=3D=
"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;colo=
r:#1f497d">
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">Note: Transcoding can be av=
oided by negotiating optional codecs and finding a matching pair.=C2=A0 The=
 GSMA white paper doesn=E2=80=99t focus on that case, but instead specifica=
lly
 highlights the scenario where a pure WebRTC endpoint is communicating with=
 a pure 3GPP LTE VoLTE/ViLTE IMS endpoint, assuming each endpoint ONLY has =
access to the MTI codecs in its respective domain.=C2=A0 </span></p></div><=
/div></blockquote><div><br><div class=3D"gmail_default" style=3D"font-famil=
y:tahoma,sans-serif;font-size:small">=E2=80=8BWhile WebRTC has re-affirmed =
multiple times that a mandatory-to-implement codec is an important strategy=
 for avoiding interoperability failure, the negotiation of mutually accepta=
ble codecs is far more fundamental.=C2=A0 You don&#39;t get a lot of list t=
raffic on it because it is a given.=C2=A0 <br><br></div><div class=3D"gmail=
_default" style=3D"font-family:tahoma,sans-serif;font-size:small">For clien=
ts which expect to interoperate with telephony systems, the inclusion of ot=
her codecs that assist with that interoperation would be a far more natural=
 strategy than relying on transcoding.=C2=A0 There&#39;s been discussion of=
 which those might be already, and we can have more.<br><br>But please real=
ize that there is a broad variety of client use cases here and that not all=
 of them are in the realm of telephony.=C2=A0 For those, the handheld devic=
e is not really an &quot;IMS endpoint&quot;, it is a mobile platform--and t=
he choice of what to include there is not limited to the mandatory to imple=
ment IMS codecs in any way of which I am aware.<br></div><br><div class=3D"=
gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=
=80=8Bregards,<br><br>Ted Hardie<br>/wearing no hat at the moment.=E2=80=8B=
</div><br><br><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link=3D"b=
lue" vlink=3D"purple" lang=3D"EN-US"><div><p class=3D"MsoNormal"><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;=
color:#1f497d">In that case, the network is responsible to transcode betwee=
n
 (Opus or G.711, &lt;WebRTC MTI video codec TBD&gt;) and (AMR or AMR-WB, H.=
264).=C2=A0 <u></u>
<u></u></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"><u></u>=C2=A0<u></u></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 [=
mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Cavigioli, Chris<br>
<b>Sent:</b> Sunday, October 19, 2014 6:20 PM<br>
<b>To:</b> Harald Alvestrand; Bernard Aboba</span></p><div><div class=3D"h5=
"><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></div><=
/div><p></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:#1f497d">Harald said:=C2=A0
<u></u><u></u></span></p>
<p class=3D"MsoNormal">=E2=80=9CWe made the mistake of having an MTI discus=
sion previously with not enough details on that subject=E2=80=A6=E2=80=9D<s=
pan style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-seri=
f&quot;;color:#1f497d"><u></u><u></u></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"><u></u>=C2=A0<u></u></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">We didn=E2=80=99t have quan=
titative data earlier for a data-driven decision.=C2=A0 Future Web =E2=80=
=93 LTE interop is important.=C2=A0 A group of us in GSMA have been looking=
 at WebRTC
 interop with LTE (where 3GPP / GSMA defined IMS-based voice/video/messagin=
g), specifically looking at user experience impact of latency introduced by=
 added in-network transcoding.=C2=A0 GSMA has approved the whitepaper and i=
t is being prepared for public distribution.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">WebRTC =E2=80=93 LTE transc=
oding is now MANDATORY because of =E2=80=9CWebRTC Audio Codec and Processin=
g Requirements=E2=80=9D.
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-05</a>, where M=
TI audio codecs in WebRTC and in LTE are dissimilar.=C2=A0 Thus, REGARDLESS=
 of the video codecs, there is already a significant degradation
 imposed on Web =E2=80=93 LTE links if only MTI codecs are used.=C2=A0 The =
only systems to avoid this are those which implement OPTIONAL audio codecs =
to be transcoder-free with LTE.=C2=A0 The same can apply for video codecs.<=
u></u><u></u></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"><u></u>=C2=A0<u></u></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">The GSMA whitepaper refers =
to:<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">Quantitative voice-ove=
r-LTE (VoLTE) delay limits were agreed-to by 3GPP SA4 in August 2014.=C2=A0=
 To paraphrase, performance objectives for maximum VoLTE
 device delays in error and jitter free conditions and AMR speech codec ope=
ration, the sum of the UE delays in sending and receiving directions (T<sub=
>S</sub> + T<sub>R</sub>) should be =E2=89=A4 150ms. If this performance ob=
jective cannot be met, the sum of the UE
 delays in sending and receiving directions (T<sub>S</sub> + T<sub>R</sub>)=
 shall in any case be =E2=89=A4 190ms.<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">OPUS =E2=80=93 AMR tra=
nscoding adds an additional 211.5 ms implementation-agnostic (T<sub>S</sub>=
 + T<sub>R</sub>) delay, including 40 ms for jitter buffer
 and 40 ms for discontinuous reception (DRX).<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&qu=
ot;sans-serif&quot;;color:#1f497d"><span>3.<span style=3D"font:7.0pt &quot;=
Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Arial&quot;,&quot;sans-serif&quot;;color:#1f497d">To put these numbers i=
n perspective, it is in general desirable to minimize UE delays to ensure l=
ow enough end-to-end delays and hence a good conversational
 experience.=C2=A0 Increasing delay causes people to double-talk making con=
versations increasingly frustrating.=C2=A0 Guidance to stay below 400 ms ma=
ximum one-way delay is found in ITU-T Recommendation G.114 and the computat=
ional E-model is in ITU-T Recommendation G.107.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">This is a complex topic wit=
h many network factors in play.=C2=A0 LTE operators in 3GPP seek UE delays =
(T<sub>S</sub> + T<sub>R</sub>) around 150 ms.=C2=A0 Adding OPUS =E2=80=93
 AMR transcoding in the network imposes an additional 211.5 ms on top of th=
at, just for audio.=C2=A0 Optionally using AMR in WebRTC, those 211.5 ms ca=
n be eliminated, delivering a superior end-user experience.=C2=A0
<u></u><u></u></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"><u></u>=C2=A0<u></u></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">CONCLUSION: =C2=A0regardles=
s of MTI codec choices in WebRTC, to provide a good WebRTC =E2=80=93 LTE in=
terop experience, emerging WebRTC systems must accommodate MTI codecs
 already deployed in the LTE domain and in mobile operating systems, namely=
 AMR/AMR-WB and H.264.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">POSITION:=C2=A0 To be clear=
, several Intel SoCs launched at MWC this year for smartphones and tablets =
support both VP8 and H.264 hardware encode and decode =E2=80=93 so Intel
 is codec-neutral in this MTI debate.=C2=A0 Additionally, Intel has high-pe=
rformance solutions for transcoding.=C2=A0 However, we wish to inform the i=
ndustry, with quantitative data, about impact to end-user experience of man=
dating such transcoding so IETF can make a
 data-driven decision on video codecs as well as reflect on the impact of a=
lready having mandated transcoding on the audio side.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Lucida Console&quot=
;;color:#1f497d">Chris Cavigioli<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d">Intel Corp,
</span><span style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot=
;sans-serif&quot;;color:#0070c0">System Architecture and Planning</span><sp=
an style=3D"font-size:8.0pt;font-family:&quot;Verdana&quot;,&quot;sans-seri=
f&quot;;color:#1f497d">, Video/Multimedia, Mobile and Communications Group =
(MCG)<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
15%29%20254-4545" value=3D"+14152544545" target=3D"_blank">+1 (415) 254-454=
5</a> mobile<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20653-8348" value=3D"+14086538348" target=3D"_blank">+1 (408) 653-834=
8</a> desk<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:8.0pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:#1f497d"><a href=3D"tel:%2B1%20%284=
08%29%20884-2400" value=3D"+14088842400" target=3D"_blank">+1 (408) 884-240=
0</a> fax<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ver=
dana&quot;,&quot;sans-serif&quot;;color:gray">This e-mail may contain confi=
dential and privileged material for the sole use of the intended recipient(=
s).=C2=A0 Any review or distribution by others is strictly prohibited.
 If you are not the intended recipient, please contact the sender and delet=
e all copies.<u></u><u></u></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"><u></u>=C2=A0<u></u></span>=
</p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">mailto:rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, October 19, 2014 2:29 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Plan for MTI video codec?<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Harald said:=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;<span style=3D"font-size:10.0pt;font-family:&q=
uot;Arial&quot;,&quot;sans-serif&quot;">Bernard,</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">So it can&#39;t be all bad.....</span>&qu=
ot;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] =C2=A0I agree that the lack of interoperability=
 is largely &quot;codec independent&quot;: =C2=A0 that is, implementations =
based on different mechanisms for congestion control, repair, etc. will exh=
ibit interoperability problems, regardless of the codec
 chosen. =C2=A0 The real test for RTCWEB will be to move beyond &quot;inter=
operability of implementations sharing source code&quot; =C2=A0to &quot;int=
eroperability of independent implementations, based on open standards&quot;=
. =C2=A0=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 1:38 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On 10/19/2014 05:43 PM, Bernard Aboba wrote:<u></u><=
u></u></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">&quot;And its one of the issues holding up wider ado=
ption of the technology&quot;
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">[BA] Specifying an MTI encoder/decoder is not suffic=
ient for interoperability.=C2=A0 It is also necessary to specify the transp=
ort in enough detail to allow independent implementations that interoperate=
 well enough to be usable in a wide variety
 of scenarios, including wireless networks where loss is commonly experienc=
ed.=C2=A0 <u></u>
<u></u></p>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Bernard,<br>
<br>
I think this is, to a large degree, codec independent.<br>
<br>
We have demonstrated interoperability on VP8 between Firefox and Chrome, an=
d usage of various mechanisms for congestion control and repair of packet l=
oss being applied in live services.<br>
<br>
So it can&#39;t be all bad.....<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We made the mistake of having an MTI discussion prev=
iously with not enough details on that subject, particularly with respect t=
o H.264. draft-ietf-rtcweb-video sections 4.2 - 4.4 remain sketchy at best.=
 =C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">So if we are to have the discussion again, it should=
 occur in the context of detailed specifications and interoperability repor=
ts.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sun, Oct 19, 2014 at 8:14 AM, Jonathan Rosenberg =
&lt;<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrose=
n.net</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">I&#39;m in favor of taking another run at this. <u><=
/u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The working group has repeatedly said that an MTI co=
dec is something we need to produce. And its one of the issues holding up w=
ider adoption of the technology (not the only one for sure).<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">And things have changed since the last meeting, a ye=
ar ago now (November in Vancouver). Cisco&#39;s open264 plugin shipped and =
now just recently is integrated into Firefox. iOS8 shipped with APIs for H2=
64. There are other things worth noting.
 Will this change the minds of everyone? Surely not. Will it sway enough fo=
r us to achieve rough consensus? Maybe. IMHO - worth finding out.<u></u><u>=
</u></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Sat, Oct 18, 2014 at 5:08 PM, Alexandre GOUAILLAR=
D &lt;<a href=3D"mailto:agouaillard@gmail.com" target=3D"_blank">agouaillar=
d@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">+1 to not having MTI codec discussion unless some pr=
ogress has been made on VP8 at MPEG.=C2=A0Any news on that? I&#39;m sharing=
 harald&#39;s =C2=A0feeling that there was no change on the members positio=
n.=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Oct 17, 2014 at 9:22 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 10/17/2014 12:02 AM, Bernard Aboba wrote:<u></u><=
u></u></p>
<p class=3D"MsoNormal">One thing we could do instead of wasting time on MTI=
 is to actually make progress on Sections 4.2 - 4.4 of draft-IETF-RTCWEB-vi=
deo, so we could actually interoperate regardless of the codec.<u></u><u></=
u></p>
<p class=3D"MsoNormal"><br>
The big argument for an MTI is actually the one stated in -overview: It gua=
rds against interoperability failure.<br>
<br>
I would like to have an ecosystem where one can field a box, connect it to =
everything else, and run well for *some* level of &quot;well&quot; - and I =
would prefer that ecosystem to be one where it&#39;s possible to field the =
box without making prior arrangements with anyone
 about licenses.<br>
<br>
This argument hasn&#39;t changed one whit since last time we discussed it. =
And I don&#39;t see much movement on the specifics of the proposals either.=
<br>
<br>
We&#39;ll have to interoperate well with the codecs we field. So using some=
 time to discuss draft-ietf-rtcweb-video seems to make sense. (And 4.1 isn&=
#39;t finished either. There&#39;s one sentence that needs to be removed.)<=
br>
<br>
I wouldn&#39;t say I&#39;d be happy to not discuss this in Honolulu. But I&=
#39;d prefer to re-discuss based on the knowledge that some significant pla=
yers have actually changed their minds.<br>
<br>
At the moment, I don&#39;t see signs that any of the poll respondents have =
said &quot;I have changed my mind&quot;.<span style=3D"color:#888888"><br>
<br>
Harald</span> <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On Oct 16, 2014, at 2=
:28 PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">On 16 October 2014 14:17, Matthew Kaufman &lt;<a hre=
f=3D"mailto:matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt=
; wrote:<br>
And that&#39;s because something substantive has changed, or simply because=
<br>
wasting the WG time on this again is more entertaining than actually<br>
finishing a specification that can be independently implemented by all<br>
browser vendors? (A specification that we are nowhere near having, as far a=
s<br>
I can tell)<u></u><u></u></p>
<p class=3D"MsoNormal">Personally, I&#39;ve found the reprieve from this fi=
ght refreshing.=C2=A0 And<br>
it would appear that we&#39;ve made some real progress as a result.=C2=A0 I=
&#39;d<br>
suggest that if we don&#39;t have new information, we continue to spend<br>
our time productively.=C2=A0 If we can&#39;t find topics to occupy our meet=
ing<br>
agenda time, then maybe we can free an agenda slot.<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><u></u><u></u></p>
<p class=3D"MsoNormal">_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">-- <u></u><u></u></spa=
n></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Alex. Gouaillard, PhD,=
 PhD, MBA <u></u>
<u></u></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">CTO - Temasys Communic=
ations, S&#39;pore / Mountain View<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">President - CoSMo Soft=
ware, Cambridge, MA<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">----------------------=
--------------------------------------------------------------<u></u><u></u=
></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><a href=3D"http://sg.l=
inkedin.com/agouaillard" target=3D"_blank">sg.linkedin.com/agouaillard</a><=
u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-left:0in;line-height:14.4pt;vertical=
-align:baseline">
<u></u><span style=3D"font-size:10.0pt;font-family:Symbol;color:#333333"><s=
pan>=C2=B7<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:8.5pt;font-family:inhe=
rit;color:#333333"><u></u>=C2=A0<u></u></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">Jonathan Rosenberg, Ph.D.<br>
<a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.ne=
t</a><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a><u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>rtcweb mailing list<u></u><u></u></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><u></u><u></u></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
</div>
</div>
<pre><span style=3D"color:#888888">-- <u></u><u></u></span></pre>
<pre><span style=3D"color:#888888">Surveillance is pervasive. Go Dark.<u></=
u><u></u></span></pre>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>

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

--bcaec5196bddf291130506191a6f--


From nobody Thu Oct 23 09:47:45 2014
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From: "David Benham (dbenham)" <dbenham@cisco.com>
To: Stephan Wenger <stewe@stewe.org>, Mohammed Raad <mohammedsraad@raadtech.com>
Thread-Topic: [rtcweb] VCB/VP8 at ISO (was: Plan for MTI video codec?
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Subject: Re: [rtcweb] VCB/VP8 at ISO (was: Plan for MTI video codec?
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From nobody Thu Oct 23 10:03:38 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] SDP and ssrc-group,
Thread-Index: AQHP7ViUdeZEO6Q8ck+bZW3ftl2D4pw85NsDgAD8OIA=
Date: Thu, 23 Oct 2014 17:03:31 +0000
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Subject: Re: [rtcweb] SDP and ssrc-group,
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> Sergio Garcia Murillo wrote:
> > BTW, please, please, please, let's use the sssrc-group:FEC and send the
> > FEC in its own ssrc stream, so we don't have to use RED anymore..
>=20
> That may be fine for video, but the added header overhead for audio is
> substantial (and there are a number of other problems that RED solves
> there, as discussed at the last meeting).

<Raju>
I am not sure if red+ulpfec is used for audio codecs in webrtc context (in =
general it can be used for alternate encodings even for audio); OPUS does h=
ave a built-in FEC so no need for red+ulpfec.

For video, the RTP header overhead added by ssrc multiplexing is considerab=
le (not significant, but can't be ignored) especially for the non-key frame=
s and in mobile environments.
Also, there is an additional problem with ssrc multiplexing for FEC.
Use of red+ulpfec makes it easy for an SDP answerer to reject it by excludi=
ng red+ulpfec payloads if it does not support FEC. But use of a=3Dssrc-grou=
p:FEC makes it hard to reject the FEC part as the semantics are not support=
ed by http://tools.ietf.org/html/rfc5576#section-8
This issue is being discussed at http://www.ietf.org/mail-archive/web/rtcwe=
b/current/msg13249.html

While Chrome supports red+ulpfec Firefox does not (yet!). If we use a=3Dssr=
c-group:FEC then we will have an interop issue if Firefox rejects the entir=
e video offer to strictly comply to RFC 5576 section 8.

I wish the draft-ietf-rtcweb-rtp-usage (draft-ietf-rtcweb-audio or draft-ie=
tf-rtcweb-video) says one way or other about how to convey FEC encoding (re=
dundant encoding or ssrc multiplexing)=20

</Raju>

BR
raju


From nobody Thu Oct 23 10:10:57 2014
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From: Stephan Wenger <stewe@stewe.org>
To: Stephan Wenger <stewe@stewe.org>, Watson Ladd <watsonbladd@gmail.com>, Alexandre GOUAILLARD <agouaillard@gmail.com>
Thread-Topic: [rtcweb] Plan for MTI video codec?
Thread-Index: AQHOrixjgMwfvYrSbUqrcWE5foJ0c5wzqjdjgAAvKoCAAddnAIAAAwiAgAAJmQCAAQDrgIACFIiAgAEvmwCAAAgMAIAACFqAgAAAiICAACAFgIAEXFwAgAFnCIA=
Date: Thu, 23 Oct 2014 17:10:41 +0000
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Subject: Re: [rtcweb] Plan for MTI video codec?
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From nobody Thu Oct 23 10:12:13 2014
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Makaraju, Maridi Raju (Raju) wrote:
> OPUS does have a built-in FEC so no need for red+ulpfec.

As discussed previously on this list [1] the built-in FEC only covers 
the SILK layer. Existing codec-agnostic mechanisms were perfectly 
adequate for protecting CELT-mode packets, so we didn't invent something 
new.

[1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html


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Subject: Re: [rtcweb] SDP and ssrc-group,
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Sergio Garcia Murillo wrote:
>> substantial (and there are a number of other problems that RED solves
>> there, as discussed at the last meeting).
>>
> Could you elaborate please?

See slide 4 of 
<http://www.ietf.org/proceedings/90/slides/slides-90-rtcweb-7.pdf>. I 
was particularly thinking of the issues with bundle, which do not apply 
to RFC 2198 RED (which I pointed out at the mic).


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] SDP and ssrc-group,
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>=20
> Makaraju, Maridi Raju (Raju) wrote:
> > OPUS does have a built-in FEC so no need for red+ulpfec.
>=20
> As discussed previously on this list [1] the built-in FEC only covers
> the SILK layer. Existing codec-agnostic mechanisms were perfectly
> adequate for protecting CELT-mode packets, so we didn't invent something
> new.
>=20
> [1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html
[Raju] Thanks and appreciate the reference.


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--001a11c17e5e26e40e05061c1fd8
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Generally, I think there is a fair amount of confusion about how these
various types of TURN servers are going to work. The RETURN draft takes a
first stab at defining the interactions here, and ideally it can be adopted
as a WG draft and referred to by -transports.

On Thu, Oct 23, 2014 at 5:26 AM, Olle E. Johansson <oej@edvina.net> wrote:

>
> On 22 Oct 2014, at 16:57, Benjamin Schwartz <bemasc@webrtc.org> wrote:
>
> FYI, the purpose of the RETURN draft (
> https://tools.ietf.org/html/draft-schwartz-rtcweb-return-03) is to try to
> specify the precise interaction and precedence when both the browser and
> the application provide TURN servers.  (It's not as simple as "one or the
> other".)
>
> I haven't considered the issues related to browser-provided STUN servers,
> though.
>
> Cool! I'll take a look at that. Maybe Haralds draft need to refer to that.
>
> /O
>
>
> On Wed, Oct 22, 2014 at 10:51 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> The advantage of an application configuration is that it can be
>> dynamically provided/updated, based on location based etc - assuming the
>> webpage provider has knowledge about STUN/TURN servers, of course.
>>
>> Regards,
>>
>> Christer
>>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Olle E.
>> Johansson
>> Sent: 22 October 2014 17:44
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt
>>
>> Section 3.4
>>
>> " WebRTC browsers MUST support configuration of STUN and TURN servers,
>>    both from browser configuration and from an application."
>>
>>
>> Should we mention which takes precedence? If configured in the browser,
>> start with that server.
>> If not use the one provided by the application.
>>
>> Maybe we should clarify that turn DNS discovery MUST be used to provide
>> failover and load balancing.
>>
>> /O
>> _______________________________________________
>> 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
>
>

--001a11c17e5e26e40e05061c1fd8
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<div dir=3D"ltr">Generally, I think there is a fair amount of confusion abo=
ut how these various types of TURN servers are going to work. The RETURN dr=
aft takes a first stab at defining the interactions here, and ideally it ca=
n be adopted as a WG draft and referred to by -transports.</div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Oct 23, 2014 at 5:2=
6 AM, Olle E. Johansson <span dir=3D"ltr">&lt;<a href=3D"mailto:oej@edvina.=
net" target=3D"_blank">oej@edvina.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"><div style=3D"word-wrap:break-word"><br><div><span class=
=3D""><div>On 22 Oct 2014, at 16:57, Benjamin Schwartz &lt;<a href=3D"mailt=
o:bemasc@webrtc.org" target=3D"_blank">bemasc@webrtc.org</a>&gt; wrote:</di=
v><br><blockquote type=3D"cite"><div dir=3D"ltr">FYI, the purpose of the RE=
TURN draft (<a href=3D"https://tools.ietf.org/html/draft-schwartz-rtcweb-re=
turn-03" target=3D"_blank">https://tools.ietf.org/html/draft-schwartz-rtcwe=
b-return-03</a>) is to try to specify the precise interaction and precedenc=
e when both the browser and the application provide TURN servers. =C2=A0(It=
&#39;s not as simple as &quot;one or the other&quot;.)<div><br></div><div>I=
 haven&#39;t considered the issues related to browser-provided STUN servers=
, though.</div></div></blockquote></span>Cool! I&#39;ll take a look at that=
. Maybe Haralds draft need to refer to that.</div><span class=3D"HOEnZb"><f=
ont color=3D"#888888"><div><br></div></font></span><div><span class=3D"HOEn=
Zb"><font color=3D"#888888">/O</font></span><div><div class=3D"h5"><br><blo=
ckquote type=3D"cite"><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote">On Wed, Oct 22, 2014 at 10:51 AM, Christer Holmberg <span dir=3D"ltr">=
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">Hi,<br>
<br>
The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div><div><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_=
blank">rtcweb-bounces@ietf.org</a>] On Behalf Of Olle E. Johansson<br>
Sent: 22 October 2014 17:44<br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt<br>
<br>
Section 3.4<br>
<br>
&quot; WebRTC browsers MUST support configuration of STUN and TURN servers,=
<br>
=C2=A0 =C2=A0both from browser configuration and from an application.&quot;=
<br>
<br>
<br>
Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.<br>
If not use the one provided by the application.<br>
<br>
Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.<br>
<br>
/O<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
</blockquote></div></div></div><br></div><br>______________________________=
_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c17e5e26e40e05061c1fd8--


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Subject: Re: [rtcweb] Plan for MTI video codec?
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On 10/20/2014 3:09 AM, Cavigioli, Chris wrote:
>
> No, not 2 discussions ... but 2 MTI codecs.
>
> Given:
>
> -Industry is polarized c. 40% VP8 vs. c. 40% H.264
>
> -Unlikely this will change
>
> -Every single mobile device (100%) of the billions out there running a 
> mobile OS (iOS, Android, Windows, BB OS, etc) will already support 
> H.264, AMR-xx since those are MTI for 3GPP and for the OS
>
> Then:
>
> -Why not support both:  VP8, Opus = web; H.264, AMR/AMR-WB = LTE
>
> -Every single mobile device and OS already deals with H.264+AMR 
> licensing for many years.
>
> -Being royalty free, incrementally adding VP8 and Opus shouldn't be a 
> major additional impact
>

I should note that many H.264 HW implementations (or the software 
driving them) in mobile devices are NOT usable for WebRTC (or for video 
communication in general).  The bugs I've seen..... <shudder>.  You can 
certify specific sets of hardware or base chipset code, but until they 
actually get tested *well* in interactive, error-recovery, variable 
bandwidth use you have no clue if they'll work.  (And without such 
testing, generally the answer would be 'no').  Can you say "IDR to 
change bitrate"?  Learn.... OMX has *many* options, most of which are 
not supported on a given implementation, which makes generic use 
'interesting' when you go past the well-traveled (and still rocky) 
decode path.

So if Mozilla distributes Firefox to Android devices, what are we to 
do?  We can't simply assume the encoder/decoder will work for WebRTC at 
this point.  Even getting streaming video decode to leverage HW decoders 
was a massive pain with whitelists, blacklists, evil device-specific 
hacks, etc.  It has gotten better over time - but not great.

I'll note that for WebRTC we currently support OpenH264 on desktop, and 
HW H.264 on specific FxOS devices/chipsets/versions (the default for 
FxOS WebRTC OMX H.264 support is false).  As of today we support only 
VP8 on Android.

And I'll echo Bernard's comments about AMR and licensing/open APIs.

<gripe> When did people forget how to edit out irrelevant repetition of 
old messages?  Ah, yes, when everyone started top-replying.... </gripe>  
Makes searching archives much less fun.  :-/


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 10/20/2014 3:09 AM, Cavigioli, Chris
      wrote:<br>
    </div>
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">No,
            not 2 discussions &#8230; but 2 MTI codecs.&nbsp;
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Given:<o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Industry
            is polarized c. 40% VP8 vs. c. 40% H.264
            <o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Unlikely
            this will change<o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Every
            single mobile device (100%) of the billions out there
            running a mobile OS (iOS, Android, Windows, BB OS, etc) will
            already support H.264, AMR-xx since those are MTI for 3GPP
            and for the OS<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Then:<o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Why
            not support both:&nbsp; VP8, Opus = web; H.264, AMR/AMR-WB = LTE<o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Every
            single mobile device and OS already deals with H.264+AMR
            licensing for many years.<o:p></o:p></span></p>
        <p class="MsoListParagraph"
          style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
              style="mso-list:Ignore">-<span style="font:7.0pt
                &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">Being
            royalty free, incrementally adding VP8 and Opus shouldn&#8217;t be
            a major additional impact
            <o:p></o:p></span></p>
      </div>
    </blockquote>
    <br>
    I should note that many H.264 HW implementations (or the software
    driving them) in mobile devices are NOT usable for WebRTC (or for
    video communication in general).&nbsp; The bugs I've seen.....
    &lt;shudder&gt;.&nbsp; You can certify specific sets of hardware or base
    chipset code, but until they actually get tested *well* in
    interactive, error-recovery, variable bandwidth use you have no clue
    if they'll work.&nbsp; (And without such testing, generally the answer
    would be 'no').&nbsp; Can you say "IDR to change bitrate"?&nbsp; Learn....&nbsp;
    OMX has *many* options, most of which are not supported on a given
    implementation, which makes generic use 'interesting' when you go
    past the well-traveled (and still rocky) decode path.<br>
    <br>
    So if Mozilla distributes Firefox to Android devices, what are we to
    do?&nbsp; We can't simply assume the encoder/decoder will work for WebRTC
    at this point.&nbsp; Even getting streaming video decode to leverage HW
    decoders was a massive pain with whitelists, blacklists, evil
    device-specific hacks, etc.&nbsp; It has gotten better over time - but
    not great.<br>
    <br>
    I'll note that for WebRTC we currently support OpenH264 on desktop,
    and HW H.264 on specific FxOS devices/chipsets/versions (the default
    for FxOS WebRTC OMX H.264 support is false).&nbsp; As of today we support
    only VP8 on Android.<br>
    <br>
    And I'll echo Bernard's comments about AMR and licensing/open APIs.<br>
    <br>
    &lt;gripe&gt; When did people forget how to edit out irrelevant
    repetition of old messages?&nbsp; Ah, yes, when everyone started
    top-replying.... &lt;/gripe&gt;&nbsp; Makes searching archives much less
    fun.&nbsp; :-/<br>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Audio Codec and Processing Requirements
        Authors         : Jean-Marc Valin
                          Cary Bran
	Filename        : draft-ietf-rtcweb-audio-07.txt
	Pages           : 6
	Date            : 2014-10-24

Abstract:
   This document outlines the audio codec and processing requirements
   for WebRTC client application and endpoint devices.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-audio-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-07


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

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


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To: Stephan Wenger <stewe@stewe.org>, Watson Ladd <watsonbladd@gmail.com>, Alexandre GOUAILLARD <agouaillard@gmail.com>
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Subject: Re: [rtcweb] Plan for MTI video codec?
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On 10/22/14 14:45, Stephan Wenger wrote:
> I have to make one correction in the light of information that has
> surfaced at the MPEG meeting currently ongoing in Strasbourg.  (Iâ€™m not at
> that meeting this time, but a colleague is and she briefed me.)
> Nokia has made MPEG and ISO/IEC officially aware that they are not willing
> to license essential patents under RAND terms.  For those with MPEG
> document access, please see M34917.  The official declaration is dated
> 9/19/2014, and is not yet available from the respective databases, as ISO
> is apparently changing its recordation infrastructure.

Thanks for the update.

That's actually even more interesting than it first appears to be. The 
official rationale previously offered by Nokia was that the refusal to 
license patents [0] for VP8 was that VP8 had not been through an 
appropriate standards process [1]. The fact that Nokia is now using 
those same patents to *block* VP8-based work from going through an 
appropriate standards process pretty clearly exposes that claim as 
false. This isn't aimed at ensuring "open and collaborative efforts for 
standardization": this is aimed at suppressing technology.

To be clear, my position remains that either VP8 or H.264 would be a 
suitable MTI, and (although not my preference) I would accept a 
situation in which both are required. Which is to say that I don't 
really have much at stake in the MTI conversation (as long as we do 
eventually do something to guarantee interop). But it does seem that 
Nokia isn't being forthcoming about its motives here, and I find that 
pretty distasteful. I would have preferred silence over lies.


> My understanding of the joint ITU/ISO/IEC patent policy is that no
> standard can be issued that has a type 3 declaration against it.  To the
> best of my knowledge, ISO has no established procedure how to deal with
> type 3 (non-RAND) declarations and still keep the standard project going.
> Unlike, for example, W3C and its Patent Advisory Groups.
> The declaration does not list specific patents. To the best of my
> knowledge, such info is not required (only desired) for ISO and IEC
> standardization work--one of the few differences in patent policy
> guidelines between ITU and ISO/IEC.
> Therefore, I have to row back on my previous statement of likeliness of
> having an ISO number for VP8 anytime soon.  At this point, I just donâ€™t
> know whether, if ever, that will happen.


I find the ISO policy as stated to be curious -- if it doesn't require 
citing patents, and categorically refuses to issue specs with type 3 
declarations, I could effectively reduce the ISO output to zero by 
mechanically claiming a type 3 declaration on every specification under 
development.

/a

____
[0] For absolute clarity, I make no assertion one way or another about 
the applicability of such patents to VP8.

[1] As quoted in uncountably many publications: "Nokia believes that 
open and collaborative efforts for standardization are in the best 
interests of consumers, innovators and the industry as a whole. We are 
now witnessing one company attempting to force the adoption of its 
proprietary technology, which offers no advantages over existing, widely 
deployed standards such as H.264 and infringes Nokia's intellectual 
property. As a result, we have taken the unusual step of declaring to 
the Internet Engineering Task Force that we are not prepared to license 
any Nokia patents which may be needed to implement its RFC6386 
specification for VP8, or for derivative codecs."


From nobody Fri Oct 24 17:00:20 2014
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In section 3, draft states that: "WebRTC clients are REQUIRED to be able to
generate and consume the following events" followed by list of RFC4733
events. What does it mean by "consume"? Does it mean receive without
failure, while ignoring its contents or actually processing the events and
exposing them to the end point user in some way?

Supported RFC4733 event list does not match events listed in section 6.2.2
W3C TR (http://www.w3.org/TR/webrtc/#methods-4). W3C also lists support for
DTMF digits A through D.

Should the document state the duration and gap recommendations for RFC 4733
tones from section 6.2.2 of W3C document?

I did not know there is a consensus on must implement of CN.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr">In section 3, draft states that: &quot;<span style=3D"colo=
r:rgb(0,0,0);font-size:13px;line-height:1.2em">WebRTC clients are REQUIRED =
to be able to generate and consume the</span><span style=3D"color:rgb(0,0,0=
);font-size:13px;line-height:1.2em">=C2=A0following events&quot; followed b=
y list of=C2=A0</span><span style=3D"color:rgb(0,0,0);font-size:13px;line-h=
eight:1.2em">RFC4733 events. What does it mean by &quot;consume&quot;? Does=
 it mean receive without failure, while ignoring its contents or actually p=
rocessing the events and exposing them to the end point user in some way?=
=C2=A0</span><div><font color=3D"#000000"><span style=3D"line-height:15.600=
0003814697px"><br></span></font></div><div><font color=3D"#000000"><span st=
yle=3D"line-height:15.6000003814697px">Supported=C2=A0</span></font><span s=
tyle=3D"color:rgb(0,0,0);font-size:13px;line-height:15.6000003814697px">RFC=
4733</span><span style=3D"color:rgb(0,0,0);font-size:13px;line-height:15.60=
00003814697px">=C2=A0</span><span style=3D"line-height:15.6000003814697px;c=
olor:rgb(0,0,0)">event list does not match events listed in section 6.2.2 W=
3C TR (<a href=3D"http://www.w3.org/TR/webrtc/#methods-4">http://www.w3.org=
/TR/webrtc/#methods-4</a>). W3C also lists support for DTMF digits A throug=
h D.</span></div><div><font color=3D"#000000"><span style=3D"line-height:15=
.6000003814697px"><br></span></font></div><div><font color=3D"#000000"><spa=
n style=3D"line-height:15.6000003814697px">Should the document state the du=
ration and gap recommendations for RFC 4733 tones from section 6.2.2 of W3C=
 document?</span></font></div><div><font color=3D"#000000"><span style=3D"l=
ine-height:15.6000003814697px"><br></span></font></div><div><font color=3D"=
#000000"><span style=3D"line-height:15.6000003814697px">I did not know ther=
e is a consensus on must implement of CN.</span></font></div><div><font col=
or=3D"#000000"><span style=3D"line-height:15.6000003814697px"><br></span></=
font></div><div><font color=3D"#000000"><span style=3D"line-height:15.60000=
03814697px">Regards,</span></font><div><div class=3D"gmail_extra"><div>____=
_________<br>Roman Shpount</div></div></div></div></div>

--f46d04428ab21eb58a050633fb98--


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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-12.txt> (WebRTC Data Channels) to Proposed Standard
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> I had a suspicion that this was the intended use-case - and it's a real u=
se-case
> for certain applications/industries.
>=20
> So, for those applications (remembering that webrtc applications are not
> random external devices, but basically code linked to the usage) just eit=
her
> use pre-defined or external negotiation for channel assignments, or
> implement *in the application* allocation-choosing strategy.  Such as:
>=20
> middlebox: offer to A and B (actpass)
> A: accept (active)
> B: accept (active)
> ok, now both are on the same "evenness"
> both statically open (pre-defined) channel 0 always Both send a random
> token value The side with the lower random value is even, and the other i=
s
> odd All channel allocations are preceeded by message on channel 0 that
> amounts to an "I'm opening channel N with these params" that takes the
> place of DCEP Encapsulate this behavior in some simple functions that tak=
e
> the place of the default createDataChannel()/etc
>=20
> There are other solutions of course for glare resolution.  If you want, i=
f the
> two sides chose the opposite evenness, you just leave createDataChannel
> alone.  There are a LOT of ways to deal with this at the application leve=
l, since
> the application "should" know it's talking to (or may be talking to) a
> middlebox that will decrypt/forward things.

My discomfort and objection is with using the DTLS roles--even if it's just=
 as default.  Why can't we either leave it up to the application to allocat=
e the IDs, suggest some "best practice" approaches, or codify one as the pr=
eferred?  I don't feel strongly either way.  All I care about here is that =
we don't use the DTLS role for anything related to DCEP.  DTLS should just =
be an opaque transport for the endpoints' SCTP packets.  If we leave the DT=
LS role as default in the RFC, middleboxes have to worry about it because i=
t's part of the standard.  It doesn't matter that the application _can_ use=
 other means.   If it's in the RFC, it is part of an implied interface cont=
ract and 3rd party developers and system integrators will expect it to be s=
upported and work as specified. =20

Thanks,
--Felix


From nobody Mon Oct 27 05:11:00 2014
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-transports-07.txt
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In order to avoid the confusion, we recently published http://tools.ietf.or=
g/html/draft-patil-tram-turn-serv-selection-00 which describes TURN selecti=
on criteria, as guidelines, that can be used by a client to perform an info=
rmed TURN server selection decision when a client discovers multiple TURN s=
ervers.

-Tiru

From: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>
Date: Friday, October 24, 2014 1:01 AM
To: "Olle E. Johansson" <oej@edvina.net<mailto:oej@edvina.net>>
Cc: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-07.txt

Generally, I think there is a fair amount of confusion about how these vari=
ous types of TURN servers are going to work. The RETURN draft takes a first=
 stab at defining the interactions here, and ideally it can be adopted as a=
 WG draft and referred to by -transports.

On Thu, Oct 23, 2014 at 5:26 AM, Olle E. Johansson <oej@edvina.net<mailto:o=
ej@edvina.net>> wrote:

On 22 Oct 2014, at 16:57, Benjamin Schwartz <bemasc@webrtc.org<mailto:bemas=
c@webrtc.org>> wrote:


FYI, the purpose of the RETURN draft (https://tools.ietf.org/html/draft-sch=
wartz-rtcweb-return-03) is to try to specify the precise interaction and pr=
ecedence when both the browser and the application provide TURN servers.  (=
It's not as simple as "one or the other".)

I haven't considered the issues related to browser-provided STUN servers, t=
hough.
Cool! I'll take a look at that. Maybe Haralds draft need to refer to that.

/O



On Wed, Oct 22, 2014 at 10:51 AM, Christer Holmberg <christer.holmberg@eric=
sson.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org=
>] On Behalf Of Olle E. Johansson
Sent: 22 October 2014 17:44
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt

Section 3.4

" WebRTC browsers MUST support configuration of STUN and TURN servers,
   both from browser configuration and from an application."


Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.
If not use the one provided by the application.

Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.

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

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



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<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">In order to avoid the con=
fusion, we recently published http://tools.ietf.org/html/draft-patil-tram-t=
urn-serv-selection-00 which describes TURN selection criteria,
 as guidelines, that can be used by a client to perform an informed TURN se=
rver selection decision when a client discovers multiple TURN servers.&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">-Tiru<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
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0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;;color:black">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;;color:black">Justin Uberti &lt;<a href=3D"mailto:jub=
erti@google.com">juberti@google.com</a>&gt;<br>
<b>Date: </b>Friday, October 24, 2014 1:01 AM<br>
<b>To: </b>&quot;Olle E. Johansson&quot; &lt;<a href=3D"mailto:oej@edvina.n=
et">oej@edvina.net</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quo=
t; &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [rtcweb] draft-ietf-rtcweb-transports-07.txt<o:p></o:p>=
</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">Generally, I think there is=
 a fair amount of confusion about how these various types of TURN servers a=
re going to work. The RETURN draft takes a first stab at
 defining the interactions here, and ideally it can be adopted as a WG draf=
t and referred to by -transports.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">On Thu, Oct 23, 2014 at 5:2=
6 AM, Olle E. Johansson &lt;<a href=3D"mailto:oej@edvina.net" target=3D"_bl=
ank">oej@edvina.net</a>&gt; wrote:<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">On 22 Oct 2014, at 16:57, B=
enjamin Schwartz &lt;<a href=3D"mailto:bemasc@webrtc.org" target=3D"_blank"=
>bemasc@webrtc.org</a>&gt; wrote:<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><br>
<br>
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">FYI, the purpose of the RET=
URN draft (<a href=3D"https://tools.ietf.org/html/draft-schwartz-rtcweb-ret=
urn-03" target=3D"_blank">https://tools.ietf.org/html/draft-schwartz-rtcweb=
-return-03</a>)
 is to try to specify the precise interaction and precedence when both the =
browser and the application provide TURN servers. &nbsp;(It's not as simple=
 as &quot;one or the other&quot;.)
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">I haven't considered the is=
sues related to browser-provided STUN servers, though.<o:p></o:p></span></p=
>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">Cool! I'll take a look at t=
hat. Maybe Haralds draft need to refer to that.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#888888"><o:p>&nbsp;</o:p></span><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"hoenzb"><span style=3D"font-size:11.5=
pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#888888">/O=
</span></span><span style=3D"font-size:11.5pt;font-family:&quot;Calibri&quo=
t;,&quot;sans-serif&quot;;color:black"><o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><br>
<br>
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">On Wed, Oct 22, 2014 at 10:=
51 AM, Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt; wrote:<o:p></o=
:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black">Hi,<br>
<br>
The advantage of an application configuration is that it can be dynamically=
 provided/updated, based on location based etc - assuming the webpage provi=
der has knowledge about STUN/TURN servers, of course.<br>
<br>
Regards,<br>
<br>
Christer<o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_=
blank">rtcweb-bounces@ietf.org</a>] On Behalf Of Olle E. Johansson<br>
Sent: 22 October 2014 17:44<br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] draft-ietf-rtcweb-transports-07.txt<br>
<br>
Section 3.4<br>
<br>
&quot; WebRTC browsers MUST support configuration of STUN and TURN servers,=
<br>
&nbsp; &nbsp;both from browser configuration and from an application.&quot;=
<br>
<br>
<br>
Should we mention which takes precedence? If configured in the browser, sta=
rt with that server.<br>
If not use the one provided by the application.<br>
<br>
Maybe we should clarify that turn DNS discovery MUST be used to provide fai=
lover and load balancing.<br>
<br>
/O<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p=
>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:bla=
ck"><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></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.5pt;font-family:&quot;Ca=
libri&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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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Meaning of SHOULD support/use interleaving
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Subject: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

Within the CLUE WG, we had a discussion regarding the following statement i=
n section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:

"The support for message interleaving as defined in
                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."

First, it is a little unclear what "the support SHOULD be used" means.

Second, does it mean that any data channel protocol (e.g CLUE) SHOULD use i=
nterleaving, even if the characteristics of the protocol wouldn't require i=
t?

Regards,

Christer

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Within the CLUE WG, we had a discussion regarding th=
e following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.t=
xt:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;The support for =
message interleaving as defined in<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata] SHOULD be use=
d.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">First, it is a little unclear what &#8220;the suppor=
t SHOULD be used&#8221; means.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Second, does it mean that any data channel protocol =
(e.g CLUE) SHOULD use interleaving, even if the characteristics of the prot=
ocol wouldn&#8217;t require it?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
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From nobody Mon Oct 27 07:11:28 2014
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 Oct 2014, at 14:12, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
> =20
> =93The support for message interleaving as defined in
>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used.=94
> =20
> First, it is a little unclear what =93the support SHOULD be used=94 =
means.
My understanding is that SCTP implementations supporting the extension =
will use it.
This is negotiated during the setup of the SCTP association.
Whether messages are interleaved or not depends on the stream scheduler. =
This is
a sender side only decision. The receiver has to deal with it.
It is not a MUST, since there are implementations now in use which don't =
support
the extension.

Best regards
Michael
> =20
> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD =
use interleaving, even if the characteristics of the protocol wouldn=92t =
require it?
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

>> Within the CLUE WG, we had a discussion regarding the following statemen=
t in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>> =20
>> "The support for message interleaving as defined in
>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>> =20
>> First, it is a little unclear what "the support SHOULD be used" means.
>
> My understanding is that SCTP implementations supporting the extension wi=
ll use it.
> This is negotiated during the setup of the SCTP association.

If it's done on SCTP level, why do we need to talk about it in the data cha=
nnel draft?=20

Is there a reason why it is important to use it for data channels? If so, d=
oes it apply to data channels in general?=20

Regards,

Christer



Whether messages are interleaved or not depends on the stream scheduler. Th=
is is a sender side only decision. The receiver has to deal with it.
It is not a MUST, since there are implementations now in use which don't su=
pport the extension.

Best regards
Michael
> =20
> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD use=
 interleaving, even if the characteristics of the protocol wouldn't require=
 it?
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Oct 27 09:13:33 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : STUN Usage for Consent Freshness
        Authors         : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
                          Martin Thomson
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-08.txt
	Pages           : 9
	Date            : 2014-10-27

Abstract:
   To prevent sending excessive traffic to an endpoint, periodic consent
   needs to be obtained from that remote endpoint.

   This document describes a consent mechanism using a new Session
   Traversal Utilities for NAT (STUN) usage.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-08


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

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


From nobody Mon Oct 27 09:36:41 2014
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Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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Hi,

On 21 Oct 2014, at 19:58, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
> Not sure if it is done on pourpose, but according to the RTP usage =
draft, it may seem that full RFC 4588 is mandated at the recevier side:
>=20
>     Receivers are REQUIRED to implement support for RTP retransmission
>     packets [RFC4588].
>=20
> That would include both modes, session and ssrc multiplexing. Given =
the extensive usage of bundle and current implementations, session =
multiplexing support doesn't make much sense.
>=20
> Should we drop it, and state that only ssrc-multiplexing shall be     =
supported at the receiving end?

I don=92t see any advantage to doing so, given that support for =
non-BUNDLE sessions is REQUIRED. You need to implement the signalling =
needed for session-multiplexing of retransmission packet anyway, so =
disallowing it buys you nothing.

--=20
Colin Perkins
https://csperkins.org/





--Apple-Mail=_0412D1B8-350B-4622-BEB7-F757E7724EBA
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;">Hi,<div><br><div><div>On 21 Oct 2014, at 19:58, =
Sergio Garcia Murillo &lt;<a =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt; wrote:</div><blockquote type=3D"cite"><div =
bgcolor=3D"#FFFFFF" text=3D"#000000">Not sure if it is done on pourpose, =
but according to the RTP usage
    draft, it may seem that full RFC 4588 is mandated at the recevier
    side:<br>
    <br>
    <pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;">    Receivers are REQUIRED to implement =
support for RTP retransmission
   &nbsp;packets [<a href=3D"https://tools.ietf.org/html/rfc4588" =
title=3D"&quot;RTP Retransmission Payload =
Format&quot;">RFC4588</a>].</pre>
    <br>
    That would include both modes, session and ssrc multiplexing. Given
    the extensive usage of bundle and current implementations, session
    multiplexing support doesn't make much sense.<br>
    <br>
    Should we drop it, and state that only ssrc-multiplexing shall be
    supported at the receiving =
end?<br></div></blockquote><br></div><div>I don=92t see any advantage to =
doing so, given that support for non-BUNDLE sessions is REQUIRED. You =
need to implement the signalling needed for session-multiplexing of =
retransmission packet anyway, so disallowing it buys you =
nothing.</div><div><br></div><div apple-content-edited=3D"true"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; 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; border-spacing: 0px; -webkit-text-decorations-in-effect: none; =
-webkit-text-stroke-width: =
0px;"><div>--&nbsp;</div><div></div><div>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

--Apple-Mail=_0412D1B8-350B-4622-BEB7-F757E7724EBA--


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Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27/10/2014 17:36, Colin Perkins wrote:
> Hi,
>
> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>> Not sure if it is done on pourpose, but according to the RTP usage 
>> draft, it may seem that full RFC 4588 is mandated at the recevier side:
>>
>>      Receivers are REQUIRED to implement support for RTP retransmission
>>      packets [RFC4588  <https://tools.ietf.org/html/rfc4588>].
>>
>> That would include both modes, session and ssrc multiplexing. Given 
>> the extensive usage of bundle and current implementations, session 
>> multiplexing support doesn't make much sense.
>>
>> Should we drop it, and state that only ssrc-multiplexing shall be 
>> supported at the receiving end?
>
> I don’t see any advantage to doing so, given that support for 
> non-BUNDLE sessions is REQUIRED. You need to implement the signalling 
> needed for session-multiplexing of retransmission packet anyway, so 
> disallowing it buys you nothing.
>

You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, what I 
don't see is how to do session multiplexing with BUNDLE sessions.

Best regards
Sergio

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    <div class="moz-cite-prefix">On 27/10/2014 17:36, Colin Perkins
      wrote:<br>
    </div>
    <blockquote
      cite="mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      Hi,
      <div><br>
        <div>
          <div>On 21 Oct 2014, at 19:58, Sergio Garcia Murillo &lt;<a
              moz-do-not-send="true"
              href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;
            wrote:</div>
          <blockquote type="cite">
            <div bgcolor="#FFFFFF" text="#000000">Not sure if it is done
              on pourpose, but according to the RTP usage draft, it may
              seem that full RFC 4588 is mandated at the recevier side:<br>
              <br>
              <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are REQUIRED to implement support for RTP retransmission
    packets [<a moz-do-not-send="true" href="https://tools.ietf.org/html/rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].</pre>
              <br>
              That would include both modes, session and ssrc
              multiplexing. Given the extensive usage of bundle and
              current implementations, session multiplexing support
              doesn't make much sense.<br>
              <br>
              Should we drop it, and state that only ssrc-multiplexing
              shall be supported at the receiving end?<br>
            </div>
          </blockquote>
          <br>
        </div>
        <div>I don’t see any advantage to doing so, given that support
          for non-BUNDLE sessions is REQUIRED. You need to implement the
          signalling needed for session-multiplexing of retransmission
          packet anyway, so disallowing it buys you nothing.</div>
        <br>
      </div>
    </blockquote>
    <br>
    You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions,
    what I don't see is how to do session multiplexing with BUNDLE
    sessions. <br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------060408080202040307060508--


From nobody Mon Oct 27 09:42:11 2014
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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Subject: [rtcweb] In case you missed it... MSFT and WebRTC
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http://blogs.skype.com/2014/10/27/bringing-interoperable-real-time-communications-to-the-web/

https://status.modern.ie/webrtcwebrtcv10api?term=webrtc

-Victor


From nobody Mon Oct 27 09:46:11 2014
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Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27 Oct 2014, at 16:40, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
> On 27/10/2014 17:36, Colin Perkins wrote:
>> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>> Not sure if it is done on pourpose, but according to the RTP usage =
draft, it may seem that full RFC 4588 is mandated at the recevier side:
>>>=20
>>>     Receivers are REQUIRED to implement support for RTP =
retransmission
>>>     packets [RFC4588].
>>>=20
>>> That would include both modes, session and ssrc multiplexing. Given =
the extensive usage of bundle and current implementations, session =
multiplexing support doesn't make much sense.
>>>=20
>>> Should we drop it, and state that only ssrc-multiplexing shall be =
supported at the receiving end?
>>=20
>> I don=92t see any advantage to doing so, given that support for =
non-BUNDLE sessions is REQUIRED. You need to implement the signalling =
needed for session-multiplexing of retransmission packet anyway, so =
disallowing it buys you nothing.
>=20
> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, what =
I don't see is how to do session multiplexing with BUNDLE sessions.=20

You can=92t do session multiplexing for BUNDLE sessions; by definition =
they use SSRC multiplexing. You could do non-BUNDLE sessions, with =
retransmission sent on a separate RTP session though.

--=20
Colin Perkins
https://csperkins.org/





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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On 27 =
Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt; wrote:<div><blockquote type=3D"cite">
 =20
    <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 27/10/2014 17:36, Colin Perkins
      wrote:<br>
    </div>
    <blockquote =
cite=3D"mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org" =
type=3D"cite"><div><div><div>On 21 Oct 2014, at 19:58, Sergio Garcia =
Murillo &lt;<a moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;
            wrote:</div>
          <blockquote type=3D"cite">
            <div bgcolor=3D"#FFFFFF" text=3D"#000000">Not sure if it is =
done
              on pourpose, but according to the RTP usage draft, it may
              seem that full RFC 4588 is mandated at the recevier =
side:<br>
              <br>
              <pre class=3D"newpage" style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are =
REQUIRED to implement support for RTP retransmission
   &nbsp;packets [<a moz-do-not-send=3D"true" =
href=3D"https://tools.ietf.org/html/rfc4588" title=3D"&quot;RTP =
Retransmission Payload Format&quot;">RFC4588</a>].</pre>
              <br>
              That would include both modes, session and ssrc
              multiplexing. Given the extensive usage of bundle and
              current implementations, session multiplexing support
              doesn't make much sense.<br>
              <br>
              Should we drop it, and state that only ssrc-multiplexing
              shall be supported at the receiving end?<br>
            </div>
          </blockquote>
          <br>
        </div>
        <div>I don=92t see any advantage to doing so, given that support
          for non-BUNDLE sessions is REQUIRED. You need to implement the
          signalling needed for session-multiplexing of retransmission
          packet anyway, so disallowing it buys you nothing.</div>
      </div>
    </blockquote>
    <br>
    You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions,
    what I don't see is how to do session multiplexing with BUNDLE
    sessions. <br></div></blockquote><br></div><div>You can=92t do =
session multiplexing for BUNDLE sessions; by definition they use SSRC =
multiplexing. You could do non-BUNDLE sessions, with retransmission sent =
on a separate RTP session though.</div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; border-spacing: 0px;"><div><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></body></html>=

--Apple-Mail=_0EE27D34-EFAB-41C1-9854-73CFFF84BDA8--


From nobody Mon Oct 27 09:51:54 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-08.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-08.txt
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This revision addresses all the WGLC comments received on the ver 07 of
draft.=20

Summary of major things addressed:
Comments received from Chritser, Marc, Christian among others.
Removed the Liveness portion from the document as per the discussions in
the mailer -=20
http://www.ietf.org/mail-archive/web/rtcweb/current/msg13135.html
Clarified on scope of Consent (Consent is for sender of data/media)

Editorial changes by Martin to make the text simple (with out changing the
functionality or Algorithm).

Diff:=20
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshness=
-0
8

Regards,
Ram

-----Original Message-----
From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
Date: Monday, 27 October 2014 9:43 pm
To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-08.txt

>
>A New Internet-Draft is available from the on-line Internet-Drafts
>directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers
>Working Group of the IETF.
>
>        Title           : STUN Usage for Consent Freshness
>        Authors         : Muthu Arul Mozhi Perumal
>                          Dan Wing
>                          Ram Mohan Ravindranath
>                          Tirumaleswar Reddy
>                          Martin Thomson
>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-08.txt
>	Pages           : 9
>	Date            : 2014-10-27
>
>Abstract:
>   To prevent sending excessive traffic to an endpoint, periodic consent
>   needs to be obtained from that remote endpoint.
>
>   This document describes a consent mechanism using a new Session
>   Traversal Utilities for NAT (STUN) usage.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-08
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshnes=
s-
>08
>
>
>Please note that it may take a couple of minutes from the time of
>submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Mon, 27 Oct 2014 17:51:41 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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To: Colin Perkins <csp@csperkins.org>
References: <5446ACD8.1010004@gmail.com> <B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org> <544E7586.4080703@gmail.com> <22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27/10/2014 17:45, Colin Perkins wrote:
> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>> On 27/10/2014 17:36, Colin Perkins wrote:
>>> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo 
>>> <sergio.garcia.murillo@gmail.com 
>>> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>>>> Not sure if it is done on pourpose, but according to the RTP usage 
>>>> draft, it may seem that full RFC 4588 is mandated at the recevier side:
>>>>
>>>>      Receivers are REQUIRED to implement support for RTP retransmission
>>>>      packets [RFC4588  <https://tools.ietf.org/html/rfc4588>].
>>>>
>>>> That would include both modes, session and ssrc multiplexing. Given 
>>>> the extensive usage of bundle and current implementations, session 
>>>> multiplexing support doesn't make much sense.
>>>>
>>>> Should we drop it, and state that only ssrc-multiplexing shall be 
>>>> supported at the receiving end?
>>>
>>> I don’t see any advantage to doing so, given that support for 
>>> non-BUNDLE sessions is REQUIRED. You need to implement the 
>>> signalling needed for session-multiplexing of retransmission packet 
>>> anyway, so disallowing it buys you nothing.
>>
>> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, 
>> what I don't see is how to do session multiplexing with BUNDLE sessions.
>
> You can’t do session multiplexing for BUNDLE sessions; by definition 
> they use SSRC multiplexing. You could do non-BUNDLE sessions, with 
> retransmission sent on a separate RTP session though.
>
So, you are saying exactly the same than me. SSRC multiplexing supports 
both BUNDLE and NON-BUNDLE. So, why require support for session 
multiplexing at all? As a developer, I don't see why I would have to 
implement something that would be rarely used and provide no extra benefit.

Also, the original discussion came from the ORTC list, as ORTC API only 
supports ssrc-multiplexing RTX. If we require both modes on WebRTC "just 
for fun", then ORTC API will not be able to comply with the RTP usage draft.

Best regards
Sergio

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    <div class="moz-cite-prefix">On 27/10/2014 17:45, Colin Perkins
      wrote:<br>
    </div>
    <blockquote
      cite="mid:22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a
        moz-do-not-send="true"
        href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;
      wrote:
      <div>
        <blockquote type="cite">
          <meta content="text/html; charset=windows-1252"
            http-equiv="Content-Type">
          <div bgcolor="#FFFFFF" text="#000000">
            <div class="moz-cite-prefix">On 27/10/2014 17:36, Colin
              Perkins wrote:<br>
            </div>
            <blockquote
              cite="mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org"
              type="cite">
              <div>
                <div>
                  <div>On 21 Oct 2014, at 19:58, Sergio Garcia Murillo
                    &lt;<a moz-do-not-send="true"
                      href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;

                    wrote:</div>
                  <blockquote type="cite">
                    <div bgcolor="#FFFFFF" text="#000000">Not sure if it
                      is done on pourpose, but according to the RTP
                      usage draft, it may seem that full RFC 4588 is
                      mandated at the recevier side:<br>
                      <br>
                      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are REQUIRED to implement support for RTP retransmission
    packets [<a moz-do-not-send="true" href="https://tools.ietf.org/html/rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].</pre>
                      <br>
                      That would include both modes, session and ssrc
                      multiplexing. Given the extensive usage of bundle
                      and current implementations, session multiplexing
                      support doesn't make much sense.<br>
                      <br>
                      Should we drop it, and state that only
                      ssrc-multiplexing shall be supported at the
                      receiving end?<br>
                    </div>
                  </blockquote>
                  <br>
                </div>
                <div>I don’t see any advantage to doing so, given that
                  support for non-BUNDLE sessions is REQUIRED. You need
                  to implement the signalling needed for
                  session-multiplexing of retransmission packet anyway,
                  so disallowing it buys you nothing.</div>
              </div>
            </blockquote>
            <br>
            You can do SSRC multiplexing with BUNDLE and non-BUNDLE
            sessions, what I don't see is how to do session multiplexing
            with BUNDLE sessions. <br>
          </div>
        </blockquote>
        <br>
      </div>
      <div>You can’t do session multiplexing for BUNDLE sessions; by
        definition they use SSRC multiplexing. You could do non-BUNDLE
        sessions, with retransmission sent on a separate RTP session
        though.</div>
      <div apple-content-edited="true"><span class="Apple-style-span"
          style="border-collapse: separate; border-spacing: 0px;">
          <div><br class="khtml-block-placeholder">
          </div>
        </span></div>
    </blockquote>
    So, you are saying exactly the same than me. SSRC multiplexing
    supports both BUNDLE and NON-BUNDLE. So, why require support for
    session multiplexing at all? As a developer, I don't see why I would
    have to implement something that would be rarely used and provide no
    extra benefit.<br>
    <br>
    Also, the original discussion came from the ORTC list, as ORTC API
    only supports ssrc-multiplexing RTX. If we require both modes on
    WebRTC "just for fun", then ORTC API will not be able to comply with
    the RTP usage draft.<br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------010507060802070206040809--


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Date: Mon, 27 Oct 2014 17:10:45 +0000
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References: <5446ACD8.1010004@gmail.com> <B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org> <544E7586.4080703@gmail.com> <22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org> <544E781D.50305@gmail.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27 Oct 2014, at 16:51, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
> On 27/10/2014 17:45, Colin Perkins wrote:
>> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>> On 27/10/2014 17:36, Colin Perkins wrote:
>>>> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>>>> Not sure if it is done on pourpose, but according to the RTP usage =
draft, it may seem that full RFC 4588 is mandated at the recevier side:
>>>>>=20
>>>>>     Receivers are REQUIRED to implement support for RTP =
retransmission
>>>>>     packets [RFC4588].
>>>>>=20
>>>>> That would include both modes, session and ssrc multiplexing. =
Given the extensive usage of bundle and current implementations, session =
multiplexing support doesn't make much sense.
>>>>>=20
>>>>> Should we drop it, and state that only ssrc-multiplexing shall be =
supported at the receiving end?
>>>>=20
>>>> I don=92t see any advantage to doing so, given that support for =
non-BUNDLE sessions is REQUIRED. You need to implement the signalling =
needed for session-multiplexing of retransmission packet anyway, so =
disallowing it buys you nothing.
>>>=20
>>> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, =
what I don't see is how to do session multiplexing with BUNDLE sessions.=20=

>>=20
>> You can=92t do session multiplexing for BUNDLE sessions; by =
definition they use SSRC multiplexing. You could do non-BUNDLE sessions, =
with retransmission sent on a separate RTP session though.
>>=20
> So, you are saying exactly the same than me. SSRC multiplexing =
supports both BUNDLE and NON-BUNDLE. So, why require support for session =
multiplexing at all? As a developer, I don't see why I would have to =
implement something that would be rarely used and provide no extra =
benefit.

Non-BUNDLE is session multiplexing. It uses a separate RTP session for =
the retransmissions.=20

> Also, the original discussion came from the ORTC list, as ORTC API =
only supports ssrc-multiplexing RTX. If we require both modes on WebRTC =
"just for fun", then ORTC API will not be able to comply with the RTP =
usage draft.

I=92m perhaps missing something, but it seems that the features needed =
for session-multiplexed retransmission are required anyway (a=3Dgroup, =
plus the ability to send non-BUNDLE media).=20

--=20
Colin Perkins
https://csperkins.org/





--Apple-Mail=_1EBC4EB5-4902-496C-88C9-AC65E685503A
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On 27 =
Oct 2014, at 16:51, Sergio Garcia Murillo &lt;<a =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt; wrote:<div><blockquote type=3D"cite">
 =20
    <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 27/10/2014 17:45, Colin Perkins
      wrote:<br>
    </div>
    <blockquote =
cite=3D"mid:22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org" =
type=3D"cite">
      <meta http-equiv=3D"Content-Type" content=3D"text/html;
        charset=3Dwindows-1252">
      On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a =
moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;
      wrote:
      <div>
        <blockquote type=3D"cite">
          <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
          <div bgcolor=3D"#FFFFFF" text=3D"#000000">
            <div class=3D"moz-cite-prefix">On 27/10/2014 17:36, Colin
              Perkins wrote:<br>
            </div>
            <blockquote =
cite=3D"mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org" =
type=3D"cite">
              <div>
                <div>
                  <div>On 21 Oct 2014, at 19:58, Sergio Garcia Murillo
                    &lt;<a moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;

                    wrote:</div>
                  <blockquote type=3D"cite">
                    <div bgcolor=3D"#FFFFFF" text=3D"#000000">Not sure =
if it
                      is done on pourpose, but according to the RTP
                      usage draft, it may seem that full RFC 4588 is
                      mandated at the recevier side:<br>
                      <br>
                      <pre class=3D"newpage" style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are =
REQUIRED to implement support for RTP retransmission
   &nbsp;packets [<a moz-do-not-send=3D"true" =
href=3D"https://tools.ietf.org/html/rfc4588" title=3D"&quot;RTP =
Retransmission Payload Format&quot;">RFC4588</a>].</pre>
                      <br>
                      That would include both modes, session and ssrc
                      multiplexing. Given the extensive usage of bundle
                      and current implementations, session multiplexing
                      support doesn't make much sense.<br>
                      <br>
                      Should we drop it, and state that only
                      ssrc-multiplexing shall be supported at the
                      receiving end?<br>
                    </div>
                  </blockquote>
                  <br>
                </div>
                <div>I don=92t see any advantage to doing so, given that
                  support for non-BUNDLE sessions is REQUIRED. You need
                  to implement the signalling needed for
                  session-multiplexing of retransmission packet anyway,
                  so disallowing it buys you nothing.</div>
              </div>
            </blockquote>
            <br>
            You can do SSRC multiplexing with BUNDLE and non-BUNDLE
            sessions, what I don't see is how to do session multiplexing
            with BUNDLE sessions. <br>
          </div>
        </blockquote>
        <br>
      </div>
      <div>You can=92t do session multiplexing for BUNDLE sessions; by
        definition they use SSRC multiplexing. You could do non-BUNDLE
        sessions, with retransmission sent on a separate RTP session
        though.</div>
      <div apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; border-spacing: 0px;">
          <div><br class=3D"khtml-block-placeholder">
          </div>
        </span></div>
    </blockquote>
    So, you are saying exactly the same than me. SSRC multiplexing
    supports both BUNDLE and NON-BUNDLE. So, why require support for
    session multiplexing at all? As a developer, I don't see why I would
    have to implement something that would be rarely used and provide no
    extra benefit.<br></div></blockquote><div><br></div><div>Non-BUNDLE =
is session multiplexing. It uses a separate RTP session for the =
retransmissions.&nbsp;</div><br><blockquote type=3D"cite"><div =
bgcolor=3D"#FFFFFF" text=3D"#000000">
    Also, the original discussion came from the ORTC list, as ORTC API
    only supports ssrc-multiplexing RTX. If we require both modes on
    WebRTC "just for fun", then ORTC API will not be able to comply with
    the RTP usage draft.<br></div></blockquote><br></div><div>I=92m =
perhaps missing something, but it seems that the features needed for =
session-multiplexed retransmission are required anyway (a=3Dgroup, plus =
the ability to send non-BUNDLE media).&nbsp;</div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; 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; border-spacing: =
0px; -webkit-text-decorations-in-effect: none; =
-webkit-text-stroke-width: 0px;"><div><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></body></html>=

--Apple-Mail=_1EBC4EB5-4902-496C-88C9-AC65E685503A--


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Date: Mon, 27 Oct 2014 18:17:31 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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To: Colin Perkins <csp@csperkins.org>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27/10/2014 18:10, Colin Perkins wrote:
> On 27 Oct 2014, at 16:51, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>> On 27/10/2014 17:45, Colin Perkins wrote:
>>> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo 
>>> <sergio.garcia.murillo@gmail.com 
>>> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>>>> On 27/10/2014 17:36, Colin Perkins wrote:
>>>>> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo 
>>>>> <sergio.garcia.murillo@gmail.com 
>>>>> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>>>>>> Not sure if it is done on pourpose, but according to the RTP 
>>>>>> usage draft, it may seem that full RFC 4588 is mandated at the 
>>>>>> recevier side:
>>>>>>
>>>>>>      Receivers are REQUIRED to implement support for RTP retransmission
>>>>>>      packets [RFC4588  <https://tools.ietf.org/html/rfc4588>].
>>>>>>
>>>>>> That would include both modes, session and ssrc multiplexing. 
>>>>>> Given the extensive usage of bundle and current implementations, 
>>>>>> session multiplexing support doesn't make much sense.
>>>>>>
>>>>>> Should we drop it, and state that only ssrc-multiplexing shall be 
>>>>>> supported at the receiving end?
>>>>>
>>>>> I don’t see any advantage to doing so, given that support for 
>>>>> non-BUNDLE sessions is REQUIRED. You need to implement the 
>>>>> signalling needed for session-multiplexing of retransmission 
>>>>> packet anyway, so disallowing it buys you nothing.
>>>>
>>>> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, 
>>>> what I don't see is how to do session multiplexing with BUNDLE 
>>>> sessions.
>>>
>>> You can’t do session multiplexing for BUNDLE sessions; by definition 
>>> they use SSRC multiplexing. You could do non-BUNDLE sessions, with 
>>> retransmission sent on a separate RTP session though.
>>>
>> So, you are saying exactly the same than me. SSRC multiplexing 
>> supports both BUNDLE and NON-BUNDLE. So, why require support for 
>> session multiplexing at all? As a developer, I don't see why I would 
>> have to implement something that would be rarely used and provide no 
>> extra benefit.
>
> Non-BUNDLE is session multiplexing. It uses a separate RTP session for 
> the retransmissions.
Maybe I am the missing something, if you don't use bundle to send the 
audio/video on same rtpsession, you can still send rtx+video on same 
session. That's it non-bundle with ssrc-multiplexing. Are we referring 
to different things?

>
>> Also, the original discussion came from the ORTC list, as ORTC API 
>> only supports ssrc-multiplexing RTX. If we require both modes on 
>> WebRTC "just for fun", then ORTC API will not be able to comply with 
>> the RTP usage draft.
>
> I’m perhaps missing something, but it seems that the features needed 
> for session-multiplexed retransmission are required anyway (a=group, 
> plus the ability to send non-BUNDLE media).
>

In current ORTC API, the RTX parameters are configured on the 
RTPReceiver/Sender object used for the media.

Best regards
Sergio

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    <div class="moz-cite-prefix">On 27/10/2014 18:10, Colin Perkins
      wrote:<br>
    </div>
    <blockquote
      cite="mid:17742E9C-CCDD-4EFE-A2C1-C84531A0523F@csperkins.org"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      On 27 Oct 2014, at 16:51, Sergio Garcia Murillo &lt;<a
        moz-do-not-send="true"
        href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;
      wrote:
      <div>
        <blockquote type="cite">
          <meta content="text/html; charset=windows-1252"
            http-equiv="Content-Type">
          <div bgcolor="#FFFFFF" text="#000000">
            <div class="moz-cite-prefix">On 27/10/2014 17:45, Colin
              Perkins wrote:<br>
            </div>
            <blockquote
              cite="mid:22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org"
              type="cite">
              <meta http-equiv="Content-Type" content="text/html;
                charset=windows-1252">
              On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a
                moz-do-not-send="true"
                href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;

              wrote:
              <div>
                <blockquote type="cite">
                  <meta content="text/html; charset=windows-1252"
                    http-equiv="Content-Type">
                  <div bgcolor="#FFFFFF" text="#000000">
                    <div class="moz-cite-prefix">On 27/10/2014 17:36,
                      Colin Perkins wrote:<br>
                    </div>
                    <blockquote
                      cite="mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org"
                      type="cite">
                      <div>
                        <div>
                          <div>On 21 Oct 2014, at 19:58, Sergio Garcia
                            Murillo &lt;<a moz-do-not-send="true"
                              href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a>&gt;


                            wrote:</div>
                          <blockquote type="cite">
                            <div bgcolor="#FFFFFF" text="#000000">Not
                              sure if it is done on pourpose, but
                              according to the RTP usage draft, it may
                              seem that full RFC 4588 is mandated at the
                              recevier side:<br>
                              <br>
                              <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are REQUIRED to implement support for RTP retransmission
    packets [<a moz-do-not-send="true" href="https://tools.ietf.org/html/rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].</pre>
                              <br>
                              That would include both modes, session and
                              ssrc multiplexing. Given the extensive
                              usage of bundle and current
                              implementations, session multiplexing
                              support doesn't make much sense.<br>
                              <br>
                              Should we drop it, and state that only
                              ssrc-multiplexing shall be supported at
                              the receiving end?<br>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                        <div>I don’t see any advantage to doing so,
                          given that support for non-BUNDLE sessions is
                          REQUIRED. You need to implement the signalling
                          needed for session-multiplexing of
                          retransmission packet anyway, so disallowing
                          it buys you nothing.</div>
                      </div>
                    </blockquote>
                    <br>
                    You can do SSRC multiplexing with BUNDLE and
                    non-BUNDLE sessions, what I don't see is how to do
                    session multiplexing with BUNDLE sessions. <br>
                  </div>
                </blockquote>
                <br>
              </div>
              <div>You can’t do session multiplexing for BUNDLE
                sessions; by definition they use SSRC multiplexing. You
                could do non-BUNDLE sessions, with retransmission sent
                on a separate RTP session though.</div>
              <div apple-content-edited="true"><span
                  class="Apple-style-span" style="border-collapse:
                  separate; border-spacing: 0px;">
                  <div><br class="khtml-block-placeholder">
                  </div>
                </span></div>
            </blockquote>
            So, you are saying exactly the same than me. SSRC
            multiplexing supports both BUNDLE and NON-BUNDLE. So, why
            require support for session multiplexing at all? As a
            developer, I don't see why I would have to implement
            something that would be rarely used and provide no extra
            benefit.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>Non-BUNDLE is session multiplexing. It uses a separate RTP
          session for the retransmissions. <br>
        </div>
      </div>
    </blockquote>
    Maybe I am the missing something, if you don't use bundle to send
    the audio/video on same rtpsession, you can still send rtx+video on
    same session. That's it non-bundle with ssrc-multiplexing. Are we
    referring to different things?<br>
    <br>
    <blockquote
      cite="mid:17742E9C-CCDD-4EFE-A2C1-C84531A0523F@csperkins.org"
      type="cite">
      <div><br>
        <blockquote type="cite">
          <div bgcolor="#FFFFFF" text="#000000"> Also, the original
            discussion came from the ORTC list, as ORTC API only
            supports ssrc-multiplexing RTX. If we require both modes on
            WebRTC "just for fun", then ORTC API will not be able to
            comply with the RTP usage draft.<br>
          </div>
        </blockquote>
        <br>
      </div>
      <div>I’m perhaps missing something, but it seems that the features
        needed for session-multiplexed retransmission are required
        anyway (a=group, plus the ability to send non-BUNDLE media). </div>
      <br>
    </blockquote>
    <br>
    In current ORTC API, the RTX parameters are configured on the
    RTPReceiver/Sender object used for the media. <br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------090509010505020809070501--


From nobody Mon Oct 27 11:11:06 2014
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To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-08.txt - Some additional questions
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-08.txt - Some additional questions
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Hi,

Thanks for putting together the new version! It makes a number of things mo=
re clear :)

I do still have a few comments/questions (hopefully editorial ones), though=
:

Q1:
-----

I think it would be good to explicitly clarify that, when consent expires o=
n a 5-tuple/candidate pair, it does not affect other 5-tuples within the se=
ssion. I have received questions on that, so I think some words would be us=
eful to make it more clear.

(Of course, one CAN choose to terminate the whole session if consent expire=
s on a candidate pair, but that depends on application policy).


Q2:
-----

If consent expires, and I stop sending application data, will I still send =
STUN keep-alives etc, to keep the candidate pair, NAT bindings etc alive?=20

There IS text saying:

	"An endpoint that is not sending any application data does not need to
   	maintain consent.  However, the endpoint needs to ensure its NAT or
   	firewall mappings persist which can be done using keepalive or other
   	techniques (see Section 10 of [RFC5245] and see [RFC6263])."

However, that seems to be more related to cases where I don't intend to sen=
d application data to begin with.

So, perhaps the text above can be modified, to clarify that keepalives etc =
will still be sent if consent expires.


Q3:
-----

Section 4.1. says:

	"If the endpoint wants to send application data, it needs to first obtain
   	consent if its consent has expired."

If my consent has expired, is the a time period before I can try to obtain =
consent again? Can I just continue sending STUN binding requests even if my=
 consent expires, and wait to obtain consent again?

(I obviously won't send any application data until I obtain consent again.)


Q4:
-----

When my consent expires, is it considered an error if I also choose to not =
receive any data? If not, I think it would be good to indicate that one may=
 choose to also stop receiving data in case consent expires.


Q5:
-----

Assuming I send RTP and RTCP on separate 5-tuples, and consent expires on o=
ne of the 5-tuples. I assume that will impact both 5-tuples, and in that ca=
se I think it would be good to clarify.

Thanks!

Regards,

Christer


From nobody Mon Oct 27 11:27:19 2014
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 10/27/2014 07:19 AM, Christer Holmberg wrote:
> Hi,
>
>>> Within the CLUE WG, we had a discussion regarding the following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>  
>>> "The support for message interleaving as defined in
>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>  
>>> First, it is a little unclear what "the support SHOULD be used" means.
>> My understanding is that SCTP implementations supporting the extension will use it.
>> This is negotiated during the setup of the SCTP association.
> If it's done on SCTP level, why do we need to talk about it in the data channel draft? 
>
> Is there a reason why it is important to use it for data channels? If so, does it apply to data channels in general? 

NDATA was added in order to avoid head-of-line blocking on the transport
(if I understand this correctly, until this was added, sending a huge
message would block the delivery of small messages on all channels until
the huge message was fully delivered).

Unlike Michael, I see no reason to make this a SHOULD; I think it should
be a MUST, and the older implementations in browsers should just be
called out as non-conformant.

That said, I think that data channels ought to interoperate successfully
with implementations that don't support the extension - but data channel
implementations in WebRTC endpoints should be under a "MUST implement,
MUST offer" ruleset.



>
> Regards,
>
> Christer
>
>
>
> Whether messages are interleaved or not depends on the stream scheduler. This is a sender side only decision. The receiver has to deal with it.
> It is not a MUST, since there are implementations now in use which don't support the extension.
>
> Best regards
> Michael
>>  
>> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD use interleaving, even if the characteristics of the protocol wouldn't require it?
>>  
>> 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


-- 
Surveillance is pervasive. Go Dark.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

>>>> Within the CLUE WG, we had a discussion regarding the following statem=
ent in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>> =20
>>>> "The support for message interleaving as defined in
>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>> =20
>>>> First, it is a little unclear what "the support SHOULD be used" means.
>>> My understanding is that SCTP implementations supporting the extension =
will use it.
>>> This is negotiated during the setup of the SCTP association.
>> If it's done on SCTP level, why do we need to talk about it in the data =
channel draft?=20
>>
>> Is there a reason why it is important to use it for data channels? If so=
, does it apply to data channels in general?=20
>
> NDATA was added in order to avoid head-of-line blocking on the transport =
(if I understand this correctly, until=20
> this was added, sending a huge message would block the delivery of small =
messages on all channels until the huge message was fully delivered).
>
> Unlike Michael, I see no reason to make this a SHOULD; I think it should =
be a MUST, and the older=20
> implementations in browsers should just be called out as non-conformant.
>
> That said, I think that data channels ought to interoperate successfully =
with implementations that don't support the=20
> extension - but data channel implementations in WebRTC endpoints should b=
e under a "MUST implement, MUST offer" ruleset.

First, keep in mind that I am NOT talking about WebRTC endpoints (which is =
one reason we shouldn't call it webrtc data channel to begin with, but that=
's another story...), but ANY type of endpoint that wants to use a data cha=
nnel.=20

Second, I am not talking about MUST support, but MUST use/offer.

Assume I have a CLUE endpoint, which will use a data channel ONLY to transp=
ort CLUE messages. CLUE messages aren't that hugeto begin with, and there w=
ill be no blocking of small messages on other channels - as there are no ot=
her channels.

So, why does the CLUE endpoint need to offer interleaving?

Regards,

Christer


> Whether messages are interleaved or not depends on the stream scheduler. =
This is a sender side only decision. The receiver has to deal with it.
> It is not a MUST, since there are implementations now in use which don't =
support the extension.
>
> Best regards
> Michael
>> =20
>> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD us=
e interleaving, even if the characteristics of the protocol wouldn't requir=
e it?
>> =20
>> Regards,
>> =20
>> 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


--
Surveillance is pervasive. Go Dark.

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


From nobody Mon Oct 27 12:17:19 2014
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 27 Oct 2014, at 17:17, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
> On 27/10/2014 18:10, Colin Perkins wrote:
>> On 27 Oct 2014, at 16:51, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>> On 27/10/2014 17:45, Colin Perkins wrote:
>>>> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>>>> On 27/10/2014 17:36, Colin Perkins wrote:
>>>>>> On 21 Oct 2014, at 19:58, Sergio Garcia Murillo =
<sergio.garcia.murillo@gmail.com> wrote:
>>>>>>> Not sure if it is done on pourpose, but according to the RTP =
usage draft, it may seem that full RFC 4588 is mandated at the recevier =
side:
>>>>>>>=20
>>>>>>>     Receivers are REQUIRED to implement support for RTP =
retransmission
>>>>>>>     packets [RFC4588].
>>>>>>>=20
>>>>>>> That would include both modes, session and ssrc multiplexing. =
Given the extensive usage of bundle and current implementations, session =
multiplexing support doesn't make much sense.
>>>>>>>=20
>>>>>>> Should we drop it, and state that only ssrc-multiplexing shall =
be supported at the receiving end?
>>>>>>=20
>>>>>> I don=92t see any advantage to doing so, given that support for =
non-BUNDLE sessions is REQUIRED. You need to implement the signalling =
needed for session-multiplexing of retransmission packet anyway, so =
disallowing it buys you nothing.
>>>>>=20
>>>>> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, =
what I don't see is how to do session multiplexing with BUNDLE sessions.=20=

>>>>=20
>>>> You can=92t do session multiplexing for BUNDLE sessions; by =
definition they use SSRC multiplexing. You could do non-BUNDLE sessions, =
with retransmission sent on a separate RTP session though.
>>>>=20
>>> So, you are saying exactly the same than me. SSRC multiplexing =
supports both BUNDLE and NON-BUNDLE. So, why require support for session =
multiplexing at all? As a developer, I don't see why I would have to =
implement something that would be rarely used and provide no extra =
benefit.
>>=20
>> Non-BUNDLE is session multiplexing. It uses a separate RTP session =
for the retransmissions.=20
> Maybe I am the missing something, if you don't use bundle to send the =
audio/video on same rtpsession, you can still send rtx+video on     same =
session. That's it non-bundle with ssrc-multiplexing. Are we referring =
to different things?

Sure, but that doesn=92t alter the fact that the group decided that =
non-BUNDLE media needs to be supported. Sending retransmission on a =
separate RTP session is as needed for interoperability with legacy =
systems as sending audio and video on separate RTP sessions (and =
shouldn=92t be hard to support, since it uses the same mechanisms).

Colin



--=20
Colin Perkins
https://csperkins.org/





--Apple-Mail=_F0FBFB17-07D7-484B-BB29-D25DBC347429
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On 27 =
Oct 2014, at 17:17, Sergio Garcia Murillo &lt;<a =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt; wrote:<div><blockquote type=3D"cite">
 =20
    <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 27/10/2014 18:10, Colin Perkins
      wrote:<br>
    </div>
    <blockquote =
cite=3D"mid:17742E9C-CCDD-4EFE-A2C1-C84531A0523F@csperkins.org" =
type=3D"cite">
      <meta http-equiv=3D"Content-Type" content=3D"text/html;
        charset=3Dwindows-1252">
      On 27 Oct 2014, at 16:51, Sergio Garcia Murillo &lt;<a =
moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;
      wrote:
      <div>
        <blockquote type=3D"cite">
          <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
          <div bgcolor=3D"#FFFFFF" text=3D"#000000">
            <div class=3D"moz-cite-prefix">On 27/10/2014 17:45, Colin
              Perkins wrote:<br>
            </div>
            <blockquote =
cite=3D"mid:22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org" =
type=3D"cite">
              <meta http-equiv=3D"Content-Type" content=3D"text/html;
                charset=3Dwindows-1252">
              On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a =
moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;

              wrote:
              <div>
                <blockquote type=3D"cite">
                  <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type">
                  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
                    <div class=3D"moz-cite-prefix">On 27/10/2014 17:36,
                      Colin Perkins wrote:<br>
                    </div>
                    <blockquote =
cite=3D"mid:B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org" =
type=3D"cite">
                      <div>
                        <div>
                          <div>On 21 Oct 2014, at 19:58, Sergio Garcia
                            Murillo &lt;<a moz-do-not-send=3D"true" =
href=3D"mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmai=
l.com</a>&gt;


                            wrote:</div>
                          <blockquote type=3D"cite">
                            <div bgcolor=3D"#FFFFFF" text=3D"#000000">Not
                              sure if it is done on pourpose, but
                              according to the RTP usage draft, it may
                              seem that full RFC 4588 is mandated at the
                              recevier side:<br>
                              <br>
                              <pre class=3D"newpage" style=3D"font-size: =
1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;">    Receivers are =
REQUIRED to implement support for RTP retransmission
   &nbsp;packets [<a moz-do-not-send=3D"true" =
href=3D"https://tools.ietf.org/html/rfc4588" title=3D"&quot;RTP =
Retransmission Payload Format&quot;">RFC4588</a>].</pre>
                              <br>
                              That would include both modes, session and
                              ssrc multiplexing. Given the extensive
                              usage of bundle and current
                              implementations, session multiplexing
                              support doesn't make much sense.<br>
                              <br>
                              Should we drop it, and state that only
                              ssrc-multiplexing shall be supported at
                              the receiving end?<br>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                        <div>I don=92t see any advantage to doing so,
                          given that support for non-BUNDLE sessions is
                          REQUIRED. You need to implement the signalling
                          needed for session-multiplexing of
                          retransmission packet anyway, so disallowing
                          it buys you nothing.</div>
                      </div>
                    </blockquote>
                    <br>
                    You can do SSRC multiplexing with BUNDLE and
                    non-BUNDLE sessions, what I don't see is how to do
                    session multiplexing with BUNDLE sessions. <br>
                  </div>
                </blockquote>
                <br>
              </div>
              <div>You can=92t do session multiplexing for BUNDLE
                sessions; by definition they use SSRC multiplexing. You
                could do non-BUNDLE sessions, with retransmission sent
                on a separate RTP session though.</div>
              <div apple-content-edited=3D"true"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;">
                  <div><br class=3D"khtml-block-placeholder">
                  </div>
                </span></div>
            </blockquote>
            So, you are saying exactly the same than me. SSRC
            multiplexing supports both BUNDLE and NON-BUNDLE. So, why
            require support for session multiplexing at all? As a
            developer, I don't see why I would have to implement
            something that would be rarely used and provide no extra
            benefit.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>Non-BUNDLE is session multiplexing. It uses a separate RTP
          session for the retransmissions. <br>
        </div>
      </div>
    </blockquote>
    Maybe I am the missing something, if you don't use bundle to send
    the audio/video on same rtpsession, you can still send rtx+video on
    same session. That's it non-bundle with ssrc-multiplexing. Are we
    referring to different =
things?</div></blockquote><div><br></div><div>Sure, but that doesn=92t =
alter the fact that the group decided that non-BUNDLE media needs to be =
supported. Sending retransmission on a separate RTP session is as needed =
for interoperability with legacy systems as sending audio and video on =
separate RTP sessions (and shouldn=92t be hard to support, since it uses =
the same =
mechanisms).</div><div><br></div><div>Colin</div><div><br></div><div><br><=
/div></div><div apple-content-edited=3D"true"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; 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; border-spacing: 0px; -webkit-text-decorations-in-effect: none; =
-webkit-text-stroke-width: 0px;"><div><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></body></html>=

--Apple-Mail=_F0FBFB17-07D7-484B-BB29-D25DBC347429--


From nobody Mon Oct 27 12:26:47 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-19.txt
	Pages           : 44
	Date            : 2014-10-27

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-19

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-19


Please note that it may take a couple of minutes from the time of submission
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From nobody Mon Oct 27 12:34:01 2014
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This version clarifies that RTP retransmission has to be supported in =
both session and SSRC multiplexed mode (i.e., non-BUNDLE and BUNDLE =
modes). The aim is to make what I believe is the consensus clear in the =
draft. If this is not the consensus of the group, then we can update =
this draft.

Colin



On 27 Oct 2014, at 19:26, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
>        Authors         : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-19.txt
> 	Pages           : 44
> 	Date            : 2014-10-27
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-19
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-19
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
https://csperkins.org/





From nobody Mon Oct 27 12:40:27 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
	Filename        : draft-ietf-rtcweb-jsep-08.txt
	Pages           : 65
	Date            : 2014-10-27

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


The IETF datatracker status page for this draft is:
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From nobody Mon Oct 27 12:50:24 2014
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On 27 Oct 2014, at 15:19, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>=20
>>> "The support for message interleaving as defined in
>>>                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>=20
>>> First, it is a little unclear what "the support SHOULD be used" =
means.
>>=20
>> My understanding is that SCTP implementations supporting the =
extension will use it.
>> This is negotiated during the setup of the SCTP association.
>=20
> If it's done on SCTP level, why do we need to talk about it in the =
data channel draft?=20
We need to specify somewhere that the extension is used and how to map =
the priorities
of data channels to some protocol mechanism. This is done in this =
document.
>=20
> Is there a reason why it is important to use it for data channels? If =
so, does it apply to data channels in general?=20
As it is written right now: yes.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> Whether messages are interleaved or not depends on the stream =
scheduler. This is a sender side only decision. The receiver has to deal =
with it.
> It is not a MUST, since there are implementations now in use which =
don't support the extension.
>=20
> Best regards
> Michael
>>=20
>> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD =
use interleaving, even if the characteristics of the protocol wouldn't =
require it?
>>=20
>> Regards,
>>=20
>> Christer
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 Oct 2014, at 19:23, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 10/27/2014 07:19 AM, Christer Holmberg wrote:
>> Hi,
>>=20
>>>> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>=20
>>>> "The support for message interleaving as defined in
>>>>                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>=20
>>>> First, it is a little unclear what "the support SHOULD be used" =
means.
>>> My understanding is that SCTP implementations supporting the =
extension will use it.
>>> This is negotiated during the setup of the SCTP association.
>> If it's done on SCTP level, why do we need to talk about it in the =
data channel draft?=20
>>=20
>> Is there a reason why it is important to use it for data channels? If =
so, does it apply to data channels in general?=20
>=20
> NDATA was added in order to avoid head-of-line blocking on the =
transport
> (if I understand this correctly, until this was added, sending a huge
> message would block the delivery of small messages on all channels =
until
> the huge message was fully delivered).
>=20
> Unlike Michael, I see no reason to make this a SHOULD; I think it =
should
> be a MUST, and the older implementations in browsers should just be
> called out as non-conformant.
I personally also prefer a MUST, but the reason I gave was the one which
was used in the discussion resulting in a SHOULD. Since this is also =
brought
up by David, we possibly need to reconsider the issue and change it to a =
MUST.

Best regards
Michael
>=20
> That said, I think that data channels ought to interoperate =
successfully
> with implementations that don't support the extension - but data =
channel
> implementations in WebRTC endpoints should be under a "MUST implement,
> MUST offer" ruleset.
>=20
>=20
>=20
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>>=20
>>=20
>> Whether messages are interleaved or not depends on the stream =
scheduler. This is a sender side only decision. The receiver has to deal =
with it.
>> It is not a MUST, since there are implementations now in use which =
don't support the extension.
>>=20
>> Best regards
>> Michael
>>>=20
>>> Second, does it mean that any data channel protocol (e.g CLUE) =
SHOULD use interleaving, even if the characteristics of the protocol =
wouldn't require it?
>>>=20
>>> Regards,
>>>=20
>>> 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
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 Oct 2014, at 19:41, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>>> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>>=20
>>>>> "The support for message interleaving as defined in
>>>>>                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>>=20
>>>>> First, it is a little unclear what "the support SHOULD be used" =
means.
>>>> My understanding is that SCTP implementations supporting the =
extension will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the =
data channel draft?=20
>>>=20
>>> Is there a reason why it is important to use it for data channels? =
If so, does it apply to data channels in general?=20
>>=20
>> NDATA was added in order to avoid head-of-line blocking on the =
transport (if I understand this correctly, until=20
>> this was added, sending a huge message would block the delivery of =
small messages on all channels until the huge message was fully =
delivered).
>>=20
>> Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older=20
>> implementations in browsers should just be called out as =
non-conformant.
>>=20
>> That said, I think that data channels ought to interoperate =
successfully with implementations that don't support the=20
>> extension - but data channel implementations in WebRTC endpoints =
should be under a "MUST implement, MUST offer" ruleset.
>=20
> First, keep in mind that I am NOT talking about WebRTC endpoints =
(which is one reason we shouldn't call it webrtc data channel to begin =
with, but that's another story...), but ANY type of endpoint that wants =
to use a data channel.=20
>=20
> Second, I am not talking about MUST support, but MUST use/offer.
>=20
> Assume I have a CLUE endpoint, which will use a data channel ONLY to =
transport CLUE messages. CLUE messages aren't that hugeto begin with, =
and there will be no blocking of small messages on other channels - as =
there are no other channels.
>=20
> So, why does the CLUE endpoint need to offer interleaving?
As far as I know we don't have different flavours of data channels. At =
least right now.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
>=20
>> Whether messages are interleaved or not depends on the stream =
scheduler. This is a sender side only decision. The receiver has to deal =
with it.
>> It is not a MUST, since there are implementations now in use which =
don't support the extension.
>>=20
>> Best regards
>> Michael
>>>=20
>>> Second, does it mean that any data channel protocol (e.g CLUE) =
SHOULD use interleaving, even if the characteristics of the protocol =
wouldn't require it?
>>>=20
>>> Regards,
>>>=20
>>> 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
>=20
>=20
> --
> Surveillance is pervasive. Go Dark.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> Hi,
>
>>>>> Within the CLUE WG, we had a discussion regarding the following sta=
tement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>> =20
>>>>> "The support for message interleaving as defined in
>>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>> =20
>>>>> First, it is a little unclear what "the support SHOULD be used" mea=
ns.
>>>> My understanding is that SCTP implementations supporting the extensi=
on will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the da=
ta channel draft?=20
>>>
>>> Is there a reason why it is important to use it for data channels? If=
 so, does it apply to data channels in general?=20
>> NDATA was added in order to avoid head-of-line blocking on the transpo=
rt (if I understand this correctly, until=20
>> this was added, sending a huge message would block the delivery of sma=
ll messages on all channels until the huge message was fully delivered).
>>
>> Unlike Michael, I see no reason to make this a SHOULD; I think it shou=
ld be a MUST, and the older=20
>> implementations in browsers should just be called out as non-conforman=
t.
>>
>> That said, I think that data channels ought to interoperate successful=
ly with implementations that don't support the=20
>> extension - but data channel implementations in WebRTC endpoints shoul=
d be under a "MUST implement, MUST offer" ruleset.
> First, keep in mind that I am NOT talking about WebRTC endpoints (which=
 is one reason we shouldn't call it webrtc data channel to begin with, bu=
t that's another story...), but ANY type of endpoint that wants to use a =
data channel.=20

I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
that it might have been a mistake to float the option of saying "other
things can use these features", and that we should admit only arguments
that bear upon their usage in WebRTC.

I'm not sure I have WG consensus (even rough consensus) on this point,
however.
>
> Second, I am not talking about MUST support, but MUST use/offer.
>
> Assume I have a CLUE endpoint, which will use a data channel ONLY to tr=
ansport CLUE messages. CLUE messages aren't that hugeto begin with, and t=
here will be no blocking of small messages on other channels - as there a=
re no other channels.
>
> So, why does the CLUE endpoint need to offer interleaving?

Flipping the question around - if a programming mistake in the
Javascript implementing a CLUE endpoint in a WebRTC browser happens to
send a multimegabyte-sized object down the (out of order) datachannel,
is it OK to have all the subsequent CLUE messages delivered 30 seconds la=
te?



--=20
Surveillance is pervasive. Go Dark.



From nobody Mon Oct 27 13:06:27 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

I am not suggesting different flavors - I am suggesting to make the usage o=
f interleaving optional. Other SCTP features are also optional (including r=
eliability, ordering etc).

We CAN write/reference guidelines on when it DOES make sense to use interle=
aving.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Michael Tuexen<mailto:Michael.Tuexen@lurchi.franken.de>
Sent: =FD27/=FD10/=FD2014 21:55
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: Harald Alvestrand<mailto:harald@alvestrand.no>; rtcweb@ietf.org<mailto:=
rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving

On 27 Oct 2014, at 19:41, Christer Holmberg <christer.holmberg@ericsson.com=
> wrote:

> Hi,
>
>>>>> Within the CLUE WG, we had a discussion regarding the following state=
ment in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>>
>>>>> "The support for message interleaving as defined in
>>>>>                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>>
>>>>> First, it is a little unclear what "the support SHOULD be used" means=
.
>>>> My understanding is that SCTP implementations supporting the extension=
 will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the data=
 channel draft?
>>>
>>> Is there a reason why it is important to use it for data channels? If s=
o, does it apply to data channels in general?
>>
>> NDATA was added in order to avoid head-of-line blocking on the transport=
 (if I understand this correctly, until
>> this was added, sending a huge message would block the delivery of small=
 messages on all channels until the huge message was fully delivered).
>>
>> Unlike Michael, I see no reason to make this a SHOULD; I think it should=
 be a MUST, and the older
>> implementations in browsers should just be called out as non-conformant.
>>
>> That said, I think that data channels ought to interoperate successfully=
 with implementations that don't support the
>> extension - but data channel implementations in WebRTC endpoints should =
be under a "MUST implement, MUST offer" ruleset.
>
> First, keep in mind that I am NOT talking about WebRTC endpoints (which i=
s one reason we shouldn't call it webrtc data channel to begin with, but th=
at's another story...), but ANY type of endpoint that wants to use a data c=
hannel.
>
> Second, I am not talking about MUST support, but MUST use/offer.
>
> Assume I have a CLUE endpoint, which will use a data channel ONLY to tran=
sport CLUE messages. CLUE messages aren't that hugeto begin with, and there=
 will be no blocking of small messages on other channels - as there are no =
other channels.
>
> So, why does the CLUE endpoint need to offer interleaving?
As far as I know we don't have different flavours of data channels. At leas=
t right now.

Best regards
Michael
>
> Regards,
>
> Christer
>
>
>> Whether messages are interleaved or not depends on the stream scheduler.=
 This is a sender side only decision. The receiver has to deal with it.
>> It is not a MUST, since there are implementations now in use which don't=
 support the extension.
>>
>> Best regards
>> Michael
>>>
>>> Second, does it mean that any data channel protocol (e.g CLUE) SHOULD u=
se interleaving, even if the characteristics of the protocol wouldn't requi=
re it?
>>>
>>> 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
>
>
> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
I am not suggesting different flavors - I am suggesting to make the usage o=
f interleaving optional. Other SCTP features are also optional (including r=
eliability, ordering etc).<br>
<br>
We CAN write/reference guidelines on when it DOES make sense to use interle=
aving.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:Michael.Tuexen@lurchi.franken.de">Michael Tuexen</a></span><br=
>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD27=
/=FD10/=FD2014 21:55</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:harald@alvestrand.no">Harald Alvestrand</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] Meaning of SHOULD support/use interleaving</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 27 Oct 2014, at 19:41, Christer Holmberg &lt;ch=
rister.holmberg@ericsson.com&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt; <br>
&gt;&gt;&gt;&gt;&gt; Within the CLUE WG, we had a discussion regarding the =
following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt=
:<br>
&gt;&gt;&gt;&gt;&gt; <br>
&gt;&gt;&gt;&gt;&gt; &quot;The support for message interleaving as defined =
in<br>
&gt;&gt;&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata] SHOULD be u=
sed.&quot;<br>
&gt;&gt;&gt;&gt;&gt; <br>
&gt;&gt;&gt;&gt;&gt; First, it is a little unclear what &quot;the support S=
HOULD be used&quot; means.<br>
&gt;&gt;&gt;&gt; My understanding is that SCTP implementations supporting t=
he extension will use it.<br>
&gt;&gt;&gt;&gt; This is negotiated during the setup of the SCTP associatio=
n.<br>
&gt;&gt;&gt; If it's done on SCTP level, why do we need to talk about it in=
 the data channel draft?
<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Is there a reason why it is important to use it for data chann=
els? If so, does it apply to data channels in general?
<br>
&gt;&gt; <br>
&gt;&gt; NDATA was added in order to avoid head-of-line blocking on the tra=
nsport (if I understand this correctly, until
<br>
&gt;&gt; this was added, sending a huge message would block the delivery of=
 small messages on all channels until the huge message was fully delivered)=
.<br>
&gt;&gt; <br>
&gt;&gt; Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older
<br>
&gt;&gt; implementations in browsers should just be called out as non-confo=
rmant.<br>
&gt;&gt; <br>
&gt;&gt; That said, I think that data channels ought to interoperate succes=
sfully with implementations that don't support the
<br>
&gt;&gt; extension - but data channel implementations in WebRTC endpoints s=
hould be under a &quot;MUST implement, MUST offer&quot; ruleset.<br>
&gt; <br>
&gt; First, keep in mind that I am NOT talking about WebRTC endpoints (whic=
h is one reason we shouldn't call it webrtc data channel to begin with, but=
 that's another story...), but ANY type of endpoint that wants to use a dat=
a channel.
<br>
&gt; <br>
&gt; Second, I am not talking about MUST support, but MUST use/offer.<br>
&gt; <br>
&gt; Assume I have a CLUE endpoint, which will use a data channel ONLY to t=
ransport CLUE messages. CLUE messages aren't that hugeto begin with, and th=
ere will be no blocking of small messages on other channels - as there are =
no other channels.<br>
&gt; <br>
&gt; So, why does the CLUE endpoint need to offer interleaving?<br>
As far as I know we don't have different flavours of data channels. At leas=
t right now.<br>
<br>
Best regards<br>
Michael<br>
&gt; <br>
&gt; Regards,<br>
&gt; <br>
&gt; Christer<br>
&gt; <br>
&gt; <br>
&gt;&gt; Whether messages are interleaved or not depends on the stream sche=
duler. This is a sender side only decision. The receiver has to deal with i=
t.<br>
&gt;&gt; It is not a MUST, since there are implementations now in use which=
 don't support the extension.<br>
&gt;&gt; <br>
&gt;&gt; Best regards<br>
&gt;&gt; Michael<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Second, does it mean that any data channel protocol (e.g CLUE)=
 SHOULD use interleaving, even if the characteristics of the protocol would=
n't require it?<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Christer<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; rtcweb@ietf.org<br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https=
://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; rtcweb@ietf.org<br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://w=
ww.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; <br>
&gt; <br>
&gt; --<br>
&gt; Surveillance is pervasive. Go Dark.<br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; rtcweb@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; rtcweb@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt; <br>
<br>
</div>
</span></font>
</body>
</html>

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From nobody Mon Oct 27 13:11:06 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se>, <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 Oct 2014, at 21:05, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> I am not suggesting different flavors - I am suggesting to make the =
usage of interleaving optional. Other SCTP features are also optional =
(including reliability, ordering etc).
Where are these things optional? According to
=
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section-6.1
an implementation MUST implement PR-SCTP and the support of the =
FORWARD-TSN
chunk will be negotiated similar to the NDATA chunk. I don't see a =
difference,
especially not if we move to MUST support NDATA.

Best regards
Michael
>=20
> We CAN write/reference guidelines on when it DOES make sense to use =
interleaving.
>=20
> Regards,
>=20
> Christer
>=20
> Sent from my Windows Phone
> From: Michael Tuexen
> Sent: =FD27/=FD10/=FD2014 21:55
> To: Christer Holmberg
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
>=20
> On 27 Oct 2014, at 19:41, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> > Hi,
> >=20
> >>>>> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
> >>>>>=20
> >>>>> "The support for message interleaving as defined in
> >>>>>                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
> >>>>>=20
> >>>>> First, it is a little unclear what "the support SHOULD be used" =
means.
> >>>> My understanding is that SCTP implementations supporting the =
extension will use it.
> >>>> This is negotiated during the setup of the SCTP association.
> >>> If it's done on SCTP level, why do we need to talk about it in the =
data channel draft?=20
> >>>=20
> >>> Is there a reason why it is important to use it for data channels? =
If so, does it apply to data channels in general?=20
> >>=20
> >> NDATA was added in order to avoid head-of-line blocking on the =
transport (if I understand this correctly, until=20
> >> this was added, sending a huge message would block the delivery of =
small messages on all channels until the huge message was fully =
delivered).
> >>=20
> >> Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older=20
> >> implementations in browsers should just be called out as =
non-conformant.
> >>=20
> >> That said, I think that data channels ought to interoperate =
successfully with implementations that don't support the=20
> >> extension - but data channel implementations in WebRTC endpoints =
should be under a "MUST implement, MUST offer" ruleset.
> >=20
> > First, keep in mind that I am NOT talking about WebRTC endpoints =
(which is one reason we shouldn't call it webrtc data channel to begin =
with, but that's another story...), but ANY type of endpoint that wants =
to use a data channel.=20
> >=20
> > Second, I am not talking about MUST support, but MUST use/offer.
> >=20
> > Assume I have a CLUE endpoint, which will use a data channel ONLY to =
transport CLUE messages. CLUE messages aren't that hugeto begin with, =
and there will be no blocking of small messages on other channels - as =
there are no other channels.
> >=20
> > So, why does the CLUE endpoint need to offer interleaving?
> As far as I know we don't have different flavours of data channels. At =
least right now.
>=20
> Best regards
> Michael
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >> Whether messages are interleaved or not depends on the stream =
scheduler. This is a sender side only decision. The receiver has to deal =
with it.
> >> It is not a MUST, since there are implementations now in use which =
don't support the extension.
> >>=20
> >> Best regards
> >> Michael
> >>>=20
> >>> Second, does it mean that any data channel protocol (e.g CLUE) =
SHOULD use interleaving, even if the characteristics of the protocol =
wouldn't require it?
> >>>=20
> >>> Regards,
> >>>=20
> >>> 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
> >=20
> >=20
> > --
> > Surveillance is pervasive. Go Dark.
> >=20
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >=20
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >


From nobody Mon Oct 27 13:11:07 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Harald Alvestrand<mailto:harald@alvestrand.no>
Sent: =FD27/=FD10/=FD2014 22:00
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; rtcweb@ietf.o=
rg<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving

On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> Hi,
>
>>>>> Within the CLUE WG, we had a discussion regarding the following state=
ment in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>>
>>>>> "The support for message interleaving as defined in
>>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>>
>>>>> First, it is a little unclear what "the support SHOULD be used" means=
.
>>>> My understanding is that SCTP implementations supporting the extension=
 will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the data=
 channel draft?
>>>
>>> Is there a reason why it is important to use it for data channels? If s=
o, does it apply to data channels in general?
>> NDATA was added in order to avoid head-of-line blocking on the transport=
 (if I understand this correctly, until
>> this was added, sending a huge message would block the delivery of small=
 messages on all channels until the huge message was fully delivered).
>>
>> Unlike Michael, I see no reason to make this a SHOULD; I think it should=
 be a MUST, and the older
>> implementations in browsers should just be called out as non-conformant.
>>
>> That said, I think that data channels ought to interoperate successfully=
 with implementations that don't support the
>> extension - but data channel implementations in WebRTC endpoints should =
be under a "MUST implement, MUST offer" ruleset.
> First, keep in mind that I am NOT talking about WebRTC endpoints (which i=
s one reason we shouldn't call it webrtc data channel to begin with, but th=
at's another story...), but ANY type of endpoint that wants to use a data c=
hannel.

I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
that it might have been a mistake to float the option of saying "other
things can use these features", and that we should admit only arguments
that bear upon their usage in WebRTC.

I'm not sure I have WG consensus (even rough consensus) on this point,
however.
>
> Second, I am not talking about MUST support, but MUST use/offer.
>
> Assume I have a CLUE endpoint, which will use a data channel ONLY to tran=
sport CLUE messages. CLUE messages aren't that hugeto begin with, and there=
 will be no blocking of small messages on other channels - as there are no =
other channels.
>
> So, why does the CLUE endpoint need to offer interleaving?

Flipping the question around - if a programming mistake in the
Javascript implementing a CLUE endpoint in a WebRTC browser happens to
send a multimegabyte-sized object down the (out of order) datachannel,
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?



--
Surveillance is pervasive. Go Dark.



--_000_7594FB04B1934943A5C02806D1A2204B1D4CEBEDESESSMB209erics_
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<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:harald@alvestrand.no">Harald Alvestrand</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD27=
/=FD10/=FD2014 22:00</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] Meaning of SHOULD support/use interleaving</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 10/27/2014 11:41 AM, Christer Holmberg wrote:<b=
r>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt;&gt;&gt; Within the CLUE WG, we had a discussion regarding the =
following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt=
:<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; &quot;The support for message interleaving as defined =
in<br>
&gt;&gt;&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata] SHOUL=
D be used.&quot;<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; First, it is a little unclear what &quot;the support S=
HOULD be used&quot; means.<br>
&gt;&gt;&gt;&gt; My understanding is that SCTP implementations supporting t=
he extension will use it.<br>
&gt;&gt;&gt;&gt; This is negotiated during the setup of the SCTP associatio=
n.<br>
&gt;&gt;&gt; If it's done on SCTP level, why do we need to talk about it in=
 the data channel draft?
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Is there a reason why it is important to use it for data chann=
els? If so, does it apply to data channels in general?
<br>
&gt;&gt; NDATA was added in order to avoid head-of-line blocking on the tra=
nsport (if I understand this correctly, until
<br>
&gt;&gt; this was added, sending a huge message would block the delivery of=
 small messages on all channels until the huge message was fully delivered)=
.<br>
&gt;&gt;<br>
&gt;&gt; Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older
<br>
&gt;&gt; implementations in browsers should just be called out as non-confo=
rmant.<br>
&gt;&gt;<br>
&gt;&gt; That said, I think that data channels ought to interoperate succes=
sfully with implementations that don't support the
<br>
&gt;&gt; extension - but data channel implementations in WebRTC endpoints s=
hould be under a &quot;MUST implement, MUST offer&quot; ruleset.<br>
&gt; First, keep in mind that I am NOT talking about WebRTC endpoints (whic=
h is one reason we shouldn't call it webrtc data channel to begin with, but=
 that's another story...), but ANY type of endpoint that wants to use a dat=
a channel.
<br>
<br>
I'm a bit self-centric on behalf of WebRTC, but I feel the other way -<br>
that it might have been a mistake to float the option of saying &quot;other=
<br>
things can use these features&quot;, and that we should admit only argument=
s<br>
that bear upon their usage in WebRTC.<br>
<br>
I'm not sure I have WG consensus (even rough consensus) on this point,<br>
however.<br>
&gt;<br>
&gt; Second, I am not talking about MUST support, but MUST use/offer.<br>
&gt;<br>
&gt; Assume I have a CLUE endpoint, which will use a data channel ONLY to t=
ransport CLUE messages. CLUE messages aren't that hugeto begin with, and th=
ere will be no blocking of small messages on other channels - as there are =
no other channels.<br>
&gt;<br>
&gt; So, why does the CLUE endpoint need to offer interleaving?<br>
<br>
Flipping the question around - if a programming mistake in the<br>
Javascript implementing a CLUE endpoint in a WebRTC browser happens to<br>
send a multimegabyte-sized object down the (out of order) datachannel,<br>
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?<br>
<br>
<br>
<br>
-- <br>
Surveillance is pervasive. Go Dark.<br>
<br>
<br>
</div>
</span></font>
</body>
</html>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Date: Mon, 27 Oct 2014 20:31:26 +0000
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+1 to make N-DATA optional.

Data channel provides a robust generic peer to peer communication channel. =
So, one can see that data channel may be used under non-webrtc contexts (li=
ke as in CLUE) as a general transport or as a replacement of native to SCTP=
 to traverse firewalls. IMHO, majority of these use cases:
a. May just involve a single data channel (with possible multiplexing on to=
p of it)
b. And may not use the DCEP either (both ends could assume same data channe=
l stream id, this is similar to SCTP PPIDs).
We already made DCEP(b) as optional.

If an app is using just a single data channel then there is no real benefit=
 in using N-DATA. Then why not make it optional? Also, as already pointed o=
ut, making it optional won=92t break interop as it is still negotiated duri=
ng SCTP setup time anyway.

I would like to point out that there will be some webrtc applications which=
 might just use one webrtc data channel for simplicity (and may use some mu=
ltiplexing of different types of msgs/objects on top of it) or one is just =
sufficient as the app does not send/recv huge chunks of data. So, N-DATA is=
n=92t needed for such webrtc clients either. Then why add burden to such we=
brtc implementations!?

BR
Raju


From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: Monday, October 27, 2014 3:11 PM
To: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving

Hi,

We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Harald Alvestrand<mailto:harald@alvestrand.no>
Sent: =FD27/=FD10/=FD2014 22:00
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; rtcweb@ietf.o=
rg<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> Hi,
>
>>>>> Within the CLUE WG, we had a discussion regarding the following state=
ment in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>>
>>>>> "The support for message interleaving as defined in
>>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>>
>>>>> First, it is a little unclear what "the support SHOULD be used" means=
.
>>>> My understanding is that SCTP implementations supporting the extension=
 will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the data=
 channel draft?
>>>
>>> Is there a reason why it is important to use it for data channels? If s=
o, does it apply to data channels in general?
>> NDATA was added in order to avoid head-of-line blocking on the transport=
 (if I understand this correctly, until
>> this was added, sending a huge message would block the delivery of small=
 messages on all channels until the huge message was fully delivered).
>>
>> Unlike Michael, I see no reason to make this a SHOULD; I think it should=
 be a MUST, and the older
>> implementations in browsers should just be called out as non-conformant.
>>
>> That said, I think that data channels ought to interoperate successfully=
 with implementations that don't support the
>> extension - but data channel implementations in WebRTC endpoints should =
be under a "MUST implement, MUST offer" ruleset.
> First, keep in mind that I am NOT talking about WebRTC endpoints (which i=
s one reason we shouldn't call it webrtc data channel to begin with, but th=
at's another story...), but ANY type of endpoint that wants to use a data c=
hannel.

I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
that it might have been a mistake to float the option of saying "other
things can use these features", and that we should admit only arguments
that bear upon their usage in WebRTC.

I'm not sure I have WG consensus (even rough consensus) on this point,
however.
>
> Second, I am not talking about MUST support, but MUST use/offer.
>
> Assume I have a CLUE endpoint, which will use a data channel ONLY to tran=
sport CLUE messages. CLUE messages aren't that hugeto begin with, and there=
 will be no blocking of small messages on other channels - as there are no =
other channels.
>
> So, why does the CLUE endpoint need to offer interleaving?

Flipping the question around - if a programming mistake in the
Javascript implementing a CLUE endpoint in a WebRTC browser happens to
send a multimegabyte-sized object down the (out of order) datachannel,
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?



--
Surveillance is pervasive. Go Dark.


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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">&#43;1 to make N-DATA opt=
ional.<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">Data channel provides a r=
obust generic peer to peer communication channel. So, one can see that data=
 channel may be used under non-webrtc contexts (like as
 in CLUE) as a general transport or as a replacement of native to SCTP to t=
raverse firewalls. IMHO, majority of these use cases:<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">a. May just involve a sin=
gle data channel (with possible multiplexing on top of it)
<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">b. And may not use the DC=
EP either (both ends could assume same data channel stream id, this is simi=
lar to SCTP PPIDs).
<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">We already made DCEP(b) a=
s optional.<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 an app is using just a=
 single data channel then there is no real benefit in using N-DATA. Then wh=
y not make it optional? Also, as already pointed out, making
 it optional won=92t break interop as it is still negotiated during SCTP se=
tup time anyway.<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 would like to point out=
 that there will be some webrtc applications which might just use one webrt=
c data channel for simplicity (and may use some multiplexing
 of different types of msgs/objects on top of it) or one is just sufficient=
 as the app does not send/recv huge chunks of data. So, N-DATA isn=92t need=
ed for such webrtc clients either. Then why add burden to such webrtc imple=
mentations!?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<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">Raju<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>
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<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 [=
mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> Monday, October 27, 2014 3:11 PM<br>
<b>To:</b> Harald Alvestrand; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Meaning of SHOULD support/use interleaving<o:p=
></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></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;">Hi,<br>
<br>
We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone<o:p></o:p></span></p>
</div>
</div>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><span style=3D"fon=
t-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;"><a href=3D"mailto:harald@alvestrand.no">Harald Alve=
strand</a></span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">Sent: </span>
</b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;">=FD27/=FD10/=FD2014 22:00</span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">To: </span></b><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;"><a href=3D"mailto:christer.holm=
berg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">Subject: </span>
</b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;">Re: [rtcweb] Meaning of SHOULD support/use interleaving</s=
pan><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt">On 10/27/2014 11:41 AM, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt;&gt;&gt; Within the CLUE WG, we had a discussion regarding the =
following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt=
:<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; &quot;The support for message interleaving as defined =
in<br>
&gt;&gt;&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata] SHOUL=
D be used.&quot;<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; First, it is a little unclear what &quot;the support S=
HOULD be used&quot; means.<br>
&gt;&gt;&gt;&gt; My understanding is that SCTP implementations supporting t=
he extension will use it.<br>
&gt;&gt;&gt;&gt; This is negotiated during the setup of the SCTP associatio=
n.<br>
&gt;&gt;&gt; If it's done on SCTP level, why do we need to talk about it in=
 the data channel draft?
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Is there a reason why it is important to use it for data chann=
els? If so, does it apply to data channels in general?
<br>
&gt;&gt; NDATA was added in order to avoid head-of-line blocking on the tra=
nsport (if I understand this correctly, until
<br>
&gt;&gt; this was added, sending a huge message would block the delivery of=
 small messages on all channels until the huge message was fully delivered)=
.<br>
&gt;&gt;<br>
&gt;&gt; Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older
<br>
&gt;&gt; implementations in browsers should just be called out as non-confo=
rmant.<br>
&gt;&gt;<br>
&gt;&gt; That said, I think that data channels ought to interoperate succes=
sfully with implementations that don't support the
<br>
&gt;&gt; extension - but data channel implementations in WebRTC endpoints s=
hould be under a &quot;MUST implement, MUST offer&quot; ruleset.<br>
&gt; First, keep in mind that I am NOT talking about WebRTC endpoints (whic=
h is one reason we shouldn't call it webrtc data channel to begin with, but=
 that's another story...), but ANY type of endpoint that wants to use a dat=
a channel.
<br>
<br>
I'm a bit self-centric on behalf of WebRTC, but I feel the other way -<br>
that it might have been a mistake to float the option of saying &quot;other=
<br>
things can use these features&quot;, and that we should admit only argument=
s<br>
that bear upon their usage in WebRTC.<br>
<br>
I'm not sure I have WG consensus (even rough consensus) on this point,<br>
however.<br>
&gt;<br>
&gt; Second, I am not talking about MUST support, but MUST use/offer.<br>
&gt;<br>
&gt; Assume I have a CLUE endpoint, which will use a data channel ONLY to t=
ransport CLUE messages. CLUE messages aren't that hugeto begin with, and th=
ere will be no blocking of small messages on other channels - as there are =
no other channels.<br>
&gt;<br>
&gt; So, why does the CLUE endpoint need to offer interleaving?<br>
<br>
Flipping the question around - if a programming mistake in the<br>
Javascript implementing a CLUE endpoint in a WebRTC browser happens to<br>
send a multimegabyte-sized object down the (out of order) datachannel,<br>
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?<br>
<br>
<br>
<br>
-- <br>
Surveillance is pervasive. Go Dark.<br>
<br>
<o:p></o:p></span></p>
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From nobody Mon Oct 27 13:35:32 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Video Processing and Codec Requirements
        Author          : Adam Roach
	Filename        : draft-ietf-rtcweb-video-01.txt
	Pages           : 9
	Date            : 2014-10-27

Abstract:
   This specification provides the requirements and considerations for
   WebRTC applications to send and receive video across a network.  It
   specifies the video processing that is required, as well as video
   codecs and their parameters.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-video-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-video-01


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

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


From nobody Mon Oct 27 13:49:11 2014
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Based on feedback from (and since) Toronto, I've put out a new version 
of the video draft:
https://tools.ietf.org/html/draft-ietf-rtcweb-video-01

Deltas:
https://tools.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-video-01.txt

Some of this is forward-leaning (e.g. the colorspace) so as to provoke a 
disucssion. As such, I will indicate ahead of time that some of this 
text still needs discussion and agreement within the working group. The 
areas where I have done this are clearly marked with "TODO."

/a


From nobody Mon Oct 27 13:58:29 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Video Processing and Codec Requirements
        Author          : Adam Roach
	Filename        : draft-ietf-rtcweb-video-02.txt
	Pages           : 9
	Date            : 2014-10-27

Abstract:
   This specification provides the requirements and considerations for
   WebRTC applications to send and receive video across a network.  It
   specifies the video processing that is required, as well as video
   codecs and their parameters.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-video-02

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


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

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


From nobody Mon Oct 27 13:59:59 2014
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...and pretty much right after I sent this out, someone pointed out to 
me that I'd mistakenly put "H.264" in the VP8 section when I was talking 
about VP8. So now there's a -02 out there. It simply fixes this rather 
confusing typo.

/a

On 10/27/14 15:48, Adam Roach wrote:
> Based on feedback from (and since) Toronto, I've put out a new version 
> of the video draft:
> https://tools.ietf.org/html/draft-ietf-rtcweb-video-01
>
> Deltas:
> https://tools.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-video-01.txt
>
> Some of this is forward-leaning (e.g. the colorspace) so as to provoke 
> a disucssion. As such, I will indicate ahead of time that some of this 
> text still needs discussion and agreement within the working group. 
> The areas where I have done this are clearly marked with "TODO."
>
> /a
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Oct 27 14:20:22 2014
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 Oct 2014, at 21:31, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:

> +1 to make N-DATA optional.
> =20
> Data channel provides a robust generic peer to peer communication =
channel. So, one can see that data channel may be used under non-webrtc =
contexts (like as in CLUE) as a general transport or as a replacement of =
native to SCTP to traverse firewalls. IMHO, majority of these use cases:
> a. May just involve a single data channel (with possible multiplexing =
on top of it)
> b. And may not use the DCEP either (both ends could assume same data =
channel stream id, this is similar to SCTP PPIDs).
> We already made DCEP(b) as optional.
> =20
> If an app is using just a single data channel then there is no real =
benefit in using N-DATA. Then why not make it optional? Also, as already =
pointed out, making it optional won=92t break interop as it is still =
negotiated during SCTP setup time anyway.
> =20
> I would like to point out that there will be some webrtc applications =
which might just use one webrtc data channel for simplicity (and may use =
some multiplexing of different types of msgs/objects on top of it) or =
one is just sufficient as the app does not send/recv huge chunks of =
data. So, N-DATA isn=92t needed for such webrtc clients either. Then why =
add burden to such webrtc implementations!?
Are you implementing this? I would say that adding N-DATA to an SCTP =
implementation with support
for Stream Reconfiguration and PR-SCTP does not increase the complexity =
substantially...

Best regards
Michael
> =20
> BR
> Raju
> =20
> =20
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer =
Holmberg
> Sent: Monday, October 27, 2014 3:11 PM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
> =20
> Hi,
>=20
> We shall not add burden just because there are bad programmers out =
there - they will anyway always manage to use our specs in a wrong way =
:)
>=20
> Regards,
>=20
> Christer
>=20
> Sent from my Windows Phone
> From: Harald Alvestrand
> Sent: =FD27/=FD10/=FD2014 22:00
> To: Christer Holmberg; rtcweb@ietf.org
> Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
>=20
> On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> > Hi,
> >
> >>>>> Within the CLUE WG, we had a discussion regarding the following =
statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
> >>>>> =20
> >>>>> "The support for message interleaving as defined in
> >>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
> >>>>> =20
> >>>>> First, it is a little unclear what "the support SHOULD be used" =
means.
> >>>> My understanding is that SCTP implementations supporting the =
extension will use it.
> >>>> This is negotiated during the setup of the SCTP association.
> >>> If it's done on SCTP level, why do we need to talk about it in the =
data channel draft?=20
> >>>
> >>> Is there a reason why it is important to use it for data channels? =
If so, does it apply to data channels in general?=20
> >> NDATA was added in order to avoid head-of-line blocking on the =
transport (if I understand this correctly, until=20
> >> this was added, sending a huge message would block the delivery of =
small messages on all channels until the huge message was fully =
delivered).
> >>
> >> Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older=20
> >> implementations in browsers should just be called out as =
non-conformant.
> >>
> >> That said, I think that data channels ought to interoperate =
successfully with implementations that don't support the=20
> >> extension - but data channel implementations in WebRTC endpoints =
should be under a "MUST implement, MUST offer" ruleset.
> > First, keep in mind that I am NOT talking about WebRTC endpoints =
(which is one reason we shouldn't call it webrtc data channel to begin =
with, but that's another story...), but ANY type of endpoint that wants =
to use a data channel.=20
>=20
> I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
> that it might have been a mistake to float the option of saying "other
> things can use these features", and that we should admit only =
arguments
> that bear upon their usage in WebRTC.
>=20
> I'm not sure I have WG consensus (even rough consensus) on this point,
> however.
> >
> > Second, I am not talking about MUST support, but MUST use/offer.
> >
> > Assume I have a CLUE endpoint, which will use a data channel ONLY to =
transport CLUE messages. CLUE messages aren't that hugeto begin with, =
and there will be no blocking of small messages on other channels - as =
there are no other channels.
> >
> > So, why does the CLUE endpoint need to offer interleaving?
>=20
> Flipping the question around - if a programming mistake in the
> Javascript implementing a CLUE endpoint in a WebRTC browser happens to
> send a multimegabyte-sized object down the (out of order) datachannel,
> is it OK to have all the subsequent CLUE messages delivered 30 seconds =
late?
>=20
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se>, <544EA473.6040700@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CEBED@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17828E5F2ED0@US70UWXCHMBA02.zam.alcatel-lucent.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju) wrote:
>
> +1 to make N-DATA optional.
>
> =20
>
> Data channel provides a robust generic peer to peer communication
> channel. So, one can see that data channel may be used under
> non-webrtc contexts (like as in CLUE) as a general transport or as a
> replacement of native to SCTP to traverse firewalls. IMHO, majority of
> these use cases:
>
> a. May just involve a single data channel (with possible multiplexing
> on top of it)
>
> b. And may not use the DCEP either (both ends could assume same data
> channel stream id, this is similar to SCTP PPIDs).
>
> We already made DCEP(b) as optional.
>
> =20
>
> If an app is using just a single data channel then there is no real
> benefit in using N-DATA. Then why not make it optional? Also, as
> already pointed out, making it optional won=92t break interop as it is
> still negotiated during SCTP setup time anyway.
>

Negotiation means that transport initiator gets to choose whether to use
it or not.
This means that a responder has no choice in the matter.

In the situation where the responder knows better than the initiator
what the usage is going to be, this is problematic.

> =20
>
> I would like to point out that there will be some webrtc applications
> which might just use one webrtc data channel for simplicity (and may
> use some multiplexing of different types of msgs/objects on top of it)
> or one is just sufficient as the app does not send/recv huge chunks of
> data. So, N-DATA isn=92t needed for such webrtc clients either. Then wh=
y
> add burden to such webrtc implementations!?
>

You're confusing the webrtc application (what runs in the browser) with
the webrtc implementation (the browser itself). Optional NDATA makes the
browser *more* complex, since it has to expose API surface to let the
application choose whether or not to offer NDATA, and it has to have
code to decide whether or not to offer NDATA.

If you're arguing that we should make NDATA optional to implement for
browsers ..... that is an issue that I thought we had already settled.

> =20
>
> BR
>
> Raju
>
> =20
>
> =20
>
> *From:*rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Christer
> Holmberg
> *Sent:* Monday, October 27, 2014 3:11 PM
> *To:* Harald Alvestrand; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Meaning of SHOULD support/use interleaving
>
> =20
>
> Hi,
>
> We shall not add burden just because there are bad programmers out
> there - they will anyway always manage to use our specs in a wrong way =
:)
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
>
> -----------------------------------------------------------------------=
-
>
> *From: *Harald Alvestrand <mailto:harald@alvestrand.no>
> *Sent: *=FD27/=FD10/=FD2014 22:00
> *To: *Christer Holmberg <mailto:christer.holmberg@ericsson.com>;
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject: *Re: [rtcweb] Meaning of SHOULD support/use interleaving
>
> On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> > Hi,
> >
> >>>>> Within the CLUE WG, we had a discussion regarding the following
> statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
> >>>>>=20
> >>>>> "The support for message interleaving as defined in
> >>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
> >>>>>=20
> >>>>> First, it is a little unclear what "the support SHOULD be used"
> means.
> >>>> My understanding is that SCTP implementations supporting the
> extension will use it.
> >>>> This is negotiated during the setup of the SCTP association.
> >>> If it's done on SCTP level, why do we need to talk about it in the
> data channel draft?
> >>>
> >>> Is there a reason why it is important to use it for data channels?
> If so, does it apply to data channels in general?
> >> NDATA was added in order to avoid head-of-line blocking on the
> transport (if I understand this correctly, until
> >> this was added, sending a huge message would block the delivery of
> small messages on all channels until the huge message was fully
> delivered).
> >>
> >> Unlike Michael, I see no reason to make this a SHOULD; I think it
> should be a MUST, and the older
> >> implementations in browsers should just be called out as
> non-conformant.
> >>
> >> That said, I think that data channels ought to interoperate
> successfully with implementations that don't support the
> >> extension - but data channel implementations in WebRTC endpoints
> should be under a "MUST implement, MUST offer" ruleset.
> > First, keep in mind that I am NOT talking about WebRTC endpoints
> (which is one reason we shouldn't call it webrtc data channel to begin
> with, but that's another story...), but ANY type of endpoint that
> wants to use a data channel.
>
> I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
> that it might have been a mistake to float the option of saying "other
> things can use these features", and that we should admit only arguments=

> that bear upon their usage in WebRTC.
>
> I'm not sure I have WG consensus (even rough consensus) on this point,
> however.
> >
> > Second, I am not talking about MUST support, but MUST use/offer.
> >
> > Assume I have a CLUE endpoint, which will use a data channel ONLY to
> transport CLUE messages. CLUE messages aren't that hugeto begin with,
> and there will be no blocking of small messages on other channels - as
> there are no other channels.
> >
> > So, why does the CLUE endpoint need to offer interleaving?
>
> Flipping the question around - if a programming mistake in the
> Javascript implementing a CLUE endpoint in a WebRTC browser happens to
> send a multimegabyte-sized object down the (out of order) datachannel,
> is it OK to have all the subsequent CLUE messages delivered 30 seconds
> late?
>
>
>
> --=20
> Surveillance is pervasive. Go Dark.
>


--=20
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 10/27/2014 01:31 PM, Makaraju,
      Maridi Raju (Raju) wrote:<br>
    </div>
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">+1
            to make N-DATA optional.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Data
            channel provides a robust generic peer to peer communication
            channel. So, one can see that data channel may be used under
            non-webrtc contexts (like as in CLUE) as a general transport
            or as a replacement of native to SCTP to traverse firewalls.
            IMHO, majority of these use cases:<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">a.
            May just involve a single data channel (with possible
            multiplexing on top of it)
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">b.
            And may not use the DCEP either (both ends could assume same
            data channel stream id, this is similar to SCTP PPIDs).
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">We
            already made DCEP(b) as optional.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If
            an app is using just a single data channel then there is no
            real benefit in using N-DATA. Then why not make it optional?
            Also, as already pointed out, making it optional won’t break
            interop as it is still negotiated during SCTP setup time
            anyway.</span></p>
      </div>
    </blockquote>
    <br>
    Negotiation means that transport initiator gets to choose whether to
    use it or not.<br>
    This means that a responder has no choice in the matter.<br>
    <br>
    In the situation where the responder knows better than the initiator
    what the usage is going to be, this is problematic.<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5F2ED0@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            would like to point out that there will be some webrtc
            applications which might just use one webrtc data channel
            for simplicity (and may use some multiplexing of different
            types of msgs/objects on top of it) or one is just
            sufficient as the app does not send/recv huge chunks of
            data. So, N-DATA isn’t needed for such webrtc clients
            either. Then why add burden to such webrtc implementations!?</span></p>
      </div>
    </blockquote>
    <br>
    You're confusing the webrtc application (what runs in the browser)
    with the webrtc implementation (the browser itself). Optional NDATA
    makes the browser *more* complex, since it has to expose API surface
    to let the application choose whether or not to offer NDATA, and it
    has to have code to decide whether or not to offer NDATA.<br>
    <br>
    If you're arguing that we should make NDATA optional to implement
    for browsers ..... that is an issue that I thought we had already
    settled.<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5F2ED0@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <div>
            <div style="border:none;border-top:solid #B5C4DF
              1.0pt;padding:3.0pt 0in 0in 0in">
              <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                  rtcweb [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                  <b>On Behalf Of </b>Christer Holmberg<br>
                  <b>Sent:</b> Monday, October 27, 2014 3:11 PM<br>
                  <b>To:</b> Harald Alvestrand; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> Re: [rtcweb] Meaning of SHOULD
                  support/use interleaving<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p> </o:p></p>
          <div>
            <div>
              <div>
                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Hi,<br>
                    <br>
                    We shall not add burden just because there are bad
                    programmers out there - they will anyway always
                    manage to use our specs in a wrong way :)<br>
                    <br>
                    Regards,<br>
                    <br>
                    Christer<br>
                    <br>
                    Sent from my Windows Phone<o:p></o:p></span></p>
              </div>
            </div>
            <div>
              <div class="MsoNormal" style="text-align:center"
                align="center">
                <hr align="center" size="2" width="100%">
              </div>
              <p class="MsoNormal" style="margin-bottom:12.0pt"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:
                  </span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><a
                    moz-do-not-send="true"
                    href="mailto:harald@alvestrand.no">Harald Alvestrand</a></span><br>
                <b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Sent:
                  </span>
                </b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">ý27/ý10/ý2014
                  22:00</span><br>
                <b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">To:
                  </span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><a
                    moz-do-not-send="true"
                    href="mailto:christer.holmberg@ericsson.com">Christer
                    Holmberg</a>;
                  <a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
                <b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Subject:
                  </span>
                </b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Re:
                  [rtcweb] Meaning of SHOULD support/use interleaving</span><o:p></o:p></p>
            </div>
          </div>
          <div>
            <p class="MsoNormal" style="margin-bottom:12.0pt"><span
                style="font-size:10.0pt">On 10/27/2014 11:41 AM,
                Christer Holmberg wrote:<br>
                &gt; Hi,<br>
                &gt;<br>
                &gt;&gt;&gt;&gt;&gt; Within the CLUE WG, we had a
                discussion regarding the following statement in section
                6.1 of draft-ietf-rtcweb-data-channel-12.txt:<br>
                &gt;&gt;&gt;&gt;&gt;  <br>
                &gt;&gt;&gt;&gt;&gt; "The support for message
                interleaving as defined in<br>
                &gt;&gt;&gt;&gt;&gt;                
                [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."<br>
                &gt;&gt;&gt;&gt;&gt;  <br>
                &gt;&gt;&gt;&gt;&gt; First, it is a little unclear what
                "the support SHOULD be used" means.<br>
                &gt;&gt;&gt;&gt; My understanding is that SCTP
                implementations supporting the extension will use it.<br>
                &gt;&gt;&gt;&gt; This is negotiated during the setup of
                the SCTP association.<br>
                &gt;&gt;&gt; If it's done on SCTP level, why do we need
                to talk about it in the data channel draft?
                <br>
                &gt;&gt;&gt;<br>
                &gt;&gt;&gt; Is there a reason why it is important to
                use it for data channels? If so, does it apply to data
                channels in general?
                <br>
                &gt;&gt; NDATA was added in order to avoid head-of-line
                blocking on the transport (if I understand this
                correctly, until
                <br>
                &gt;&gt; this was added, sending a huge message would
                block the delivery of small messages on all channels
                until the huge message was fully delivered).<br>
                &gt;&gt;<br>
                &gt;&gt; Unlike Michael, I see no reason to make this a
                SHOULD; I think it should be a MUST, and the older
                <br>
                &gt;&gt; implementations in browsers should just be
                called out as non-conformant.<br>
                &gt;&gt;<br>
                &gt;&gt; That said, I think that data channels ought to
                interoperate successfully with implementations that
                don't support the
                <br>
                &gt;&gt; extension - but data channel implementations in
                WebRTC endpoints should be under a "MUST implement, MUST
                offer" ruleset.<br>
                &gt; First, keep in mind that I am NOT talking about
                WebRTC endpoints (which is one reason we shouldn't call
                it webrtc data channel to begin with, but that's another
                story...), but ANY type of endpoint that wants to use a
                data channel.
                <br>
                <br>
                I'm a bit self-centric on behalf of WebRTC, but I feel
                the other way -<br>
                that it might have been a mistake to float the option of
                saying "other<br>
                things can use these features", and that we should admit
                only arguments<br>
                that bear upon their usage in WebRTC.<br>
                <br>
                I'm not sure I have WG consensus (even rough consensus)
                on this point,<br>
                however.<br>
                &gt;<br>
                &gt; Second, I am not talking about MUST support, but
                MUST use/offer.<br>
                &gt;<br>
                &gt; Assume I have a CLUE endpoint, which will use a
                data channel ONLY to transport CLUE messages. CLUE
                messages aren't that hugeto begin with, and there will
                be no blocking of small messages on other channels - as
                there are no other channels.<br>
                &gt;<br>
                &gt; So, why does the CLUE endpoint need to offer
                interleaving?<br>
                <br>
                Flipping the question around - if a programming mistake
                in the<br>
                Javascript implementing a CLUE endpoint in a WebRTC
                browser happens to<br>
                send a multimegabyte-sized object down the (out of
                order) datachannel,<br>
                is it OK to have all the subsequent CLUE messages
                delivered 30 seconds late?<br>
                <br>
                <br>
                <br>
                -- <br>
                Surveillance is pervasive. Go Dark.<br>
                <br>
                <o:p></o:p></span></p>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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From nobody Mon Oct 27 16:57:52 2014
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Date: Mon, 27 Oct 2014 16:57:02 -0700
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--089e01229176ab6cdd0506704ac2
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An initial version of FEC requirements for WebRTC. I'd like to get some
agenda time to discuss this in Honolulu.

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Oct 27, 2014 at 4:48 PM
Subject: New Version Notification for draft-uberti-rtcweb-fec-00.txt
To: Justin Uberti <justin@uberti.name>



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

Name:           draft-uberti-rtcweb-fec
Revision:       00
Title:          WebRTC Forward Error Correction Requirements
Document date:  2014-10-27
Group:          Individual Submission
Pages:          7
URL:
http://www.ietf.org/internet-drafts/draft-uberti-rtcweb-fec-00.txt
Status:         https://datatracker.ietf.org/doc/draft-uberti-rtcweb-fec/
Htmlized:       http://tools.ietf.org/html/draft-uberti-rtcweb-fec-00


Abstract:
   This document makes recommendations for how Forward Error Correction
   (FEC) should be used by WebRTC applications.




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

The IETF Secretariat

--089e01229176ab6cdd0506704ac2
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<div dir=3D"ltr"><div class=3D"gmail_quote">An initial version of FEC requi=
rements for WebRTC. I&#39;d like to get some agenda time to discuss this in=
 Honolulu.<br><div dir=3D"ltr"><br><div class=3D"gmail_quote">---------- Fo=
rwarded message ----------<br>From: <b class=3D"gmail_sendername"></b> <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_bl=
ank">internet-drafts@ietf.org</a>&gt;</span><br>Date: Mon, Oct 27, 2014 at =
4:48 PM<br>Subject: New Version Notification for draft-uberti-rtcweb-fec-00=
.txt<br>To: Justin Uberti &lt;<a href=3D"mailto:justin@uberti.name" target=
=3D"_blank">justin@uberti.name</a>&gt;<br><br><br><br>
A new version of I-D, draft-uberti-rtcweb-fec-00.txt<br>
has been successfully submitted by Justin Uberti and posted to the<br>
IETF repository.<br>
<br>
Name:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-uberti-rtcweb-fec<br>
Revision:=C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 WebRTC Forward Error Correction Re=
quirements<br>
Document date:=C2=A0 2014-10-27<br>
Group:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Pages:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 7<br>
URL:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.or=
g/internet-drafts/draft-uberti-rtcweb-fec-00.txt" target=3D"_blank">http://=
www.ietf.org/internet-drafts/draft-uberti-rtcweb-fec-00.txt</a><br>
Status:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://datatracker.iet=
f.org/doc/draft-uberti-rtcweb-fec/" target=3D"_blank">https://datatracker.i=
etf.org/doc/draft-uberti-rtcweb-fec/</a><br>
Htmlized:=C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/d=
raft-uberti-rtcweb-fec-00" target=3D"_blank">http://tools.ietf.org/html/dra=
ft-uberti-rtcweb-fec-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document makes recommendations for how Forward Error Corr=
ection<br>
=C2=A0 =C2=A0(FEC) should be used by WebRTC applications.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>
</div><br></div>

--089e01229176ab6cdd0506704ac2--


From nobody Mon Oct 27 17:15:45 2014
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David Benham (dbenham) wrote:
>     There remains a great deal of confusion in the industry about the
>     state of patent litigation and IPR disclosures around VP8. To

Thanks for compiling this report, David. Hopefully this can avoid 
wasting time on arguments about what should be publicly documented 
facts. I just wanted to comment that your use of the acronym "FRAND" to 
mean "Free under Reasonable And Non-Discriminatory" terms may lead to 
some confusion, as there is already an existing expansion in common use, 
namely "Fair, Reasonable, And Non-Discriminatory". The term doesn't 
appear in the Duane Morris report you cite (except in a citation, where 
I believe it is intended to refer to the latter expansion, not the former).

I've been looking for a good term to use for technology which is covered 
by patents, but for which royalty-free licenses with open-source 
compatible terms exist. "Royalty-Free" doesn't quite fit the bill, as 
there are examples of royalty-free licenses which would preclude an 
open-source implementation [1]. "Patent-free" is clearly wrong and 
"patent-unencumbered", while better, might still be misleading (however, 
it's the best I've been able to think of).

If someone can think of a better way to describe this class of licenses, 
I am all ears. RFLWOSCT doesn't quite roll off the tongue.

[1] Clause 2.5 (among others) of 
http://www.polycom.com/content/dam/polycom/www/documents/program-details/siren-license-agreement.pdf 
being one example.


From nobody Mon Oct 27 17:18:33 2014
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Date: Mon, 27 Oct 2014 17:18:07 -0700
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Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On Mon, Oct 27, 2014 at 12:16 PM, Colin Perkins <csp@csperkins.org> wrote:

> On 27 Oct 2014, at 17:17, Sergio Garcia Murillo <
> sergio.garcia.murillo@gmail.com> wrote:
>
>  On 27/10/2014 18:10, Colin Perkins wrote:
>
> On 27 Oct 2014, at 16:51, Sergio Garcia Murillo <
> sergio.garcia.murillo@gmail.com> wrote:
>
>  On 27/10/2014 17:45, Colin Perkins wrote:
>
> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo <
> sergio.garcia.murillo@gmail.com> wrote:
>
>  On 27/10/2014 17:36, Colin Perkins wrote:
>
>  On 21 Oct 2014, at 19:58, Sergio Garcia Murillo <
> sergio.garcia.murillo@gmail.com> wrote:
>
> Not sure if it is done on pourpose, but according to the RTP usage draft,
> it may seem that full RFC 4588 is mandated at the recevier side:
>
>     Receivers are REQUIRED to implement support for RTP retransmission
>     packets [RFC4588 <https://tools.ietf.org/html/rfc4588>].
>
>
> That would include both modes, session and ssrc multiplexing. Given the
> extensive usage of bundle and current implementations, session multiplexi=
ng
> support doesn't make much sense.
>
> Should we drop it, and state that only ssrc-multiplexing shall be
> supported at the receiving end?
>
>
>  I don=E2=80=99t see any advantage to doing so, given that support for no=
n-BUNDLE
> sessions is REQUIRED. You need to implement the signalling needed for
> session-multiplexing of retransmission packet anyway, so disallowing it
> buys you nothing.
>
>
> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, what I
> don't see is how to do session multiplexing with BUNDLE sessions.
>
>
>  You can=E2=80=99t do session multiplexing for BUNDLE sessions; by defini=
tion
> they use SSRC multiplexing. You could do non-BUNDLE sessions, with
> retransmission sent on a separate RTP session though.
>
>   So, you are saying exactly the same than me. SSRC multiplexing supports
> both BUNDLE and NON-BUNDLE. So, why require support for session
> multiplexing at all? As a developer, I don't see why I would have to
> implement something that would be rarely used and provide no extra benefi=
t.
>
>
>  Non-BUNDLE is session multiplexing. It uses a separate RTP session for
> the retransmissions.
>
> Maybe I am the missing something, if you don't use bundle to send the
> audio/video on same rtpsession, you can still send rtx+video on same
> session. That's it non-bundle with ssrc-multiplexing. Are we referring to
> different things?
>
>
> Sure, but that doesn=E2=80=99t alter the fact that the group decided that
> non-BUNDLE media needs to be supported. Sending retransmission on a
> separate RTP session is as needed for interoperability with legacy system=
s
> as sending audio and video on separate RTP sessions (and shouldn=E2=80=99=
t be hard
> to support, since it uses the same mechanisms).
>
> Non-BUNDLE media needs to be supported, but I don't think the WG ever
explicitly signed up for session-multiplexing of RTX data. Unified plan is
quite clear in its assertion that the primary, RTX, and FEC flows for a
given media stream should all be represented by a single m=3D line, e.g.
SSRC-multiplexed.

--001a11c31dd813c1f705067096a8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Oct 27, 2014 at 12:16 PM, Colin Perkins <span dir=3D"ltr">&lt;<=
a href=3D"mailto:csp@csperkins.org" target=3D"_blank">csp@csperkins.org</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wra=
p:break-word"><span class=3D"">On 27 Oct 2014, at 17:17, Sergio Garcia Muri=
llo &lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank=
">sergio.garcia.murillo@gmail.com</a>&gt; wrote:</span><div><span class=3D"=
"><blockquote type=3D"cite">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>On 27/10/2014 18:10, Colin Perkins
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      On 27 Oct 2014, at 16:51, Sergio Garcia Murillo &lt;<a href=3D"mailto=
:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia.murillo@g=
mail.com</a>&gt;
      wrote:
      <div>
        <blockquote type=3D"cite">
         =20
          <div bgcolor=3D"#FFFFFF" text=3D"#000000">
            <div>On 27/10/2014 17:45, Colin
              Perkins wrote:<br>
            </div>
            <blockquote type=3D"cite">
             =20
              On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a href=
=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia=
.murillo@gmail.com</a>&gt;

              wrote:
              <div>
                <blockquote type=3D"cite">
                 =20
                  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
                    <div>On 27/10/2014 17:36,
                      Colin Perkins wrote:<br>
                    </div>
                    <blockquote type=3D"cite">
                      <div>
                        <div>
                          <div>On 21 Oct 2014, at 19:58, Sergio Garcia
                            Murillo &lt;<a href=3D"mailto:sergio.garcia.mur=
illo@gmail.com" target=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt;


                            wrote:</div>
                          <blockquote type=3D"cite">
                            <div bgcolor=3D"#FFFFFF" text=3D"#000000">Not
                              sure if it is done on pourpose, but
                              according to the RTP usage draft, it may
                              seem that full RFC 4588 is mandated at the
                              recevier side:<br>
                              <br>
                              <pre style=3D"font-size:1em;margin-top:0px;ma=
rgin-bottom:0px;font-style:normal;font-variant:normal;font-weight:normal;le=
tter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;tex=
t-transform:none;word-spacing:0px">    Receivers are REQUIRED to implement =
support for RTP retransmission
   =C2=A0packets [<a href=3D"https://tools.ietf.org/html/rfc4588" title=3D"=
&quot;RTP Retransmission Payload Format&quot;" target=3D"_blank">RFC4588</a=
>].</pre>
                              <br>
                              That would include both modes, session and
                              ssrc multiplexing. Given the extensive
                              usage of bundle and current
                              implementations, session multiplexing
                              support doesn&#39;t make much sense.<br>
                              <br>
                              Should we drop it, and state that only
                              ssrc-multiplexing shall be supported at
                              the receiving end?<br>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                        <div>I don=E2=80=99t see any advantage to doing so,
                          given that support for non-BUNDLE sessions is
                          REQUIRED. You need to implement the signalling
                          needed for session-multiplexing of
                          retransmission packet anyway, so disallowing
                          it buys you nothing.</div>
                      </div>
                    </blockquote>
                    <br>
                    You can do SSRC multiplexing with BUNDLE and
                    non-BUNDLE sessions, what I don&#39;t see is how to do
                    session multiplexing with BUNDLE sessions. <br>
                  </div>
                </blockquote>
                <br>
              </div>
              <div>You can=E2=80=99t do session multiplexing for BUNDLE
                sessions; by definition they use SSRC multiplexing. You
                could do non-BUNDLE sessions, with retransmission sent
                on a separate RTP session though.</div>
              <div><span style=3D"border-collapse:separate;border-spacing:0=
px">
                  <div><br>
                  </div>
                </span></div>
            </blockquote>
            So, you are saying exactly the same than me. SSRC
            multiplexing supports both BUNDLE and NON-BUNDLE. So, why
            require support for session multiplexing at all? As a
            developer, I don&#39;t see why I would have to implement
            something that would be rarely used and provide no extra
            benefit.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>Non-BUNDLE is session multiplexing. It uses a separate RTP
          session for the retransmissions. <br>
        </div>
      </div>
    </blockquote>
    Maybe I am the missing something, if you don&#39;t use bundle to send
    the audio/video on same rtpsession, you can still send rtx+video on
    same session. That&#39;s it non-bundle with ssrc-multiplexing. Are we
    referring to different things?</div></blockquote><div><br></div></span>=
<div>Sure, but that doesn=E2=80=99t alter the fact that the group decided t=
hat non-BUNDLE media needs to be supported. Sending retransmission on a sep=
arate RTP session is as needed for interoperability with legacy systems as =
sending audio and video on separate RTP sessions (and shouldn=E2=80=99t be =
hard to support, since it uses the same mechanisms).</div><span class=3D"HO=
EnZb"><font color=3D"#888888"><div><br></div></font></span></div></div></bl=
ockquote><div>Non-BUNDLE media needs to be supported, but I don&#39;t think=
 the WG ever explicitly signed up for session-multiplexing of RTX data. Uni=
fied plan is quite clear in its assertion that the primary, RTX, and FEC fl=
ows for a given media stream should all be represented by a single m=3D lin=
e, e.g. SSRC-multiplexed.=C2=A0</div></div></div></div>

--001a11c31dd813c1f705067096a8--


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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 27 October 2014 15:02, Harald Alvestrand <harald@alvestrand.no> wrote:
> You're confusing the webrtc application (what runs in the browser) with the
> webrtc implementation (the browser itself). Optional NDATA makes the browser
> *more* complex, since it has to expose API surface to let the application
> choose whether or not to offer NDATA, and it has to have code to decide
> whether or not to offer NDATA.


Yes.  I don't understand why we would want to say no to making NDATA
mandatory for browsers, other than to note that not all peers will
support NDATA.

SCTP without NDATA simply isn't fit for purpose.


From nobody Mon Oct 27 17:30:11 2014
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 27 Oct 2014 17:29:26 -0700
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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On Mon, Oct 27, 2014 at 5:25 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 27 October 2014 15:02, Harald Alvestrand <harald@alvestrand.no> wrote:
> > You're confusing the webrtc application (what runs in the browser) with
> the
> > webrtc implementation (the browser itself). Optional NDATA makes the
> browser
> > *more* complex, since it has to expose API surface to let the application
> > choose whether or not to offer NDATA, and it has to have code to decide
> > whether or not to offer NDATA.
>
>
> Yes.  I don't understand why we would want to say no to making NDATA
> mandatory for browsers, other than to note that not all peers will
> support NDATA.
>

I agree with this. Let's require what we need.

-Ekr

SCTP without NDATA simply isn't fit for purpose.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Oct 27, 2014 at 5:25 PM, Martin Thomson <span dir=3D"ltr">&lt;<=
a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson=
@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span cl=
ass=3D"">On 27 October 2014 15:02, Harald Alvestrand &lt;<a href=3D"mailto:=
harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; You&#39;re confusing the webrtc application (what runs in the browser)=
 with the<br>
&gt; webrtc implementation (the browser itself). Optional NDATA makes the b=
rowser<br>
&gt; *more* complex, since it has to expose API surface to let the applicat=
ion<br>
&gt; choose whether or not to offer NDATA, and it has to have code to decid=
e<br>
&gt; whether or not to offer NDATA.<br>
<br>
<br>
</span>Yes.=C2=A0 I don&#39;t understand why we would want to say no to mak=
ing NDATA<br>
mandatory for browsers, other than to note that not all peers will<br>
support NDATA.<br></blockquote><div><br></div><div>I agree with this. Let&#=
39;s require what we need.</div><div><br></div><div>-Ekr</div><div><br></di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
SCTP without NDATA simply isn&#39;t fit for purpose.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 27 Oct 2014 17:39:11 -0700
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To: Eric Rescorla <ekr@rtfm.com>
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On Mon, Oct 27, 2014 at 5:29 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
> On Mon, Oct 27, 2014 at 5:25 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
>> On 27 October 2014 15:02, Harald Alvestrand <harald@alvestrand.no> wrote:
>> > You're confusing the webrtc application (what runs in the browser) with
>> the
>> > webrtc implementation (the browser itself). Optional NDATA makes the
>> browser
>> > *more* complex, since it has to expose API surface to let the
>> application
>> > choose whether or not to offer NDATA, and it has to have code to decide
>> > whether or not to offer NDATA.
>>
>>
>> Yes.  I don't understand why we would want to say no to making NDATA
>> mandatory for browsers, other than to note that not all peers will
>> support NDATA.
>>
>
> I agree with this. Let's require what we need.
>

+1. The fewer optional things we have, the more likely things will Just
Work.

--089e0149c8e26e0aeb050670e199
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Oct 27, 2014 at 5:29 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote"><span class=3D"">On Mon, Oc=
t 27, 2014 at 5:25 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mail=
to:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On 27 October 20=
14 15:02, Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no" tar=
get=3D"_blank">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; You&#39;re confusing the webrtc application (what runs in the browser)=
 with the<br>
&gt; webrtc implementation (the browser itself). Optional NDATA makes the b=
rowser<br>
&gt; *more* complex, since it has to expose API surface to let the applicat=
ion<br>
&gt; choose whether or not to offer NDATA, and it has to have code to decid=
e<br>
&gt; whether or not to offer NDATA.<br>
<br>
<br>
</span>Yes.=C2=A0 I don&#39;t understand why we would want to say no to mak=
ing NDATA<br>
mandatory for browsers, other than to note that not all peers will<br>
support NDATA.<br></blockquote><div><br></div></span><div>I agree with this=
. Let&#39;s require what we need.</div></div></div></div></blockquote><div>=
<br></div><div>+1. The fewer optional things we have, the more likely thing=
s will Just Work.=C2=A0</div></div></div></div>

--089e0149c8e26e0aeb050670e199--


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References: <5446ACD8.1010004@gmail.com> <B9AC89FB-C656-42C7-9204-C2B3AC6B8E29@csperkins.org> <544E7586.4080703@gmail.com> <22D97583-2E07-417C-84CC-923FD83C008C@csperkins.org> <544E781D.50305@gmail.com> <17742E9C-CCDD-4EFE-A2C1-C84531A0523F@csperkins.org> <544E7E2B.6040809@gmail.com> <9DEB3428-1B9E-4876-A6B5-B4CCE69D84AE@csperkins.org> <CAOJ7v-2wN-ff4RrM3rARYU=kwefj=MJ6mMD21a9_FXFpdYg8Og@mail.gmail.com>
From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Mon, 27 Oct 2014 20:41:12 -0700
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To: Justin Uberti <juberti@google.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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Justin said:

"I don't think the WG ever explicitly signed up for session-multiplexing of
RTX data"

[BA] I agree, and in addition would say the same thing with respect to FEC
- one of the fundamental objections to RFC 5109 is that it only support
session-multiplexing, not SSRC multiplexing.

On Mon, Oct 27, 2014 at 5:18 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Mon, Oct 27, 2014 at 12:16 PM, Colin Perkins <csp@csperkins.org> wrote=
:
>
>> On 27 Oct 2014, at 17:17, Sergio Garcia Murillo <
>> sergio.garcia.murillo@gmail.com> wrote:
>>
>>  On 27/10/2014 18:10, Colin Perkins wrote:
>>
>> On 27 Oct 2014, at 16:51, Sergio Garcia Murillo <
>> sergio.garcia.murillo@gmail.com> wrote:
>>
>>  On 27/10/2014 17:45, Colin Perkins wrote:
>>
>> On 27 Oct 2014, at 16:40, Sergio Garcia Murillo <
>> sergio.garcia.murillo@gmail.com> wrote:
>>
>>  On 27/10/2014 17:36, Colin Perkins wrote:
>>
>>  On 21 Oct 2014, at 19:58, Sergio Garcia Murillo <
>> sergio.garcia.murillo@gmail.com> wrote:
>>
>> Not sure if it is done on pourpose, but according to the RTP usage draft=
,
>> it may seem that full RFC 4588 is mandated at the recevier side:
>>
>>     Receivers are REQUIRED to implement support for RTP retransmission
>>     packets [RFC4588 <https://tools.ietf.org/html/rfc4588>].
>>
>>
>> That would include both modes, session and ssrc multiplexing. Given the
>> extensive usage of bundle and current implementations, session multiplex=
ing
>> support doesn't make much sense.
>>
>> Should we drop it, and state that only ssrc-multiplexing shall be
>> supported at the receiving end?
>>
>>
>>  I don=E2=80=99t see any advantage to doing so, given that support for
>> non-BUNDLE sessions is REQUIRED. You need to implement the signalling
>> needed for session-multiplexing of retransmission packet anyway, so
>> disallowing it buys you nothing.
>>
>>
>> You can do SSRC multiplexing with BUNDLE and non-BUNDLE sessions, what I
>> don't see is how to do session multiplexing with BUNDLE sessions.
>>
>>
>>  You can=E2=80=99t do session multiplexing for BUNDLE sessions; by defin=
ition
>> they use SSRC multiplexing. You could do non-BUNDLE sessions, with
>> retransmission sent on a separate RTP session though.
>>
>>   So, you are saying exactly the same than me. SSRC multiplexing
>> supports both BUNDLE and NON-BUNDLE. So, why require support for session
>> multiplexing at all? As a developer, I don't see why I would have to
>> implement something that would be rarely used and provide no extra benef=
it.
>>
>>
>>  Non-BUNDLE is session multiplexing. It uses a separate RTP session for
>> the retransmissions.
>>
>> Maybe I am the missing something, if you don't use bundle to send the
>> audio/video on same rtpsession, you can still send rtx+video on same
>> session. That's it non-bundle with ssrc-multiplexing. Are we referring t=
o
>> different things?
>>
>>
>> Sure, but that doesn=E2=80=99t alter the fact that the group decided tha=
t
>> non-BUNDLE media needs to be supported. Sending retransmission on a
>> separate RTP session is as needed for interoperability with legacy syste=
ms
>> as sending audio and video on separate RTP sessions (and shouldn=E2=80=
=99t be hard
>> to support, since it uses the same mechanisms).
>>
>> Non-BUNDLE media needs to be supported, but I don't think the WG ever
> explicitly signed up for session-multiplexing of RTX data. Unified plan i=
s
> quite clear in its assertion that the primary, RTX, and FEC flows for a
> given media stream should all be represented by a single m=3D line, e.g.
> SSRC-multiplexed.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bb70a7262d6d30506736ceb
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Justin said:=C2=A0<div><br></div><div>&quot;<span style=3D=
"font-family:arial,sans-serif;font-size:13px">I don&#39;t think the WG ever=
 explicitly signed up for session-multiplexing of RTX data</span>&quot;</di=
v><div><br></div><div>[BA] I agree, and in addition would say the same thin=
g with respect to FEC - one of the fundamental objections to RFC 5109 is th=
at it only support session-multiplexing, not SSRC multiplexing. =C2=A0</div=
></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Oc=
t 27, 2014 at 5:18 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"=
gmail_extra"><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Mon, =
Oct 27, 2014 at 12:16 PM, Colin Perkins <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:csp@csperkins.org" target=3D"_blank">csp@csperkins.org</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word=
"><span>On 27 Oct 2014, at 17:17, Sergio Garcia Murillo &lt;<a href=3D"mail=
to:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia.murillo=
@gmail.com</a>&gt; wrote:</span><div><span><blockquote type=3D"cite">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>On 27/10/2014 18:10, Colin Perkins
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      On 27 Oct 2014, at 16:51, Sergio Garcia Murillo &lt;<a href=3D"mailto=
:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia.murillo@g=
mail.com</a>&gt;
      wrote:
      <div>
        <blockquote type=3D"cite">
         =20
          <div bgcolor=3D"#FFFFFF" text=3D"#000000">
            <div>On 27/10/2014 17:45, Colin
              Perkins wrote:<br>
            </div>
            <blockquote type=3D"cite">
             =20
              On 27 Oct 2014, at 16:40, Sergio Garcia Murillo &lt;<a href=
=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia=
.murillo@gmail.com</a>&gt;

              wrote:
              <div>
                <blockquote type=3D"cite">
                 =20
                  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
                    <div>On 27/10/2014 17:36,
                      Colin Perkins wrote:<br>
                    </div>
                    <blockquote type=3D"cite">
                      <div>
                        <div>
                          <div>On 21 Oct 2014, at 19:58, Sergio Garcia
                            Murillo &lt;<a href=3D"mailto:sergio.garcia.mur=
illo@gmail.com" target=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt;


                            wrote:</div>
                          <blockquote type=3D"cite">
                            <div bgcolor=3D"#FFFFFF" text=3D"#000000">Not
                              sure if it is done on pourpose, but
                              according to the RTP usage draft, it may
                              seem that full RFC 4588 is mandated at the
                              recevier side:<br>
                              <br>
                              <pre style=3D"font-size:1em;margin-top:0px;ma=
rgin-bottom:0px;font-style:normal;font-variant:normal;font-weight:normal;le=
tter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;tex=
t-transform:none;word-spacing:0px">    Receivers are REQUIRED to implement =
support for RTP retransmission
   =C2=A0packets [<a href=3D"https://tools.ietf.org/html/rfc4588" title=3D"=
&quot;RTP Retransmission Payload Format&quot;" target=3D"_blank">RFC4588</a=
>].</pre>
                              <br>
                              That would include both modes, session and
                              ssrc multiplexing. Given the extensive
                              usage of bundle and current
                              implementations, session multiplexing
                              support doesn&#39;t make much sense.<br>
                              <br>
                              Should we drop it, and state that only
                              ssrc-multiplexing shall be supported at
                              the receiving end?<br>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                        <div>I don=E2=80=99t see any advantage to doing so,
                          given that support for non-BUNDLE sessions is
                          REQUIRED. You need to implement the signalling
                          needed for session-multiplexing of
                          retransmission packet anyway, so disallowing
                          it buys you nothing.</div>
                      </div>
                    </blockquote>
                    <br>
                    You can do SSRC multiplexing with BUNDLE and
                    non-BUNDLE sessions, what I don&#39;t see is how to do
                    session multiplexing with BUNDLE sessions. <br>
                  </div>
                </blockquote>
                <br>
              </div>
              <div>You can=E2=80=99t do session multiplexing for BUNDLE
                sessions; by definition they use SSRC multiplexing. You
                could do non-BUNDLE sessions, with retransmission sent
                on a separate RTP session though.</div>
              <div><span style=3D"border-collapse:separate;border-spacing:0=
px">
                  <div><br>
                  </div>
                </span></div>
            </blockquote>
            So, you are saying exactly the same than me. SSRC
            multiplexing supports both BUNDLE and NON-BUNDLE. So, why
            require support for session multiplexing at all? As a
            developer, I don&#39;t see why I would have to implement
            something that would be rarely used and provide no extra
            benefit.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>Non-BUNDLE is session multiplexing. It uses a separate RTP
          session for the retransmissions. <br>
        </div>
      </div>
    </blockquote>
    Maybe I am the missing something, if you don&#39;t use bundle to send
    the audio/video on same rtpsession, you can still send rtx+video on
    same session. That&#39;s it non-bundle with ssrc-multiplexing. Are we
    referring to different things?</div></blockquote><div><br></div></span>=
<div>Sure, but that doesn=E2=80=99t alter the fact that the group decided t=
hat non-BUNDLE media needs to be supported. Sending retransmission on a sep=
arate RTP session is as needed for interoperability with legacy systems as =
sending audio and video on separate RTP sessions (and shouldn=E2=80=99t be =
hard to support, since it uses the same mechanisms).</div><span><font color=
=3D"#888888"><div><br></div></font></span></div></div></blockquote></div></=
div><div>Non-BUNDLE media needs to be supported, but I don&#39;t think the =
WG ever explicitly signed up for session-multiplexing of RTX data. Unified =
plan is quite clear in its assertion that the primary, RTX, and FEC flows f=
or a given media stream should all be represented by a single m=3D line, e.=
g. SSRC-multiplexed.=C2=A0</div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7bb70a7262d6d30506736ceb--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Date: Tue, 28 Oct 2014 10:10:14 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/Ip-_4YZ0deZ2T8kFh-By3nrsOow
Subject: Re: [rtcweb] Drop RFC 4588 RTX session multiplexing support requirement from RTP USAGE
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On 28/10/2014 4:41, Bernard Aboba wrote:
> Justin said:
>
> "I don't think the WG ever explicitly signed up for 
> session-multiplexing of RTX data"
>
> [BA] I agree, and in addition would say the same thing with respect to 
> FEC - one of the fundamental objections to RFC 5109 is that it only 
> support session-multiplexing, not SSRC multiplexing.
>

+1
Sergio

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    <div class="moz-cite-prefix"><br>
      On 28/10/2014 4:41, Bernard Aboba wrote:<br>
    </div>
    <blockquote
cite="mid:CAOW+2duT7UJZavAJKVWU6vv-Aby-L=Urjyk+KiPBb1UPNhYY1w@mail.gmail.com"
      type="cite">
      <div dir="ltr">Justin said:&nbsp;
        <div><br>
        </div>
        <div>"<span style="font-family:arial,sans-serif;font-size:13px">I
            don't think the WG ever explicitly signed up for
            session-multiplexing of RTX data</span>"</div>
        <div><br>
        </div>
        <div>[BA] I agree, and in addition would say the same thing with
          respect to FEC - one of the fundamental objections to RFC 5109
          is that it only support session-multiplexing, not SSRC
          multiplexing. &nbsp;</div>
      </div>
      <br>
    </blockquote>
    &nbsp;<br>
    +1<br>
    Sergio<br>
  </body>
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--------------090003010106010009070101--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Monday, October 27, 2014 5:03 PM

On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju) wrote:
+1 to make N-DATA optional.

Data channel provides a robust generic peer to peer communication channel. =
So, one can see that data channel may be used under non-webrtc contexts (li=
ke as in CLUE) as a general transport or as a replacement of native to SCTP=
 to traverse firewalls. IMHO, majority of these use cases:
a. May just involve a single data channel (with possible multiplexing on to=
p of it)
b. And may not use the DCEP either (both ends could assume same data channe=
l stream id, this is similar to SCTP PPIDs).
We already made DCEP(b) as optional.

If an app is using just a single data channel then there is no real benefit=
 in using N-DATA. Then why not make it optional? Also, as already pointed o=
ut, making it optional won=92t break interop as it is still negotiated duri=
ng SCTP setup time anyway.

Negotiation means that transport initiator gets to choose whether to use it=
 or not.
This means that a responder has no choice in the matter.

In the situation where the responder knows better than the initiator what t=
he usage is going to be, this is problematic.
<Raju>
IMHO, it is not just the responder knowing =93better=94 which makes a scena=
rio work, rather both have to agree on it. For example, if responder wants =
to have multiple data channels is no good if the originator does not want t=
o support (CLUE use of data channels is an example of it). That=92s why, in=
 most cases I know it is the originator who offers capabilities and termina=
tor gets to choose.
</Raju>



I would like to point out that there will be some webrtc applications which=
 might just use one webrtc data channel for simplicity (and may use some mu=
ltiplexing of different types of msgs/objects on top of it) or one is just =
sufficient as the app does not send/recv huge chunks of data. So, N-DATA is=
n=92t needed for such webrtc clients either. Then why add burden to such we=
brtc implementations!?

You're confusing the webrtc application (what runs in the browser) with the=
 webrtc implementation (the browser itself). Optional NDATA makes the brows=
er *more* complex, since it has to expose API surface to let the applicatio=
n choose whether or not to offer NDATA, and it has to have code to decide w=
hether or not to offer NDATA.
<Raju>
I am talking about 2 items:

1.       Making NDATA a MUST data channel draft? I could be wrong, but I th=
ought this is the main discussion point in this thread! This is a burden on=
 non-webrtc users of data channel. My understanding is data channels are me=
ant to be used by non-webrtc as a general protocol. Please correct me if my=
 understanding is not correct.

2.       Making NDATA a MUST for webrtc clients? I am not just talking abou=
t browser, but non-browser based native/hybrid webrtc clients (e.g. a thin =
webrtc client on a cable set top box) as well. Browsers can implement it as=
 a MUST clause, no one is stopping that. But why mandate a set top box (or =
an IoT device) implement NDATA when it just wants to use a single data chan=
nel only?!

Let=92s take an example: We keep NDATA a MUST. During SCTP setup, if NDATA =
is not supported at the peer then will the SCTP setup be abandoned? If the =
SCTP association is not abandoned then there is no point in making it a MUS=
T where as in real practice it is not enforced as a MUST.
</Raju>

BR
Raju



From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: Monday, October 27, 2014 3:11 PM
To: Harald Alvestrand; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving

Hi,

We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Harald Alvestrand<mailto:harald@alvestrand.no>
Sent: =FD27/=FD10/=FD2014 22:00
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; rtcweb@ietf.o=
rg<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
On 10/27/2014 11:41 AM, Christer Holmberg wrote:
> Hi,
>
>>>>> Within the CLUE WG, we had a discussion regarding the following state=
ment in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt:
>>>>>
>>>>> "The support for message interleaving as defined in
>>>>>                 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used."
>>>>>
>>>>> First, it is a little unclear what "the support SHOULD be used" means=
.
>>>> My understanding is that SCTP implementations supporting the extension=
 will use it.
>>>> This is negotiated during the setup of the SCTP association.
>>> If it's done on SCTP level, why do we need to talk about it in the data=
 channel draft?
>>>
>>> Is there a reason why it is important to use it for data channels? If s=
o, does it apply to data channels in general?
>> NDATA was added in order to avoid head-of-line blocking on the transport=
 (if I understand this correctly, until
>> this was added, sending a huge message would block the delivery of small=
 messages on all channels until the huge message was fully delivered).
>>
>> Unlike Michael, I see no reason to make this a SHOULD; I think it should=
 be a MUST, and the older
>> implementations in browsers should just be called out as non-conformant.
>>
>> That said, I think that data channels ought to interoperate successfully=
 with implementations that don't support the
>> extension - but data channel implementations in WebRTC endpoints should =
be under a "MUST implement, MUST offer" ruleset.
> First, keep in mind that I am NOT talking about WebRTC endpoints (which i=
s one reason we shouldn't call it webrtc data channel to begin with, but th=
at's another story...), but ANY type of endpoint that wants to use a data c=
hannel.

I'm a bit self-centric on behalf of WebRTC, but I feel the other way -
that it might have been a mistake to float the option of saying "other
things can use these features", and that we should admit only arguments
that bear upon their usage in WebRTC.

I'm not sure I have WG consensus (even rough consensus) on this point,
however.
>
> Second, I am not talking about MUST support, but MUST use/offer.
>
> Assume I have a CLUE endpoint, which will use a data channel ONLY to tran=
sport CLUE messages. CLUE messages aren't that hugeto begin with, and there=
 will be no blocking of small messages on other channels - as there are no =
other channels.
>
> So, why does the CLUE endpoint need to offer interleaving?

Flipping the question around - if a programming mistake in the
Javascript implementing a CLUE endpoint in a WebRTC browser happens to
send a multimegabyte-sized object down the (out of order) datachannel,
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?



--
Surveillance is pervasive. Go Dark.






--

Surveillance is pervasive. Go Dark.

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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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
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<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<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"> Harald Alvestrand [mailto:harald@alvestrand.no]
<br>
<b>Sent:</b> Monday, October 27, 2014 5:03 PM<br>
<br>
</span></p>
<div>
<p class=3D"MsoNormal">On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju)=
 wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#43;1 to make N-DATA opt=
ional.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Data channel provides a r=
obust generic peer to peer communication channel. So, one can see that data=
 channel may be used under non-webrtc contexts (like as
 in CLUE) as a general transport or as a replacement of native to SCTP to t=
raverse firewalls. IMHO, majority of these use cases:</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">a. May just involve a sin=
gle data channel (with possible multiplexing on top of it)
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">b. And may not use the DC=
EP either (both ends could assume same data channel stream id, this is simi=
lar to SCTP PPIDs).
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">We already made DCEP(b) a=
s optional.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">If an app is using just a=
 single data channel then there is no real benefit in using N-DATA. Then wh=
y not make it optional? Also, as already pointed out, making
 it optional won=92t break interop as it is still negotiated during SCTP se=
tup time anyway.</span><o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><br>
Negotiation means that transport initiator gets to choose whether to use it=
 or not.<br>
This means that a responder has no choice in the matter.<br>
<br>
In the situation where the responder knows better than the initiator what t=
he usage is going to be, this is problematic.<span style=3D"color:#1F497D">=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;<o:p></=
o:p></span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">IMHO, it is not just t=
he responder knowing =93better=94 which makes a scenario work, rather both =
have to agree on it. For example, if responder wants to have multiple data =
channels is no good if the originator does
 not want to support (CLUE use of data channels is an example of it). That=
=92s why, in most cases I know it is the originator who offers capabilities=
 and terminator gets to choose.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<br>
</span></i></b><br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I would like to point out=
 that there will be some webrtc applications which might just use one webrt=
c data channel for simplicity (and may use some multiplexing
 of different types of msgs/objects on top of it) or one is just sufficient=
 as the app does not send/recv huge chunks of data. So, N-DATA isn=92t need=
ed for such webrtc clients either. Then why add burden to such webrtc imple=
mentations!?</span><o:p></o:p></p>
<p class=3D"MsoNormal"><br>
You're confusing the webrtc application (what runs in the browser) with the=
 webrtc implementation (the browser itself). Optional NDATA makes the brows=
er *more* complex, since it has to expose API surface to let the applicatio=
n choose whether or not to offer
 NDATA, and it has to have code to decide whether or not to offer NDATA.<sp=
an style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;<o:p></=
o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I am talking about =
2 items:<o:p></o:p></span></i></b></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=
=3D"mso-list:Ignore">1.<span style=3D"font:7.0pt &quot;Times New Roman&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><b><i><span style=3D"font-size:11.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Mak=
ing NDATA a MUST data channel draft? I could be wrong, but I thought this i=
s the main discussion point in this thread! This is a
 burden on non-webrtc users of data channel. My understanding is data chann=
els are meant to be used by non-webrtc as a general protocol. Please correc=
t me if my understanding is not correct.<o:p></o:p></span></i></b></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=
=3D"mso-list:Ignore">2.<span style=3D"font:7.0pt &quot;Times New Roman&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><b><i><span style=3D"font-size:11.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Mak=
ing NDATA a MUST for webrtc clients? I am not just talking about browser, b=
ut non-browser based native/hybrid webrtc clients (e.g.
 a thin webrtc client on a cable set top box) as well. Browsers can impleme=
nt it as a MUST clause, no one is stopping that. But why mandate a set top =
box (or an IoT device) implement NDATA when it just wants to use a single d=
ata channel only?!<o:p></o:p></span></i></b></p>
<p class=3D"MsoListParagraph"><b><i><span style=3D"font-size:11.0pt;font-fa=
mily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Let=92s take=
 an example: We keep NDATA a MUST. During SCTP setup, if NDATA is not suppo=
rted at the peer then will the SCTP setup be abandoned? If
 the SCTP association is not abandoned then there is no point in making it =
a MUST where as in real practice it is not enforced as a MUST.<o:p></o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">BR<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Raju</span><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1F497D">&nbsp;</span><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">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span 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 [=
<a href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</=
a>]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> Monday, October 27, 2014 3:11 PM<br>
<b>To:</b> Harald Alvestrand; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a><br>
<b>Subject:</b> Re: [rtcweb] Meaning of SHOULD support/use interleaving</sp=
an><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></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;">Hi,<br>
<br>
We shall not add burden just because there are bad programmers out there - =
they will anyway always manage to use our specs in a wrong way :)<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</span><o:p></o:p></p>
</div>
</div>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><span style=3D"fon=
t-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:
</span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,=
&quot;sans-serif&quot;"><a href=3D"mailto:harald@alvestrand.no">Harald Alve=
strand</a></span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">Sent: </span>
</b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;">=FD27/=FD10/=FD2014 22:00</span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">To: </span></b><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;"><a href=3D"mailto:christer.holm=
berg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;">Subject: </span>
</b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;">Re: [rtcweb] Meaning of SHOULD support/use interleaving</s=
pan><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt">On 10/27/2014 11:41 AM, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt;&gt;&gt; Within the CLUE WG, we had a discussion regarding the =
following statement in section 6.1 of draft-ietf-rtcweb-data-channel-12.txt=
:<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; &quot;The support for message interleaving as defined =
in<br>
&gt;&gt;&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata] SHOUL=
D be used.&quot;<br>
&gt;&gt;&gt;&gt;&gt;&nbsp; <br>
&gt;&gt;&gt;&gt;&gt; First, it is a little unclear what &quot;the support S=
HOULD be used&quot; means.<br>
&gt;&gt;&gt;&gt; My understanding is that SCTP implementations supporting t=
he extension will use it.<br>
&gt;&gt;&gt;&gt; This is negotiated during the setup of the SCTP associatio=
n.<br>
&gt;&gt;&gt; If it's done on SCTP level, why do we need to talk about it in=
 the data channel draft?
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Is there a reason why it is important to use it for data chann=
els? If so, does it apply to data channels in general?
<br>
&gt;&gt; NDATA was added in order to avoid head-of-line blocking on the tra=
nsport (if I understand this correctly, until
<br>
&gt;&gt; this was added, sending a huge message would block the delivery of=
 small messages on all channels until the huge message was fully delivered)=
.<br>
&gt;&gt;<br>
&gt;&gt; Unlike Michael, I see no reason to make this a SHOULD; I think it =
should be a MUST, and the older
<br>
&gt;&gt; implementations in browsers should just be called out as non-confo=
rmant.<br>
&gt;&gt;<br>
&gt;&gt; That said, I think that data channels ought to interoperate succes=
sfully with implementations that don't support the
<br>
&gt;&gt; extension - but data channel implementations in WebRTC endpoints s=
hould be under a &quot;MUST implement, MUST offer&quot; ruleset.<br>
&gt; First, keep in mind that I am NOT talking about WebRTC endpoints (whic=
h is one reason we shouldn't call it webrtc data channel to begin with, but=
 that's another story...), but ANY type of endpoint that wants to use a dat=
a channel.
<br>
<br>
I'm a bit self-centric on behalf of WebRTC, but I feel the other way -<br>
that it might have been a mistake to float the option of saying &quot;other=
<br>
things can use these features&quot;, and that we should admit only argument=
s<br>
that bear upon their usage in WebRTC.<br>
<br>
I'm not sure I have WG consensus (even rough consensus) on this point,<br>
however.<br>
&gt;<br>
&gt; Second, I am not talking about MUST support, but MUST use/offer.<br>
&gt;<br>
&gt; Assume I have a CLUE endpoint, which will use a data channel ONLY to t=
ransport CLUE messages. CLUE messages aren't that hugeto begin with, and th=
ere will be no blocking of small messages on other channels - as there are =
no other channels.<br>
&gt;<br>
&gt; So, why does the CLUE endpoint need to offer interleaving?<br>
<br>
Flipping the question around - if a programming mistake in the<br>
Javascript implementing a CLUE endpoint in a WebRTC browser happens to<br>
send a multimegabyte-sized object down the (out of order) datachannel,<br>
is it OK to have all the subsequent CLUE messages delivered 30 seconds late=
?<br>
<br>
<br>
<br>
-- <br>
Surveillance is pervasive. Go Dark.<br>
<br>
<br>
</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Surveillance is pervasive. Go Dark.<o:p></o:p></pre>
</div>
</div>
</body>
</html>

--_000_E1FE4C082A89A246A11D7F32A95A17828E5F5EBDUS70UWXCHMBA02z_--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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> > +1 to make N-DATA optional.
> >
> > Data channel provides a robust generic peer to peer communication chann=
el.
> So, one can see that data channel may be used under non-webrtc contexts
> (like as in CLUE) as a general transport or as a replacement of native to
> SCTP to traverse firewalls. IMHO, majority of these use cases:
> > a. May just involve a single data channel (with possible multiplexing o=
n
> top of it)
> > b. And may not use the DCEP either (both ends could assume same data
> channel stream id, this is similar to SCTP PPIDs).
> > We already made DCEP(b) as optional.
> >
> > If an app is using just a single data channel then there is no real
> benefit in using N-DATA. Then why not make it optional? Also, as already
> pointed out, making it optional won't break interop as it is still
> negotiated during SCTP setup time anyway.
> >
> > I would like to point out that there will be some webrtc applications
> which might just use one webrtc data channel for simplicity (and may use
> some multiplexing of different types of msgs/objects on top of it) or one=
 is
> just sufficient as the app does not send/recv huge chunks of data. So, N-
> DATA isn't needed for such webrtc clients either. Then why add burden to
> such webrtc implementations!?
> Are you implementing this? I would say that adding N-DATA to an SCTP
> implementation with support
> for Stream Reconfiguration and PR-SCTP does not increase the complexity
> substantially...

<Raju>
I don't think the question is really about if NDATA adds complexity or not.
NDATA is a new capability to be added to SCTP stacks; and adding it takes
resources to develop, test, deliver, deploy/update.

As I just responded to Harald's post, we need to look at it from
data channel use in non-webrtc contexts and also data channel use
for non-browser webrtc clients. Just focusing on webrtc+browser alone
is not fair as not all the webrtc or non-webrtc data channel implementation=
s
can keep up with the same pace as browsers do; also they don't have
resources like browser companies do!!

</Raju>

BR
Raju


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On 10/28/2014 04:58 AM, Makaraju, Maridi Raju (Raju) wrote:
>
> =20
>
> =20
>
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Monday, October 27, 2014 5:03 PM
>
> On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju) wrote:
>
>     +1 to make N-DATA optional.
>
>     =20
>
>     Data channel provides a robust generic peer to peer communication
>     channel. So, one can see that data channel may be used under
>     non-webrtc contexts (like as in CLUE) as a general transport or as
>     a replacement of native to SCTP to traverse firewalls. IMHO,
>     majority of these use cases:
>
>     a. May just involve a single data channel (with possible
>     multiplexing on top of it)
>
>     b. And may not use the DCEP either (both ends could assume same
>     data channel stream id, this is similar to SCTP PPIDs).
>
>     We already made DCEP(b) as optional.
>
>     =20
>
>     If an app is using just a single data channel then there is no
>     real benefit in using N-DATA. Then why not make it optional? Also,
>     as already pointed out, making it optional won=92t break interop as=

>     it is still negotiated during SCTP setup time anyway.
>
>
> Negotiation means that transport initiator gets to choose whether to
> use it or not.
> This means that a responder has no choice in the matter.
>
> In the situation where the responder knows better than the initiator
> what the usage is going to be, this is problematic.
>
> <Raju>
>
> IMHO, it is not just the responder knowing =93better=94 which makes a
> scenario work, rather both have to agree on it. For example, if
> responder wants to have multiple data channels is no good if the
> originator does not want to support (CLUE use of data channels is an
> example of it). That=92s why, in most cases I know it is the originator=

> who offers capabilities and terminator gets to choose.
>
> </Raju>
>
>
> =20
>
> I would like to point out that there will be some webrtc applications
> which might just use one webrtc data channel for simplicity (and may
> use some multiplexing of different types of msgs/objects on top of it)
> or one is just sufficient as the app does not send/recv huge chunks of
> data. So, N-DATA isn=92t needed for such webrtc clients either. Then wh=
y
> add burden to such webrtc implementations!?
>
>
> You're confusing the webrtc application (what runs in the browser)
> with the webrtc implementation (the browser itself). Optional NDATA
> makes the browser *more* complex, since it has to expose API surface
> to let the application choose whether or not to offer NDATA, and it
> has to have code to decide whether or not to offer NDATA.
>
> <Raju>
>
> I am talking about 2 items:
>
> 1.       Making NDATA a MUST data channel draft? I could be wrong, but
> I thought this is the main discussion point in this thread! This is a
> burden on non-webrtc users of data channel. My understanding is data
> channels are meant to be used by non-webrtc as a general protocol.
> Please correct me if my understanding is not correct.
>

Your understanding may be right or wrong, but it's different from mine.

Data channels are created to be useful for WebRTC. That's their purpose
in being created, so WebRTC requirements take precedence over all other
requirements.

NDATA makes them more useful for WebRTC (or, as Martin put it, "without
NDATA, they're not fit for purpose").

Optional NDATA makes things more complex for WebRTC.

Therefore, seen from a WebRTC viewpoint, speaking in the RTCWEB WG, I
think NDATA should be mandatory.

> 2.       Making NDATA a MUST for webrtc clients? I am not just talking
> about browser, but non-browser based native/hybrid webrtc clients
> (e.g. a thin webrtc client on a cable set top box) as well. Browsers
> can implement it as a MUST clause, no one is stopping that. But why
> mandate a set top box (or an IoT device) implement NDATA when it just
> wants to use a single data channel only?!
>
> Let=92s take an example: We keep NDATA a MUST. During SCTP setup, if
> NDATA is not supported at the peer then will the SCTP setup be
> abandoned? If the SCTP association is not abandoned then there is no
> point in making it a MUST where as in real practice it is not enforced
> as a MUST.
>

In general, when an application connects to something that does not
conform to the spec for the protocol he's using, "all bets are off".

Implementations are free to handle this situation whatever way they want.=


Note: So far, I've seen *zero* evidence that the burden of implementing
NDATA is significantly higher than the total effort spent in writing
updates on this thread. For 90+% of all implementors, not implementing
NDATA is a larger effort than implementing it, because the libraries
they use will support it, so they have to take action to take it out.


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    <div class="moz-cite-prefix">On 10/28/2014 04:58 AM, Makaraju,
      Maridi Raju (Raju) wrote:<br>
    </div>
    <blockquote
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        <p class="MsoNormal"> </p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <p class="MsoNormal">From: Harald Alvestrand
            [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>]
            <br>
            Sent: Monday, October 27, 2014 5:03 PM<br>
            <br>
          </p>
          <div>
            <p class="MsoNormal">On 10/27/2014 01:31 PM, Makaraju,
              Maridi Raju (Raju) wrote:</p>
          </div>
          <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
            <p class="MsoNormal">+1 to make N-DATA optional.</p>
            <p class="MsoNormal"> </p>
            <p class="MsoNormal">Data channel provides a robust generic
              peer to peer communication channel. So, one can see that
              data channel may be used under non-webrtc contexts (like
              as in CLUE) as a general transport or as a replacement of
              native to SCTP to traverse firewalls. IMHO, majority of
              these use cases:</p>
            <p class="MsoNormal">a. May just involve a single data
              channel (with possible multiplexing on top of it)
            </p>
            <p class="MsoNormal">b. And may not use the DCEP either
              (both ends could assume same data channel stream id, this
              is similar to SCTP PPIDs).
            </p>
            <p class="MsoNormal">We already made DCEP(b) as optional.</p>
            <p class="MsoNormal"> </p>
            <p class="MsoNormal">If an app is using just a single data
              channel then there is no real benefit in using N-DATA.
              Then why not make it optional? Also, as already pointed
              out, making it optional won’t break interop as it is still
              negotiated during SCTP setup time anyway.</p>
          </blockquote>
          <p class="MsoNormal"><br>
            Negotiation means that transport initiator gets to choose
            whether to use it or not.<br>
            This means that a responder has no choice in the matter.<br>
            <br>
            In the situation where the responder knows better than the
            initiator what the usage is going to be, this is
            problematic.</p>
          <p class="MsoNormal">&lt;Raju&gt;</p>
          <p class="MsoNormal">IMHO, it is not just the responder
            knowing “better” which makes a scenario work, rather both
            have to agree on it. For example, if responder wants to have
            multiple data channels is no good if the originator does not
            want to support (CLUE use of data channels is an example of
            it). That’s why, in most cases I know it is the originator
            who offers capabilities and terminator gets to choose.
          </p>
          <p class="MsoNormal">&lt;/Raju&gt;<br>
            <br>
            <br>
          </p>
          <p class="MsoNormal"> </p>
          <p class="MsoNormal">I would like to point out that there will
            be some webrtc applications which might just use one webrtc
            data channel for simplicity (and may use some multiplexing
            of different types of msgs/objects on top of it) or one is
            just sufficient as the app does not send/recv huge chunks of
            data. So, N-DATA isn’t needed for such webrtc clients
            either. Then why add burden to such webrtc implementations!?</p>
          <p class="MsoNormal"><br>
            You're confusing the webrtc application (what runs in the
            browser) with the webrtc implementation (the browser
            itself). Optional NDATA makes the browser *more* complex,
            since it has to expose API surface to let the application
            choose whether or not to offer NDATA, and it has to have
            code to decide whether or not to offer NDATA.</p>
          <p class="MsoNormal">&lt;Raju&gt;</p>
          <p class="MsoNormal">I am talking about 2 items:</p>
          <p class="MsoListParagraph"
            style="text-indent:-.25in;mso-list:l0 level1 lfo1"><!--[if !supportLists]-->1.      
            <!--[endif]-->Making NDATA a MUST data channel draft? I
            could be wrong, but I thought this is the main discussion
            point in this thread! This is a burden on non-webrtc users
            of data channel. My understanding is data channels are meant
            to be used by non-webrtc as a general protocol. Please
            correct me if my understanding is not correct.</p>
        </div>
      </div>
    </blockquote>
    <br>
    Your understanding may be right or wrong, but it's different from
    mine.<br>
    <br>
    Data channels are created to be useful for WebRTC. That's their
    purpose in being created, so WebRTC requirements take precedence
    over all other requirements.<br>
    <br>
    NDATA makes them more useful for WebRTC (or, as Martin put it,
    "without NDATA, they're not fit for purpose").<br>
    <br>
    Optional NDATA makes things more complex for WebRTC.<br>
    <br>
    Therefore, seen from a WebRTC viewpoint, speaking in the RTCWEB WG,
    I think NDATA should be mandatory.<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E5F5EBD@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
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          0in 0in 4.0pt">
          <p class="MsoListParagraph"
            style="text-indent:-.25in;mso-list:l0 level1 lfo1"><!--[if !supportLists]-->2.      
            <!--[endif]-->Making NDATA a MUST for webrtc clients? I am
            not just talking about browser, but non-browser based
            native/hybrid webrtc clients (e.g. a thin webrtc client on a
            cable set top box) as well. Browsers can implement it as a
            MUST clause, no one is stopping that. But why mandate a set
            top box (or an IoT device) implement NDATA when it just
            wants to use a single data channel only?!</p>
          <p class="MsoListParagraph">Let’s take an example: We keep
            NDATA a MUST. During SCTP setup, if NDATA is not supported
            at the peer then will the SCTP setup be abandoned? If the
            SCTP association is not abandoned then there is no point in
            making it a MUST where as in real practice it is not
            enforced as a MUST.</p>
        </div>
      </div>
    </blockquote>
    <br>
    In general, when an application connects to something that does not
    conform to the spec for the protocol he's using, "all bets are off".<br>
    <br>
    Implementations are free to handle this situation whatever way they
    want.<br>
    <br>
    Note: So far, I've seen *zero* evidence that the burden of
    implementing NDATA is significantly higher than the total effort
    spent in writing updates on this thread. For 90+% of all
    implementors, not implementing NDATA is a larger effort than
    implementing it, because the libraries they use will support it, so
    they have to take action to take it out.<br>
    <br>
  </body>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

>> I am not suggesting different flavors - I am suggesting to make the usag=
e of interleaving optional. Other SCTP features are also optional (includin=
g reliability, ordering etc).
>
> Where are these things optional? According to
> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section-6.1
> an implementation MUST implement PR-SCTP and the support of the FORWARD-T=
SN chunk will be negotiated similar to the NDATA chunk. I don't see a diffe=
rence, especially not if we move to MUST support NDATA.

For example, section 6.1 says:

	"The partial reliability extension defined in [RFC3758] MUST be supported.=
"

However, in CLUE we say that a CLUE entity MUST NOT *use* the extension, as=
 all CLUE messages are required to be sent reliably.

Regards,

Christer


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On 28 Oct 2014, at 14:51, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> I am not suggesting different flavors - I am suggesting to make the =
usage of interleaving optional. Other SCTP features are also optional =
(including reliability, ordering etc).
>>=20
>> Where are these things optional? According to
>> =
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section-6.1
>> an implementation MUST implement PR-SCTP and the support of the =
FORWARD-TSN chunk will be negotiated similar to the NDATA chunk. I don't =
see a difference, especially not if we move to MUST support NDATA.
>=20
> For example, section 6.1 says:
>=20
> 	"The partial reliability extension defined in [RFC3758] MUST be =
supported."
>=20
> However, in CLUE we say that a CLUE entity MUST NOT *use* the =
extension, as all CLUE messages are required to be sent reliably.
That is OK. However, support for PR-SCTP MUST be there and it will be =
negotiated. You just never
send a message with partial reliability. At least this is my =
understanding...

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi,

>>> I am not suggesting different flavors - I am suggesting to make the usa=
ge of=20
>>> interleaving optional. Other SCTP features are also optional (including=
 reliability, ordering etc).
>>>=20
>>> Where are these things optional? According to
>>> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
>>> -6.1 an implementation MUST implement PR-SCTP and the support of the=20
>>> FORWARD-TSN chunk will be negotiated similar to the NDATA chunk. I don'=
t see a difference, especially not if we move to MUST support NDATA.
>>=20
>> For example, section 6.1 says:
>>=20
>> 	"The partial reliability extension defined in [RFC3758] MUST be support=
ed."
>>=20
>> However, in CLUE we say that a CLUE entity MUST NOT *use* the extension,=
 as all CLUE messages are required to be sent reliably.
>
> That is OK. However, support for PR-SCTP MUST be there and it will be neg=
otiated. You just never send a message with partial reliability. At least t=
his is my understanding...

So, can the same be done for interleaving? I.e. the support MUST be there, =
but e.g. CLUE entities can choose not to use it?

Note that my issue is not about support, it's about usage :)

Regards,

Christer


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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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--f46d043bdff80285b105067e1b3d
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On Tue, Oct 28, 2014 at 8:46 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>> I am not suggesting different flavors - I am suggesting to make the
> usage of
> >>> interleaving optional. Other SCTP features are also optional
> (including reliability, ordering etc).
> >>>
> >>> Where are these things optional? According to
> >>> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
> >>> -6.1 an implementation MUST implement PR-SCTP and the support of the
> >>> FORWARD-TSN chunk will be negotiated similar to the NDATA chunk. I
> don't see a difference, especially not if we move to MUST support NDATA.
> >>
> >> For example, section 6.1 says:
> >>
> >>      "The partial reliability extension defined in [RFC3758] MUST be
> supported."
> >>
> >> However, in CLUE we say that a CLUE entity MUST NOT *use* the
> extension, as all CLUE messages are required to be sent reliably.
> >
> > That is OK. However, support for PR-SCTP MUST be there and it will be
> negotiated. You just never send a message with partial reliability. At
> least this is my understanding...
>
> So, can the same be done for interleaving? I.e. the support MUST be there,
> but e.g. CLUE entities can choose not to use it?
>
> Note that my issue is not about support, it's about usage :)


What resource is being conserved by not just requiring it?

-Ekr

--f46d043bdff80285b105067e1b3d
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Oct 28, 2014 at 8:46 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><span class=3D"">Hi,<br>
<br>
&gt;&gt;&gt; I am not suggesting different flavors - I am suggesting to mak=
e the usage of<br>
&gt;&gt;&gt; interleaving optional. Other SCTP features are also optional (=
including reliability, ordering etc).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Where are these things optional? According to<br>
&gt;&gt;&gt; <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-data-=
channel-12#section" target=3D"_blank">https://tools.ietf.org/html/draft-iet=
f-rtcweb-data-channel-12#section</a><br>
</span>&gt;&gt;&gt; -6.1 an implementation MUST implement PR-SCTP and the s=
upport of the<br>
<span class=3D"">&gt;&gt;&gt; FORWARD-TSN chunk will be negotiated similar =
to the NDATA chunk. I don&#39;t see a difference, especially not if we move=
 to MUST support NDATA.<br>
&gt;&gt;<br>
&gt;&gt; For example, section 6.1 says:<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 &quot;The partial reliability extension define=
d in [RFC3758] MUST be supported.&quot;<br>
&gt;&gt;<br>
&gt;&gt; However, in CLUE we say that a CLUE entity MUST NOT *use* the exte=
nsion, as all CLUE messages are required to be sent reliably.<br>
&gt;<br>
&gt; That is OK. However, support for PR-SCTP MUST be there and it will be =
negotiated. You just never send a message with partial reliability. At leas=
t this is my understanding...<br>
<br>
</span>So, can the same be done for interleaving? I.e. the support MUST be =
there, but e.g. CLUE entities can choose not to use it?<br>
<br>
Note that my issue is not about support, it&#39;s about usage :)</blockquot=
e><div><br></div><div>What resource is being conserved by not just requirin=
g it?</div><div><br></div><div>-Ekr</div></div></div></div>

--f46d043bdff80285b105067e1b3d--


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To: Harald Alvestrand <harald@alvestrand.no>, Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Please see my comments inline.

From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Tuesday, October 28, 2014 8:27 AM

From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Monday, October 27, 2014 5:03 PM
On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju) wrote:
+1 to make N-DATA optional.

Data channel provides a robust generic peer to peer communication channel. =
So, one can see that data channel may be used under non-webrtc contexts (li=
ke as in CLUE) as a general transport or as a replacement of native to SCTP=
 to traverse firewalls. IMHO, majority of these use cases:
a. May just involve a single data channel (with possible multiplexing on to=
p of it)
b. And may not use the DCEP either (both ends could assume same data channe=
l stream id, this is similar to SCTP PPIDs).
We already made DCEP(b) as optional.

If an app is using just a single data channel then there is no real benefit=
 in using N-DATA. Then why not make it optional? Also, as already pointed o=
ut, making it optional won't break interop as it is still negotiated during=
 SCTP setup time anyway.

Negotiation means that transport initiator gets to choose whether to use it=
 or not.
This means that a responder has no choice in the matter.

In the situation where the responder knows better than the initiator what t=
he usage is going to be, this is problematic.
<Raju>
IMHO, it is not just the responder knowing "better" which makes a scenario =
work, rather both have to agree on it. For example, if responder wants to h=
ave multiple data channels is no good if the originator does not want to su=
pport (CLUE use of data channels is an example of it). That's why, in most =
cases I know it is the originator who offers capabilities and terminator ge=
ts to choose.
</Raju>


I would like to point out that there will be some webrtc applications which=
 might just use one webrtc data channel for simplicity (and may use some mu=
ltiplexing of different types of msgs/objects on top of it) or one is just =
sufficient as the app does not send/recv huge chunks of data. So, N-DATA is=
n't needed for such webrtc clients either. Then why add burden to such webr=
tc implementations!?

You're confusing the webrtc application (what runs in the browser) with the=
 webrtc implementation (the browser itself). Optional NDATA makes the brows=
er *more* complex, since it has to expose API surface to let the applicatio=
n choose whether or not to offer NDATA, and it has to have code to decide w=
hether or not to offer NDATA.
<Raju>
I am talking about 2 items:

1.      Making NDATA a MUST data channel draft? I could be wrong, but I tho=
ught this is the main discussion point in this thread! This is a burden on =
non-webrtc users of data channel. My understanding is data channels are mea=
nt to be used by non-webrtc as a general protocol. Please correct me if my =
understanding is not correct.

Your understanding may be right or wrong, but it's different from mine.

Data channels are created to be useful for WebRTC. That's their purpose in =
being created, so WebRTC requirements take precedence over all other requir=
ements.
<Raju2>
IMHO, I don't think that is how IETF worked; you produce something which is=
 generally useful than very specific to one particular use case without giv=
ing up on too much for the original purpose. In fact, WebRTC is a huge bene=
ficiary of it!! IMHO, making data channels transport a generic mechanism is=
 part of that process.
</Raju2>

NDATA makes them more useful for WebRTC (or, as Martin put it, "without NDA=
TA, they're not fit for purpose").

Optional NDATA makes things more complex for WebRTC.

Therefore, seen from a WebRTC viewpoint, speaking in the RTCWEB WG, I think=
 NDATA should be mandatory.
<Raju2>
This is related to webrtc vs. non-webrtc use of data channels. To give data=
 channels independence (sorry, I did not mean in a dramatic way :)) from we=
brtc, can we make data channel drafts generic and compliancy aspects of it =
into a webrtc specific section/draft; this (separate draft) is how it is do=
ne for RTP transport. Such webrtc compliance for data channels can be part =
of the same data channel draft but a new webrtc specific section or in a ne=
w/existing webrtc specific draft (webrtc/data-channel overview draft). Simi=
larly CLUE specific compliance can be in a separate section of the data cha=
nnel draft.
</Raju2>



2.      Making NDATA a MUST for webrtc clients? I am not just talking about=
 browser, but non-browser based native/hybrid webrtc clients (e.g. a thin w=
ebrtc client on a cable set top box) as well. Browsers can implement it as =
a MUST clause, no one is stopping that. But why mandate a set top box (or a=
n IoT device) implement NDATA when it just wants to use a single data chann=
el only?!

Let's take an example: We keep NDATA a MUST. During SCTP setup, if NDATA is=
 not supported at the peer then will the SCTP setup be abandoned? If the SC=
TP association is not abandoned then there is no point in making it a MUST =
where as in real practice it is not enforced as a MUST.

In general, when an application connects to something that does not conform=
 to the spec for the protocol he's using, "all bets are off".

Implementations are free to handle this situation whatever way they want.
<Raju2>
This item is specific to data channel requirements for webrtc use. If a req=
uirement is a MUST then how can it be implementation specific? Then why put=
 a requirement as MUST but still allow implementations to have different be=
havior?
</Raju2>

Note: So far, I've seen *zero* evidence that the burden of implementing NDA=
TA is significantly higher than the total effort spent in writing updates o=
n this thread. For 90+% of all implementors, not implementing NDATA is a la=
rger effort than implementing it, because the libraries they use will suppo=
rt it, so they have to take action to take it out.
<Raju2>
The 90% can then always support NDATA... no issue with that. But why make i=
t a burden for rest of the 10% (these are the ones who may be constrained b=
y various resources)? IMHO, we don't add MORE work for an implementation ju=
st because it already has LOTs of work to do anyway! It should be other way=
 around, since there is so much stuff mandatory already why not go easy on =
the stuff which may not be needed by some applications!!?
</Raju2>

BR
Raju

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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">Please see my comments in=
line.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
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<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal" style=3D"margin-left:5.25pt"><b><span style=3D"font-=
size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:win=
dowtext">From:</span></b><span style=3D"font-size:10.0pt;font-family:&quot;=
Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> Harald Alvestrand [m=
ailto:harald@alvestrand.no]
<br>
<b>Sent:</b> Tuesday, October 28, 2014 8:27 AM<br>
<br>
</span><o:p></o:p></p>
</div>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">From: Harald Alvestra=
nd [<a href=3D"mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>=
]
<br>
Sent: Monday, October 27, 2014 5:03 PM<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 10/27/2014 01:31 PM, Makaraju, Maridi Raju (Raju)=
 wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal">&#43;1 to make N-DATA optional.<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">Data channel provides a robust generic peer to peer =
communication channel. So, one can see that data channel may be used under =
non-webrtc contexts (like as in CLUE) as a general transport or as a replac=
ement of native to SCTP to traverse
 firewalls. IMHO, majority of these use cases:<o:p></o:p></p>
<p class=3D"MsoNormal">a. May just involve a single data channel (with poss=
ible multiplexing on top of it)
<o:p></o:p></p>
<p class=3D"MsoNormal">b. And may not use the DCEP either (both ends could =
assume same data channel stream id, this is similar to SCTP PPIDs).
<o:p></o:p></p>
<p class=3D"MsoNormal">We already made DCEP(b) as optional.<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">If an app is using just a single data channel then t=
here is no real benefit in using N-DATA. Then why not make it optional? Als=
o, as already pointed out, making it optional won&#8217;t break interop as =
it is still negotiated during SCTP setup
 time anyway.<o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><br>
Negotiation means that transport initiator gets to choose whether to use it=
 or not.<br>
This means that a responder has no choice in the matter.<br>
<br>
In the situation where the responder knows better than the initiator what t=
he usage is going to be, this is problematic.<o:p></o:p></p>
<p class=3D"MsoNormal">&lt;Raju&gt;<o:p></o:p></p>
<p class=3D"MsoNormal">IMHO, it is not just the responder knowing &#8220;be=
tter&#8221; which makes a scenario work, rather both have to agree on it. F=
or example, if responder wants to have multiple data channels is no good if=
 the originator does not want to support (CLUE
 use of data channels is an example of it). That&#8217;s why, in most cases=
 I know it is the originator who offers capabilities and terminator gets to=
 choose.
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&lt;/Raju&gt;<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">I would like to point out that there will be some we=
brtc applications which might just use one webrtc data channel for simplici=
ty (and may use some multiplexing of different types of msgs/objects on top=
 of it) or one is just sufficient
 as the app does not send/recv huge chunks of data. So, N-DATA isn&#8217;t =
needed for such webrtc clients either. Then why add burden to such webrtc i=
mplementations!?<o:p></o:p></p>
<p class=3D"MsoNormal"><br>
You're confusing the webrtc application (what runs in the browser) with the=
 webrtc implementation (the browser itself). Optional NDATA makes the brows=
er *more* complex, since it has to expose API surface to let the applicatio=
n choose whether or not to offer
 NDATA, and it has to have code to decide whether or not to offer NDATA.<o:=
p></o:p></p>
<p class=3D"MsoNormal">&lt;Raju&gt;<o:p></o:p></p>
<p class=3D"MsoNormal">I am talking about 2 items:<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">1.<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Making NDATA a MUST data channel draft? I could be =
wrong, but I thought this is the main discussion point in this thread! This=
 is a burden on non-webrtc users of data channel. My understanding is data =
channels are meant to be used by
 non-webrtc as a general protocol. Please correct me if my understanding is=
 not correct.<o:p></o:p></p>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
Your understanding may be right or wrong, but it's different from mine.<br>
<br>
Data channels are created to be useful for WebRTC. That's their purpose in =
being created, so WebRTC requirements take precedence over all other requir=
ements.<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">IMHO, I don&#8217;t th=
ink that is how IETF worked; you produce something which is generally usefu=
l than very specific to one particular use case without giving up on too mu=
ch for the original purpose. In fact, WebRTC
 is a huge beneficiary of it!! IMHO, making data channels transport a gener=
ic mechanism is part of that process.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<br>
</span></i></b><br>
NDATA makes them more useful for WebRTC (or, as Martin put it, &quot;withou=
t NDATA, they're not fit for purpose&quot;).<br>
<br>
Optional NDATA makes things more complex for WebRTC.<br>
<br>
Therefore, seen from a WebRTC viewpoint, speaking in the RTCWEB WG, I think=
 NDATA should be mandatory.<span style=3D"color:#1F497D"><o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">This is related to web=
rtc vs. non-webrtc use of data channels. To give data channels independence=
 (sorry, I did not mean in a dramatic way
</span><span style=3D"font-family:Wingdings;color:#1F497D">J</span><span st=
yle=3D"color:#1F497D">) from webrtc, can we make data channel drafts generi=
c and compliancy aspects of it into a webrtc specific section/draft; this (=
separate draft) is how it is done for
 RTP transport. Such webrtc compliance for data channels can be part of the=
 same data channel draft but a new webrtc specific section or in a new/exis=
ting webrtc specific draft (webrtc/data-channel overview draft). Similarly =
CLUE specific compliance can be
 in a separate section of the data channel draft.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<br>
</span></i></b><br>
<br>
<o:p></o:p></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">2.<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Making NDATA a MUST for webrtc clients? I am not ju=
st talking about browser, but non-browser based native/hybrid webrtc client=
s (e.g. a thin webrtc client on a cable set top box) as well. Browsers can =
implement it as a MUST clause, no
 one is stopping that. But why mandate a set top box (or an IoT device) imp=
lement NDATA when it just wants to use a single data channel only?!<o:p></o=
:p></p>
<p class=3D"MsoListParagraph">Let&#8217;s take an example: We keep NDATA a =
MUST. During SCTP setup, if NDATA is not supported at the peer then will th=
e SCTP setup be abandoned? If the SCTP association is not abandoned then th=
ere is no point in making it a MUST where
 as in real practice it is not enforced as a MUST.<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
In general, when an application connects to something that does not conform=
 to the spec for the protocol he's using, &quot;all bets are off&quot;.<br>
<br>
Implementations are free to handle this situation whatever way they want.<s=
pan style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">This item is specif=
ic to data channel requirements for webrtc use. If a requirement is a MUST =
then how can it be implementation specific? Then why put
 a requirement as MUST but still allow implementations to have different be=
havior?
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Note: So far, I've seen *zero* evidence that the burden of implementing NDA=
TA is significantly higher than the total effort spent in writing updates o=
n this thread. For 90&#43;% of all implementors, not implementing NDATA is =
a larger effort than implementing it,
 because the libraries they use will support it, so they have to take actio=
n to take it out.<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The 90% can then al=
ways support NDATA&#8230; no issue with that. But why make it a burden for =
rest of the 10% (these are the ones who may be constrained by
 various resources)? IMHO, we don&#8217;t add MORE work for an implementati=
on just because it already has LOTs of work to do anyway! It should be othe=
r way around, since there is so much stuff mandatory already why not go eas=
y on the stuff which may not be needed
 by some applications!!?<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span=
></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju</span></i></b>=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D"><o:p></o:p></span></p>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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(This thread has gotten long, and its hard to figure where to jump in. 
I'm just replying to the last message I see, but am commenting on the 
thread in total.)

Regarding data channels being webrtc-specific:

I don't see it that way. Webrtc has opened up restricted communication 
among browsers. If you want to talk to a browser, you have few choices, 
and this is the one that is "general purpose". So any application that 
sometimes might need to run in a browser is a likely candidate for data 
channels. And then, when non-browser versions of that app are talking to 
one another, it will be most practical to use the same mechanism.

That is why CLUE is using data channels - even if we think that *most* 
of the endpoints won't be in browsers.

Regarding NDATA:

ISTM that SCTP just screwed up in not including the NDATA features in 
the base protocol. Going forward, it always should be included in new 
implementations. Since data channels are being defined after NDATA, it 
is perfectly reasonable for data channels to require NDATA support.

IIUC, while NDATA allows fragmentation to avoid head of line blocking, 
that doesn't mean the messages will be fragmented unnecessarily. If you 
are only using one channel I *think* it likely that even a large message 
won't be fragmented. So it will be harmless insurance.

Christer said that since CLUE only needs one data channel there is no 
need for NDATA. Someone specifying BFCP over data channel might reach a 
similar conclusion for it. But there is nothing to prevent an 
application that implements both CLUE and BFCP. Independently they 
wouldn't need NDATA, but together they would.

So I am in favor of making NDATA mandatory to implement, offer, and 
accept for data channels.

But I would still like the data channel spec to acknowledge that, while 
webrtc is a *very* important client, that the protocol solely for 
webrtc. And with that in mind I would like to see "webrtc" removed from 
the name of the protocol. (E.g., in the SDP m-line.)

	Thanks,
	Paul

On 10/28/14 1:53 PM, Christer Holmberg wrote:
> Hi,
>
>      >>> I am not suggesting different flavors - I am suggesting to make
>     the usage of
>      >>> interleaving optional. Other SCTP features are also optional
>     (including reliability, ordering etc).
>      >>>
>      >>> Where are these things optional? According to
>      >>>
>     https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
>      >>> -6.1 an implementation MUST implement PR-SCTP and the support
>     of the
>      >>> FORWARD-TSN chunk will be negotiated similar to the NDATA
>     chunk. I don't see a difference, especially not if we move to MUST
>     support NDATA.
>      >>
>      >> For example, section 6.1 says:
>      >>
>      >>      "The partial reliability extension defined in [RFC3758]
>     MUST be supported."
>      >>
>      >> However, in CLUE we say that a CLUE entity MUST NOT *use* the
>     extension, as all CLUE messages are required to be sent reliably.
>      >
>      > That is OK. However, support for PR-SCTP MUST be there and it
>     will be negotiated. You just never send a message with partial
>     reliability. At least this is my understanding...
>
>     So, can the same be done for interleaving? I.e. the support MUST be
>     there, but e.g. CLUE entities can choose not to use it?
>
>     Note that my issue is not about support, it's about usage :)
>
> What resource is being conserved by not just requiring it?
>
> I don’t know. The task is to figure out whether there are *technical*
> reasons to mandate usage of certain SCTP features for CLUE.
>
> Regards,
>
> Christer
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Eric Rescorla <ekr@rtfm.com>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, "Harald Alvestrand" <harald@alvestrand.no>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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--_000_D07572143D0B4mzanatyciscocom_
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> What resource is being conserved by not just requiring it?

8 bytes per packet (for MID+FSN).

Michael, the current NDATA draft requires it to be used for all messages if=
 negotiated. Can=92t this be relaxed / optimized to use it for all *fragmen=
ted* messages but use DATA for non-fragmented messages to avoid the 8 byte =
overhead for small messages (like game telemetry/control, an often cited us=
e of data channels).

The current NDATA draft also says you can=92t interleave fragmented message=
s in a stream, so head of line blocking remains for all fragmented messages=
, while only small non-fragmented messages can avoid it. This dilutes the u=
tility of NDATA, perhaps enough to make apps that really care about head of=
 line blocking to implement their own app layer fragmentation with messages=
 well below common MTUs, hence defeating NDATA. Can=92t this also be relaxe=
d since MID is part of the NDATA extra header?

If those two issues are resolved, I see no argument against making NDATA a =
MUST in data channels.

Mo

PS. Some believe data channels will be more widely used than WebRTC media (=
webtorrent, etc.), so it does make sense to consider the desirable properti=
es of a general peer-to-peer transport, and drop the WebRTC prefix when tal=
king about data channels.

--_000_D07572143D0B4mzanatyciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; font-family: Arial, sans-serif; font-size: 12px=
; color: rgb(0, 0, 0);">
<div>&gt; What resource is being conserved by not just requiring it?</div>
<div><br>
</div>
<div>8 bytes per packet (for MID&#43;FSN).</div>
<div><br>
</div>
<div>Michael, the current NDATA draft requires it to be used for all messag=
es if negotiated. Can=92t this be relaxed / optimized to use it for all *fr=
agmented* messages but use DATA for non-fragmented messages to avoid the 8 =
byte overhead for small messages (like
 game telemetry/control, an often cited use of data channels).</div>
<div><br>
</div>
<div>The current NDATA draft also says you can=92t interleave fragmented me=
ssages in a stream, so head of line blocking remains for all fragmented mes=
sages, while only small non-fragmented messages can avoid it. This dilutes =
the utility of NDATA, perhaps enough
 to make apps that really care about head of line blocking to implement the=
ir own app layer fragmentation with messages well below common MTUs, hence =
defeating NDATA. Can=92t this also be relaxed since MID is part of the NDAT=
A extra header?</div>
<div><br>
</div>
<div>If those two issues are resolved, I see no argument against making NDA=
TA a MUST in data channels.</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<div>PS. Some believe data channels will be more widely used than WebRTC me=
dia (webtorrent, etc.), so it does make sense to consider the desirable pro=
perties of a general peer-to-peer transport, and drop the WebRTC prefix whe=
n talking about data channels.</div>
</body>
</html>

--_000_D07572143D0B4mzanatyciscocom_--


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To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

> > What resource is being conserved by not just requiring it?
>=20
> 8 bytes per packet (for MID+FSN).
>=20
> Michael, the current NDATA draft requires it to be used for all =
messages if negotiated. Can=92t this be relaxed / optimized to use it =
for all *fragmented* messages but use DATA for non-fragmented messages =
to avoid the 8 byte overhead for small messages (like game =
telemetry/control, an often cited use of data channels).
You can bring this up at the tsvwg mailing list. However, we normally do =
not optimize for byte saving... Mixing
both makes the processing harder... Maybe we can use a bit in the flags =
to indicate if the additional fields are
there...
>=20
> The current NDATA draft also says you can=92t interleave fragmented =
messages in a stream, so head of line blocking remains for all =
fragmented messages, while only small non-fragmented messages can avoid =
it. This dilutes the utility of NDATA, perhaps enough to make apps that =
really care about head of line blocking to implement their own app layer =
fragmentation with messages well below common MTUs, hence defeating =
NDATA. Can=92t this also be relaxed since MID is part of the NDATA extra =
header?
This would only make sense for messages marked as unordered... For =
ordered it doesn't make sense.
>=20
> If those two issues are resolved, I see no argument against making =
NDATA a MUST in data channels.
Please let us discuss NDATA issues on the tsvwg mailing list...

Best regards
Michael
>=20
> Mo
>=20
> PS. Some believe data channels will be more widely used than WebRTC =
media (webtorrent, etc.), so it does make sense to consider the =
desirable properties of a general peer-to-peer transport, and drop the =
WebRTC prefix when talking about data channels.


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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Mary pointed out an important typo in this message. Correction inline.

On 10/28/14 4:13 PM, Paul Kyzivat wrote:
> (This thread has gotten long, and its hard to figure where to jump in.
> I'm just replying to the last message I see, but am commenting on the
> thread in total.)
>
> Regarding data channels being webrtc-specific:
>
> I don't see it that way. Webrtc has opened up restricted communication
> among browsers. If you want to talk to a browser, you have few choices,
> and this is the one that is "general purpose". So any application that
> sometimes might need to run in a browser is a likely candidate for data
> channels. And then, when non-browser versions of that app are talking to
> one another, it will be most practical to use the same mechanism.
>
> That is why CLUE is using data channels - even if we think that *most*
> of the endpoints won't be in browsers.
>
> Regarding NDATA:
>
> ISTM that SCTP just screwed up in not including the NDATA features in
> the base protocol. Going forward, it always should be included in new
> implementations. Since data channels are being defined after NDATA, it
> is perfectly reasonable for data channels to require NDATA support.
>
> IIUC, while NDATA allows fragmentation to avoid head of line blocking,
> that doesn't mean the messages will be fragmented unnecessarily. If you
> are only using one channel I *think* it likely that even a large message
> won't be fragmented. So it will be harmless insurance.
>
> Christer said that since CLUE only needs one data channel there is no
> need for NDATA. Someone specifying BFCP over data channel might reach a
> similar conclusion for it. But there is nothing to prevent an
> application that implements both CLUE and BFCP. Independently they
> wouldn't need NDATA, but together they would.
>
> So I am in favor of making NDATA mandatory to implement, offer, and
> accept for data channels.
>
> But I would still like the data channel spec to acknowledge that, while
> webrtc is a *very* important client, that the protocol solely for
> webrtc. And with that in mind I would like to see "webrtc" removed from
> the name of the protocol. (E.g., in the SDP m-line.)

What I meant to say in the last paragraph is:

But I would still like the data channel spec to acknowledge that, while 
webrtc is a *very* important client, that the protocol *is not* solely 
for webrtc. And with that in mind I would like to see "webrtc" removed 
from the name of the protocol. (E.g., in the SDP m-line.)

	Thanks,
	Paul


> On 10/28/14 1:53 PM, Christer Holmberg wrote:
>> Hi,
>>
>>      >>> I am not suggesting different flavors - I am suggesting to make
>>     the usage of
>>      >>> interleaving optional. Other SCTP features are also optional
>>     (including reliability, ordering etc).
>>      >>>
>>      >>> Where are these things optional? According to
>>      >>>
>>     https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
>>      >>> -6.1 an implementation MUST implement PR-SCTP and the support
>>     of the
>>      >>> FORWARD-TSN chunk will be negotiated similar to the NDATA
>>     chunk. I don't see a difference, especially not if we move to MUST
>>     support NDATA.
>>      >>
>>      >> For example, section 6.1 says:
>>      >>
>>      >>      "The partial reliability extension defined in [RFC3758]
>>     MUST be supported."
>>      >>
>>      >> However, in CLUE we say that a CLUE entity MUST NOT *use* the
>>     extension, as all CLUE messages are required to be sent reliably.
>>      >
>>      > That is OK. However, support for PR-SCTP MUST be there and it
>>     will be negotiated. You just never send a message with partial
>>     reliability. At least this is my understanding...
>>
>>     So, can the same be done for interleaving? I.e. the support MUST be
>>     there, but e.g. CLUE entities can choose not to use it?
>>
>>     Note that my issue is not about support, it's about usage :)
>>
>> What resource is being conserved by not just requiring it?
>>
>> I don’t know. The task is to figure out whether there are *technical*
>> reasons to mandate usage of certain SCTP features for CLUE.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
Thread-Index: AQHP8vEmqS2/U13JNkqofeKaZ1Gl3pxGVXUA///BxDI=
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com>, <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de>
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On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" <Michael.Tuexen@lurchi.franke=
n.de> wrote:

On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

>> What resource is being conserved by not just requiring it?
>=20
> 8 bytes per packet (for MID+FSN).
>=20
> Michael, the current NDATA draft requires it to be used for all messages =
if negotiated. Can=92t this be relaxed / optimized to use it for all *fragm=
ented* messages but use DATA for non-fragmented messages to avoid the 8 byt=
e overhead for small messages (like game telemetry/control, an often cited =
use of data channels).
You can bring this up at the tsvwg mailing list.

Mo: Will do.

However, we normally do not optimize for byte saving... Mixing
both makes the processing harder... Maybe we can use a bit in the flags to =
indicate if the additional fields are
there...

Mo: Type should already be enough without flags.

>=20
> The current NDATA draft also says you can=92t interleave fragmented messa=
ges in a stream, so head of line blocking remains for all fragmented messag=
es, while only small non-fragmented messages can avoid it. This dilutes the=
 utility of NDATA, perhaps enough to make apps that really care about head =
of line blocking to implement their own app layer fragmentation with messag=
es well below common MTUs, hence defeating NDATA. Can=92t this also be rela=
xed since MID is part of the NDATA extra header?
This would only make sense for messages marked as unordered... For ordered =
it doesn't make sense.

Mo: The restriction applies to unordered as well as ordered. Even for order=
ed, it does make sense to avoid head of line blocking at the sender.=20
>=20
> If those two issues are resolved, I see no argument against making NDATA =
a MUST in data channels.
Please let us discuss NDATA issues on the tsvwg mailing list...

Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its benefits=
 outweigh its costs. The current draft does not pass that bar for me. If my=
 app cares about HOL blocking, NDATA is not very useful, so my app must int=
ernally fragment, hence NDATA actually becomes harmful (8 bytes closer to e=
xceeding MTU, forcing SCTP fragments and hence risking HOL blocking).

Best regards
Michael
>=20
> Mo
>=20
> PS. Some believe data channels will be more widely used than WebRTC media=
 (webtorrent, etc.), so it does make sense to consider the desirable proper=
ties of a general peer-to-peer transport, and drop the WebRTC prefix when t=
alking about data channels.


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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <544FF8CE.5070304@alum.mit.edu> <5450153B.8000808@alum.mit.edu>
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+1 for removing webrtc from data channel and making it clearer that it 
can be used as a general mechanism. That's why I thought there should be 
a separate ALPN in draft-ietf-rtcweb-alpn-00 for data channel only usage 
rather than saying SRTP may be absent.

Regards, Christian

On 29/10/2014 9:14 AM, Paul Kyzivat wrote:
> Mary pointed out an important typo in this message. Correction inline.
>
> On 10/28/14 4:13 PM, Paul Kyzivat wrote:
>> (This thread has gotten long, and its hard to figure where to jump in.
>> I'm just replying to the last message I see, but am commenting on the
>> thread in total.)
>>
>> Regarding data channels being webrtc-specific:
>>
>> I don't see it that way. Webrtc has opened up restricted communication
>> among browsers. If you want to talk to a browser, you have few choices,
>> and this is the one that is "general purpose". So any application that
>> sometimes might need to run in a browser is a likely candidate for data
>> channels. And then, when non-browser versions of that app are talking to
>> one another, it will be most practical to use the same mechanism.
>>
>> That is why CLUE is using data channels - even if we think that *most*
>> of the endpoints won't be in browsers.
>>
>> Regarding NDATA:
>>
>> ISTM that SCTP just screwed up in not including the NDATA features in
>> the base protocol. Going forward, it always should be included in new
>> implementations. Since data channels are being defined after NDATA, it
>> is perfectly reasonable for data channels to require NDATA support.
>>
>> IIUC, while NDATA allows fragmentation to avoid head of line blocking,
>> that doesn't mean the messages will be fragmented unnecessarily. If you
>> are only using one channel I *think* it likely that even a large message
>> won't be fragmented. So it will be harmless insurance.
>>
>> Christer said that since CLUE only needs one data channel there is no
>> need for NDATA. Someone specifying BFCP over data channel might reach a
>> similar conclusion for it. But there is nothing to prevent an
>> application that implements both CLUE and BFCP. Independently they
>> wouldn't need NDATA, but together they would.
>>
>> So I am in favor of making NDATA mandatory to implement, offer, and
>> accept for data channels.
>>
>> But I would still like the data channel spec to acknowledge that, while
>> webrtc is a *very* important client, that the protocol solely for
>> webrtc. And with that in mind I would like to see "webrtc" removed from
>> the name of the protocol. (E.g., in the SDP m-line.)
>
> What I meant to say in the last paragraph is:
>
> But I would still like the data channel spec to acknowledge that, 
> while webrtc is a *very* important client, that the protocol *is not* 
> solely for webrtc. And with that in mind I would like to see "webrtc" 
> removed from the name of the protocol. (E.g., in the SDP m-line.)
>
>     Thanks,
>     Paul
>
>
>> On 10/28/14 1:53 PM, Christer Holmberg wrote:
>>> Hi,
>>>
>>>      >>> I am not suggesting different flavors - I am suggesting to 
>>> make
>>>     the usage of
>>>      >>> interleaving optional. Other SCTP features are also optional
>>>     (including reliability, ordering etc).
>>>      >>>
>>>      >>> Where are these things optional? According to
>>>      >>>
>>> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
>>>      >>> -6.1 an implementation MUST implement PR-SCTP and the support
>>>     of the
>>>      >>> FORWARD-TSN chunk will be negotiated similar to the NDATA
>>>     chunk. I don't see a difference, especially not if we move to MUST
>>>     support NDATA.
>>>      >>
>>>      >> For example, section 6.1 says:
>>>      >>
>>>      >>      "The partial reliability extension defined in [RFC3758]
>>>     MUST be supported."
>>>      >>
>>>      >> However, in CLUE we say that a CLUE entity MUST NOT *use* the
>>>     extension, as all CLUE messages are required to be sent reliably.
>>>      >
>>>      > That is OK. However, support for PR-SCTP MUST be there and it
>>>     will be negotiated. You just never send a message with partial
>>>     reliability. At least this is my understanding...
>>>
>>>     So, can the same be done for interleaving? I.e. the support MUST be
>>>     there, but e.g. CLUE entities can choose not to use it?
>>>
>>>     Note that my issue is not about support, it's about usage :)
>>>
>>> What resource is being conserved by not just requiring it?
>>>
>>> I don’t know. The task is to figure out whether there are *technical*
>>> reasons to mandate usage of certain SCTP features for CLUE.
>>>
>>> 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
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christian Groves <Christian.Groves@nteczone.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <544FF8CE.5070304@alum.mit.edu> <5450153B.8000808@alum.mit.edu> <54501EA8.9010308@nteczone.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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+1 for making data channel as webrtc independent.
+1 for making NDATA a MUST for webrtc use of data channels only.
   As we have seen in recent posts NDATA is not risk free and has "baggage"=
 associated to it.
My detailed view is explained in the post:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg13322.html

BR
Raju


> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christian Grov=
es
> Sent: Tuesday, October 28, 2014 5:55 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
>=20
> +1 for removing webrtc from data channel and making it clearer that it
> can be used as a general mechanism. That's why I thought there should be
> a separate ALPN in draft-ietf-rtcweb-alpn-00 for data channel only usage
> rather than saying SRTP may be absent.
>=20
> Regards, Christian
>=20
> On 29/10/2014 9:14 AM, Paul Kyzivat wrote:
> > Mary pointed out an important typo in this message. Correction inline.
> >
> > On 10/28/14 4:13 PM, Paul Kyzivat wrote:
> >> (This thread has gotten long, and its hard to figure where to jump in.
> >> I'm just replying to the last message I see, but am commenting on the
> >> thread in total.)
> >>
> >> Regarding data channels being webrtc-specific:
> >>
> >> I don't see it that way. Webrtc has opened up restricted communication
> >> among browsers. If you want to talk to a browser, you have few choices=
,
> >> and this is the one that is "general purpose". So any application that
> >> sometimes might need to run in a browser is a likely candidate for dat=
a
> >> channels. And then, when non-browser versions of that app are talking =
to
> >> one another, it will be most practical to use the same mechanism.
> >>
> >> That is why CLUE is using data channels - even if we think that *most*
> >> of the endpoints won't be in browsers.
> >>
> >> Regarding NDATA:
> >>
> >> ISTM that SCTP just screwed up in not including the NDATA features in
> >> the base protocol. Going forward, it always should be included in new
> >> implementations. Since data channels are being defined after NDATA, it
> >> is perfectly reasonable for data channels to require NDATA support.
> >>
> >> IIUC, while NDATA allows fragmentation to avoid head of line blocking,
> >> that doesn't mean the messages will be fragmented unnecessarily. If yo=
u
> >> are only using one channel I *think* it likely that even a large messa=
ge
> >> won't be fragmented. So it will be harmless insurance.
> >>
> >> Christer said that since CLUE only needs one data channel there is no
> >> need for NDATA. Someone specifying BFCP over data channel might reach =
a
> >> similar conclusion for it. But there is nothing to prevent an
> >> application that implements both CLUE and BFCP. Independently they
> >> wouldn't need NDATA, but together they would.
> >>
> >> So I am in favor of making NDATA mandatory to implement, offer, and
> >> accept for data channels.
> >>
> >> But I would still like the data channel spec to acknowledge that, whil=
e
> >> webrtc is a *very* important client, that the protocol solely for
> >> webrtc. And with that in mind I would like to see "webrtc" removed fro=
m
> >> the name of the protocol. (E.g., in the SDP m-line.)
> >
> > What I meant to say in the last paragraph is:
> >
> > But I would still like the data channel spec to acknowledge that,
> > while webrtc is a *very* important client, that the protocol *is not*
> > solely for webrtc. And with that in mind I would like to see "webrtc"
> > removed from the name of the protocol. (E.g., in the SDP m-line.)
> >
> >     Thanks,
> >     Paul
> >
> >
> >> On 10/28/14 1:53 PM, Christer Holmberg wrote:
> >>> Hi,
> >>>
> >>>      >>> I am not suggesting different flavors - I am suggesting to
> >>> make
> >>>     the usage of
> >>>      >>> interleaving optional. Other SCTP features are also optional
> >>>     (including reliability, ordering etc).
> >>>      >>>
> >>>      >>> Where are these things optional? According to
> >>>      >>>
> >>> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-12#section
> >>>      >>> -6.1 an implementation MUST implement PR-SCTP and the suppor=
t
> >>>     of the
> >>>      >>> FORWARD-TSN chunk will be negotiated similar to the NDATA
> >>>     chunk. I don't see a difference, especially not if we move to MUS=
T
> >>>     support NDATA.
> >>>      >>
> >>>      >> For example, section 6.1 says:
> >>>      >>
> >>>      >>      "The partial reliability extension defined in [RFC3758]
> >>>     MUST be supported."
> >>>      >>
> >>>      >> However, in CLUE we say that a CLUE entity MUST NOT *use* the
> >>>     extension, as all CLUE messages are required to be sent reliably.
> >>>      >
> >>>      > That is OK. However, support for PR-SCTP MUST be there and it
> >>>     will be negotiated. You just never send a message with partial
> >>>     reliability. At least this is my understanding...
> >>>
> >>>     So, can the same be done for interleaving? I.e. the support MUST =
be
> >>>     there, but e.g. CLUE entities can choose not to use it?
> >>>
> >>>     Note that my issue is not about support, it's about usage :)
> >>>
> >>> What resource is being conserved by not just requiring it?
> >>>
> >>> I don't know. The task is to figure out whether there are *technical*
> >>> reasons to mandate usage of certain SCTP features for CLUE.
> >>>
> >>> 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
> >>
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Oct 28 18:27:53 2014
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
Thread-Index: AQHP8xeLiqONAZMCbkeGY9lGqqITSg==
Date: Wed, 29 Oct 2014 01:27:46 +0000
Message-ID: <D075BA5D.3D254%mzanaty@cisco.com>
References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com> <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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--_000_D075BA5D3D254mzanatyciscocom_
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Moved to tsvwg:
http://www.ietf.org/mail-archive/web/tsvwg/current/msg12892.html

On 10/28/14, 6:28 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mailto:mzanaty=
@cisco.com>> wrote:
On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" <Michael.Tuexen@lurchi.franke=
n.de<mailto:Michael.Tuexen@lurchi.franken.de>> wrote:
On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mailto:mza=
naty@cisco.com>> wrote:
What resource is being conserved by not just requiring it?
8 bytes per packet (for MID+FSN).
Michael, the current NDATA draft requires it to be used for all messages if=
 negotiated. Can=92t this be relaxed / optimized to use it for all *fragmen=
ted* messages but use DATA for non-fragmented messages to avoid the 8 byte =
overhead for small messages (like game telemetry/control, an often cited us=
e of data channels).
You can bring this up at the tsvwg mailing list.
Mo: Will do.
However, we normally do not optimize for byte saving... Mixing
both makes the processing harder... Maybe we can use a bit in the flags to =
indicate if the additional fields are
there...
Mo: Type should already be enough without flags.
The current NDATA draft also says you can=92t interleave fragmented message=
s in a stream, so head of line blocking remains for all fragmented messages=
, while only small non-fragmented messages can avoid it. This dilutes the u=
tility of NDATA, perhaps enough to make apps that really care about head of=
 line blocking to implement their own app layer fragmentation with messages=
 well below common MTUs, hence defeating NDATA. Can=92t this also be relaxe=
d since MID is part of the NDATA extra header?
This would only make sense for messages marked as unordered... For ordered =
it doesn't make sense.
Mo: The restriction applies to unordered as well as ordered. Even for order=
ed, it does make sense to avoid head of line blocking at the sender.
If those two issues are resolved, I see no argument against making NDATA a =
MUST in data channels.
Please let us discuss NDATA issues on the tsvwg mailing list...
Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its benefits=
 outweigh its costs. The current draft does not pass that bar for me. If my=
 app cares about HOL blocking, NDATA is not very useful, so my app must int=
ernally fragment, hence NDATA actually becomes harmful (8 bytes closer to e=
xceeding MTU, forcing SCTP fragments and hence risking HOL blocking).
Best regards
Michael
Mo
PS. Some believe data channels will be more widely used than WebRTC media (=
webtorrent, etc.), so it does make sense to consider the desirable properti=
es of a general peer-to-peer transport, and drop the WebRTC prefix when tal=
king about data channels.


--_000_D075BA5D3D254mzanatyciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; font-family: Arial, sans-serif; font-size: 12px=
; color: rgb(0, 0, 0);">
<div>Moved to tsvwg:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/tsvwg/current/msg12892=
.html">http://www.ietf.org/mail-archive/web/tsvwg/current/msg12892.html</a>=
</div>
<div><br>
</div>
<div>On 10/28/14, 6:28 PM, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:mzanat=
y@cisco.com">mzanaty@cisco.com</a>&gt; wrote:</div>
<div>On Oct 28, 2014, at 5:11 PM, &quot;Michael Tuexen&quot; &lt;<a href=3D=
"mailto:Michael.Tuexen@lurchi.franken.de">Michael.Tuexen@lurchi.franken.de<=
/a>&gt; wrote:</div>
<div>On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:mz=
anaty@cisco.com">mzanaty@cisco.com</a>&gt; wrote:</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>What resource is being conserved by not just requiring it?</div>
</blockquote>
<div></div>
<div>8 bytes per packet (for MID&#43;FSN).</div>
<div></div>
<div>Michael, the current NDATA draft requires it to be used for all messag=
es if negotiated. Can=92t this be relaxed / optimized to use it for all *fr=
agmented* messages but use DATA for non-fragmented messages to avoid the 8 =
byte overhead for small messages (like
 game telemetry/control, an often cited use of data channels).</div>
</blockquote>
<div>You can bring this up at the tsvwg mailing list.</div>
<div>Mo: Will do.</div>
<div>However, we normally do not optimize for byte saving... Mixing</div>
<div>both makes the processing harder... Maybe we can use a bit in the flag=
s to indicate if the additional fields are</div>
<div>there...</div>
<div>Mo: Type should already be enough without flags.</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>The current NDATA draft also says you can=92t interleave fragmented me=
ssages in a stream, so head of line blocking remains for all fragmented mes=
sages, while only small non-fragmented messages can avoid it. This dilutes =
the utility of NDATA, perhaps enough
 to make apps that really care about head of line blocking to implement the=
ir own app layer fragmentation with messages well below common MTUs, hence =
defeating NDATA. Can=92t this also be relaxed since MID is part of the NDAT=
A extra header?</div>
</blockquote>
<div>This would only make sense for messages marked as unordered... For ord=
ered it doesn't make sense.</div>
<div>Mo: The restriction applies to unordered as well as ordered. Even for =
ordered, it does make sense to avoid head of line blocking at the sender.
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>If those two issues are resolved, I see no argument against making NDA=
TA a MUST in data channels.</div>
</blockquote>
<div>Please let us discuss NDATA issues on the tsvwg mailing list...</div>
<div>Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its ben=
efits outweigh its costs. The current draft does not pass that bar for me. =
If my app cares about HOL blocking, NDATA is not very useful, so my app mus=
t internally fragment, hence NDATA
 actually becomes harmful (8 bytes closer to exceeding MTU, forcing SCTP fr=
agments and hence risking HOL blocking).</div>
<div>Best regards</div>
<div>Michael</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>Mo</div>
<div></div>
<div>PS. Some believe data channels will be more widely used than WebRTC me=
dia (webtorrent, etc.), so it does make sense to consider the desirable pro=
perties of a general peer-to-peer transport, and drop the WebRTC prefix whe=
n talking about data channels.</div>
</blockquote>
<div><br>
</div>
</body>
</html>

--_000_D075BA5D3D254mzanatyciscocom_--


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From: Justin Uberti <juberti@google.com>
Date: Tue, 28 Oct 2014 21:23:34 -0700
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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So it's not clear to me that a=ssrc:X fmtp:Y is actually legal, without any
fmtp parameters, or that it clearly indicates that only PT Y is to be sent
on that SSRC. As such, I consider this solution to be wishful thinking.

I think it would be easier if we defined FEC/FID semantics such that the
initial SSRC was to be used for the primary encoding, and any subsequent
SSRCs were used for the secondary/tertiary streams. (e.g. RTX or FEC)

e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:

     m=video 30000 RTP/AVP 100 101 108 109 110 111
     c=IN IP4 233.252.0.1
     a=rtpmap:100 VP8/90000
     a=rtpmap:101 VP9/90000
     a=rtpmap:108 rtx/90000
     a=rtpmap:109 rtx/90000
     a=fmtp:108 apt=100
     a=fmtp:109 apt=101
     a=rtpmap:110 interleaved-parityfec/90000
     a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000
     a=rtpmap:111 non-interleaved-parityfec/90000
     a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000
     a=ssrc:1234
     a=ssrc:2345
     a=ssrc:3456
     a=ssrc:4567
     a=ssrc-group:FID 1234 2345
     a=ssrc-group:FEC-FR 1234 3456 4567

SSRC 1234 is the primary encoding (VP8 or VP9)
SSRC 2345 is RTX (either PT 108 or 109, as needed)
SSRC 3456 is FEC (either PT 110 or 111, depending on impl)
SSRC 4567 is FEC (either PT 110 or 111, depending on impl)


On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> >
> > Makaraju, Maridi Raju (Raju) wrote:
> > > OPUS does have a built-in FEC so no need for red+ulpfec.
> >
> > As discussed previously on this list [1] the built-in FEC only covers
> > the SILK layer. Existing codec-agnostic mechanisms were perfectly
> > adequate for protecting CELT-mode packets, so we didn't invent something
> > new.
> >
> > [1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html
> [Raju] Thanks and appreciate the reference.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">So it&#39;s not clear to me that a=3Dssrc:X fmtp:Y is actu=
ally legal, without any fmtp parameters, or that it clearly indicates that =
only PT Y is to be sent on that SSRC. As such, I consider this solution to =
be wishful thinking.<div><br></div><div>I think it would be easier if we de=
fined FEC/FID semantics such that the initial SSRC was to be used for the p=
rimary encoding, and any subsequent SSRCs were used for the secondary/terti=
ary streams. (e.g. RTX or FEC)</div><div><br></div><div>e.g. for a stream w=
ith 2 video codecs, RTX and 1d/2d FEC:</div><div>=C2=A0=C2=A0<br></div><div=
><div>=C2=A0 =C2=A0 =C2=A0m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111<b=
r></div><div>=C2=A0 =C2=A0 =C2=A0c=3DIN IP4 233.252.0.1</div><div>=C2=A0 =
=C2=A0 =C2=A0a=3Drtpmap:100 VP8/90000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtp=
map:101 VP9/90000<br></div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:108 rtx/9000=
0</div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:109 rtx/90000<br></div><div>=C2=
=A0 =C2=A0 =C2=A0a=3Dfmtp:108 apt=3D100</div><div>=C2=A0 =C2=A0 =C2=A0a=3Df=
mtp:109 apt=3D101<br></div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:110 interlea=
ved-parityfec/90000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dfmtp:110 L:5; D:10; T=
oP:2; repair-window:200000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:111 non=
-interleaved-parityfec/90000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dfmtp:111 L:5=
; D:10; ToP:2; repair-window:200000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:=
1234</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:2345</div><div>=C2=A0 =C2=A0 =
=C2=A0a=3Dssrc:3456</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:4567</div><div>=
=C2=A0 =C2=A0 =C2=A0a=3Dssrc-group:FID 1234 2345</div><div>=C2=A0 =C2=A0 =
=C2=A0a=3Dssrc-group:FEC-FR 1234 3456 4567</div></div><div><br></div><div>S=
SRC 1234 is the primary encoding (VP8 or VP9)</div><div>SSRC 2345 is RTX (e=
ither PT 108 or 109, as needed)</div><div>SSRC 3456 is FEC (either PT 110 o=
r 111, depending on impl)</div><div>SSRC 4567 is FEC (either PT 110 or 111,=
 depending on impl)</div><div><br></div></div><div class=3D"gmail_extra"><b=
r><div class=3D"gmail_quote">On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Ma=
ridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alca=
tel-lucent.com" target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D"">&gt;<br>
&gt; Makaraju, Maridi Raju (Raju) wrote:<br>
&gt; &gt; OPUS does have a built-in FEC so no need for red+ulpfec.<br>
&gt;<br>
&gt; As discussed previously on this list [1] the built-in FEC only covers<=
br>
&gt; the SILK layer. Existing codec-agnostic mechanisms were perfectly<br>
&gt; adequate for protecting CELT-mode packets, so we didn&#39;t invent som=
ething<br>
&gt; new.<br>
&gt;<br>
&gt; [1] <a href=3D"https://www.ietf.org/mail-archive/web/rtcweb/current/ms=
g12353.html" target=3D"_blank">https://www.ietf.org/mail-archive/web/rtcweb=
/current/msg12353.html</a><br>
</span>[Raju] Thanks and appreciate the reference.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--089e013a03c6ba0660050688216d--


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Date: Tue, 28 Oct 2014 22:22:28 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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--001a11c3ee74329387050688f315
Content-Type: text/plain; charset=UTF-8

Sounds like a reasonable idea to me, but don't you have to specify a strict
pecking order? RTX before FEC doesn't sound right to me, but that's pure
prejudice/bias (FEC is always present, and optional stuff his last)

So how would you decide that order? A hat? PT value? Order of PT on the m
line? Order of the fmtp attributes? Something else? Or can we simply fix
the order, since there are only a small number of variations to consider?
On Oct 28, 2014 9:24 PM, "Justin Uberti" <juberti@google.com> wrote:

> So it's not clear to me that a=ssrc:X fmtp:Y is actually legal, without
> any fmtp parameters, or that it clearly indicates that only PT Y is to be
> sent on that SSRC. As such, I consider this solution to be wishful thinking.
>
> I think it would be easier if we defined FEC/FID semantics such that the
> initial SSRC was to be used for the primary encoding, and any subsequent
> SSRCs were used for the secondary/tertiary streams. (e.g. RTX or FEC)
>
> e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:
>
>      m=video 30000 RTP/AVP 100 101 108 109 110 111
>      c=IN IP4 233.252.0.1
>      a=rtpmap:100 VP8/90000
>      a=rtpmap:101 VP9/90000
>      a=rtpmap:108 rtx/90000
>      a=rtpmap:109 rtx/90000
>      a=fmtp:108 apt=100
>      a=fmtp:109 apt=101
>      a=rtpmap:110 interleaved-parityfec/90000
>      a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000
>      a=rtpmap:111 non-interleaved-parityfec/90000
>      a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000
>      a=ssrc:1234
>      a=ssrc:2345
>      a=ssrc:3456
>      a=ssrc:4567
>      a=ssrc-group:FID 1234 2345
>      a=ssrc-group:FEC-FR 1234 3456 4567
>
> SSRC 1234 is the primary encoding (VP8 or VP9)
> SSRC 2345 is RTX (either PT 108 or 109, as needed)
> SSRC 3456 is FEC (either PT 110 or 111, depending on impl)
> SSRC 4567 is FEC (either PT 110 or 111, depending on impl)
>
>
> On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) <
> Raju.Makaraju@alcatel-lucent.com> wrote:
>
>> >
>> > Makaraju, Maridi Raju (Raju) wrote:
>> > > OPUS does have a built-in FEC so no need for red+ulpfec.
>> >
>> > As discussed previously on this list [1] the built-in FEC only covers
>> > the SILK layer. Existing codec-agnostic mechanisms were perfectly
>> > adequate for protecting CELT-mode packets, so we didn't invent something
>> > new.
>> >
>> > [1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html
>> [Raju] Thanks and appreciate the reference.
>>
>> _______________________________________________
>> 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
>
>

--001a11c3ee74329387050688f315
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">Sounds like a reasonable idea to me, but don&#39;t you have =
to specify a strict pecking order? RTX before FEC doesn&#39;t sound right t=
o me, but that&#39;s pure prejudice/bias (FEC is always present, and option=
al stuff his last)</p>
<p dir=3D"ltr">So how would you decide that order? A hat? PT value? Order o=
f PT on the m line? Order of the fmtp attributes? Something else? Or can we=
 simply fix the order, since there are only a small number of variations to=
 consider?</p>
<div class=3D"gmail_quote">On Oct 28, 2014 9:24 PM, &quot;Justin Uberti&quo=
t; &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wro=
te:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"=
>So it&#39;s not clear to me that a=3Dssrc:X fmtp:Y is actually legal, with=
out any fmtp parameters, or that it clearly indicates that only PT Y is to =
be sent on that SSRC. As such, I consider this solution to be wishful think=
ing.<div><br></div><div>I think it would be easier if we defined FEC/FID se=
mantics such that the initial SSRC was to be used for the primary encoding,=
 and any subsequent SSRCs were used for the secondary/tertiary streams. (e.=
g. RTX or FEC)</div><div><br></div><div>e.g. for a stream with 2 video code=
cs, RTX and 1d/2d FEC:</div><div>=C2=A0=C2=A0<br></div><div><div>=C2=A0 =C2=
=A0 =C2=A0m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111<br></div><div>=C2=
=A0 =C2=A0 =C2=A0c=3DIN IP4 233.252.0.1</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dr=
tpmap:100 VP8/90000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:101 VP9/90000<=
br></div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:108 rtx/90000</div><div>=C2=A0=
 =C2=A0 =C2=A0a=3Drtpmap:109 rtx/90000<br></div><div>=C2=A0 =C2=A0 =C2=A0a=
=3Dfmtp:108 apt=3D100</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dfmtp:109 apt=3D101<=
br></div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:110 interleaved-parityfec/9000=
0</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dfmtp:110 L:5; D:10; ToP:2; repair-windo=
w:200000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Drtpmap:111 non-interleaved-parit=
yfec/90000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dfmtp:111 L:5; D:10; ToP:2; rep=
air-window:200000</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:1234</div><div>=C2=
=A0 =C2=A0 =C2=A0a=3Dssrc:2345</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:3456<=
/div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc:4567</div><div>=C2=A0 =C2=A0 =C2=A0a=
=3Dssrc-group:FID 1234 2345</div><div>=C2=A0 =C2=A0 =C2=A0a=3Dssrc-group:FE=
C-FR 1234 3456 4567</div></div><div><br></div><div>SSRC 1234 is the primary=
 encoding (VP8 or VP9)</div><div>SSRC 2345 is RTX (either PT 108 or 109, as=
 needed)</div><div>SSRC 3456 is FEC (either PT 110 or 111, depending on imp=
l)</div><div>SSRC 4567 is FEC (either PT 110 or 111, depending on impl)</di=
v><div><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_q=
uote">On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) <span =
dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=
=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><span>&gt;<br>
&gt; Makaraju, Maridi Raju (Raju) wrote:<br>
&gt; &gt; OPUS does have a built-in FEC so no need for red+ulpfec.<br>
&gt;<br>
&gt; As discussed previously on this list [1] the built-in FEC only covers<=
br>
&gt; the SILK layer. Existing codec-agnostic mechanisms were perfectly<br>
&gt; adequate for protecting CELT-mode packets, so we didn&#39;t invent som=
ething<br>
&gt; new.<br>
&gt;<br>
&gt; [1] <a href=3D"https://www.ietf.org/mail-archive/web/rtcweb/current/ms=
g12353.html" target=3D"_blank">https://www.ietf.org/mail-archive/web/rtcweb=
/current/msg12353.html</a><br>
</span>[Raju] Thanks and appreciate the reference.<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>
</div></div></blockquote></div><br></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--001a11c3ee74329387050688f315--


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] SDP and ssrc-group,
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Date: Wed, 29 Oct 2014 05:51:05 +0000
Message-ID: <D075F3CB.3D2D6%mzanaty@cisco.com>
References: <CALiegfmH8rRyEDbJjQ=kzMv0nGC=S9gNsE7roE=kqJyVcfgy8g@mail.gmail.com> <CAJrXDUHekuQCLeCYzsnm8AuTUgiVppQHUqR7MKdQ9Q=eFFAy0w@mail.gmail.com> <CALiegfm_B5KfD5SBPzsH4YYuzD2OXdu47TtatPVmd6ihrMCh1A@mail.gmail.com> <CAJrXDUF-AXcDuQmYhH91vbgPAxLkYkB==GY9opoRk7zrEP8A7A@mail.gmail.com> <CALiegfmxnzZ6_3rKX0paNhHas6Emvu1Mekgb9caj9NLVSf_u+g@mail.gmail.com> <5F93BF20-03B2-4D2D-BEFC-ED91250993BD@gmail.com> <54480864.4050106@gmail.com> <5448583B.5090109@mozilla.com> <E1FE4C082A89A246A11D7F32A95A17828E5EE05D@US70UWXCHMBA02.zam.alcatel-lucent.com> <544936E5.2040909@mozilla.com> <E1FE4C082A89A246A11D7F32A95A17828E5EE2D3@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAOJ7v-0Z+eGPNM0JVTWq1Q90eU4pc49+O4rEOkzwTJmBiyoz0Q@mail.gmail.com> <CABkgnnUaVGDVns6ztRNRhn80HBOBWG-s8poibCRefnH8onYDdQ@mail.gmail.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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I think Justin meant the =93initial SSRC=94 in the a=3Dssrc-group lines car=
ries significance (as the source), not the order of a=3Dssrc lines.

What if new FEC schemes defined for bundle / unified plan (such as those be=
low) followed the example of RTX by including apt=3D100 in their fmtp param=
eters? So PT can be used to associate source and FEC flows without declarin=
g any SSRCs or groups.

We can also improve on RTX by allowing apt=3D100,101 for FEC.

When we consider more complex cases like RTX/FEC with simulcast and scalabl=
e streams in unified plan, using PT association may be better than declarin=
g many SSRCs (which is also problematic in cases of collisions, dynamic sou=
rces, switching mixers, etc.).

Mo


On 10/29/14, 1:22 AM, Martin Thomson <martin.thomson@gmail.com<mailto:marti=
n.thomson@gmail.com>> wrote:
Sounds like a reasonable idea to me, but don't you have to specify a strict=
 pecking order? RTX before FEC doesn't sound right to me, but that's pure p=
rejudice/bias (FEC is always present, and optional stuff his last)

So how would you decide that order? A hat? PT value? Order of PT on the m l=
ine? Order of the fmtp attributes? Something else? Or can we simply fix the=
 order, since there are only a small number of variations to consider?

On Oct 28, 2014 9:24 PM, "Justin Uberti" <juberti@google.com<mailto:juberti=
@google.com>> wrote:
So it's not clear to me that a=3Dssrc:X fmtp:Y is actually legal, without a=
ny fmtp parameters, or that it clearly indicates that only PT Y is to be se=
nt on that SSRC. As such, I consider this solution to be wishful thinking.

I think it would be easier if we defined FEC/FID semantics such that the in=
itial SSRC was to be used for the primary encoding, and any subsequent SSRC=
s were used for the secondary/tertiary streams. (e.g. RTX or FEC)

e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:

     m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111
     c=3DIN IP4 233.252.0.1
     a=3Drtpmap:100 VP8/90000
     a=3Drtpmap:101 VP9/90000
     a=3Drtpmap:108 rtx/90000
     a=3Drtpmap:109 rtx/90000
     a=3Dfmtp:108 apt=3D100
     a=3Dfmtp:109 apt=3D101
     a=3Drtpmap:110 interleaved-parityfec/90000
     a=3Dfmtp:110 L:5; D:10; ToP:2; repair-window:200000
     a=3Drtpmap:111 non-interleaved-parityfec/90000
     a=3Dfmtp:111 L:5; D:10; ToP:2; repair-window:200000
     a=3Dssrc:1234
     a=3Dssrc:2345
     a=3Dssrc:3456
     a=3Dssrc:4567
     a=3Dssrc-group:FID 1234 2345
     a=3Dssrc-group:FEC-FR 1234 3456 4567

SSRC 1234 is the primary encoding (VP8 or VP9)
SSRC 2345 is RTX (either PT 108 or 109, as needed)
SSRC 3456 is FEC (either PT 110 or 111, depending on impl)
SSRC 4567 is FEC (either PT 110 or 111, depending on impl)


On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) <Raju.Makara=
ju@alcatel-lucent.com<mailto:Raju.Makaraju@alcatel-lucent.com>> wrote:
>
> Makaraju, Maridi Raju (Raju) wrote:
> > OPUS does have a built-in FEC so no need for red+ulpfec.
>
> As discussed previously on this list [1] the built-in FEC only covers
> the SILK layer. Existing codec-agnostic mechanisms were perfectly
> adequate for protecting CELT-mode packets, so we didn't invent something
> new.
>
> [1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html
[Raju] Thanks and appreciate the reference.

_______________________________________________
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


--_000_D075F3CB3D2D6mzanatyciscocom_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <32A74E8E34251149B593802A844AB565@emea.cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 12px; font-fami=
ly: Arial, sans-serif;">
<div>I think Justin meant the =93initial SSRC=94 in the a=3Dssrc-group line=
s carries significance (as the source), not the order of a=3Dssrc lines.</d=
iv>
<div><br>
</div>
<div>What if new FEC schemes defined for bundle / unified plan (such as tho=
se below) followed the example of RTX by including apt=3D100 in their fmtp =
parameters? So PT can be used to associate source and FEC flows without dec=
laring any SSRCs or groups.</div>
<div><br>
</div>
<div>We can also improve on RTX by allowing apt=3D100,101 for FEC.</div>
<div><br>
</div>
<div>When we consider more complex cases like RTX/FEC with simulcast and sc=
alable streams in unified plan, using PT association may be better than dec=
laring many SSRCs (which is also problematic in cases of collisions, dynami=
c sources, switching mixers, etc.).</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>On 10/29/14, 1:22 AM, Martin Thomson &lt;<a href=3D"mailto:martin.thom=
son@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:</div>
</div>
<div>Sounds like a reasonable idea to me, but don't you have to specify a s=
trict pecking order? RTX before FEC doesn't sound right to me, but that's p=
ure prejudice/bias (FEC is always present, and optional stuff his last)</di=
v>
<div>
<p dir=3D"ltr">So how would you decide that order? A hat? PT value? Order o=
f PT on the m line? Order of the fmtp attributes? Something else? Or can we=
 simply fix the order, since there are only a small number of variations to=
 consider?</p>
<div class=3D"gmail_quote">On Oct 28, 2014 9:24 PM, &quot;Justin Uberti&quo=
t; &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wro=
te:<br type=3D"attribution">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">So it's not clear to me that a=3Dssrc:X fmtp:Y is actually=
 legal, without any fmtp parameters, or that it clearly indicates that only=
 PT Y is to be sent on that SSRC. As such, I consider this solution to be w=
ishful thinking.
<div><br>
</div>
<div>I think it would be easier if we defined FEC/FID semantics such that t=
he initial SSRC was to be used for the primary encoding, and any subsequent=
 SSRCs were used for the secondary/tertiary streams. (e.g. RTX or FEC)</div=
>
<div><br>
</div>
<div>e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:</div>
<div>&nbsp;&nbsp;<br>
</div>
<div>
<div>&nbsp; &nbsp; &nbsp;m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111<br=
>
</div>
<div>&nbsp; &nbsp; &nbsp;c=3DIN IP4 233.252.0.1</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:100 VP8/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:101 VP9/90000<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:108 rtx/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:109 rtx/90000<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:108 apt=3D100</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:109 apt=3D101<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:110 interleaved-parityfec/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:110 L:5; D:10; ToP:2; repair-window:20000=
0</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:111 non-interleaved-parityfec/90000</di=
v>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:111 L:5; D:10; ToP:2; repair-window:20000=
0</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:1234</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:2345</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:3456</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:4567</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc-group:FID 1234 2345</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc-group:FEC-FR 1234 3456 4567</div>
</div>
<div><br>
</div>
<div>SSRC 1234 is the primary encoding (VP8 or VP9)</div>
<div>SSRC 2345 is RTX (either PT 108 or 109, as needed)</div>
<div>SSRC 3456 is FEC (either PT 110 or 111, depending on impl)</div>
<div>SSRC 4567 is FEC (either PT 110 or 111, depending on impl)</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Mari=
di Raju (Raju)
<span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" t=
arget=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<span>&gt;<br>
&gt; Makaraju, Maridi Raju (Raju) wrote:<br>
&gt; &gt; OPUS does have a built-in FEC so no need for red&#43;ulpfec.<br>
&gt;<br>
&gt; As discussed previously on this list [1] the built-in FEC only covers<=
br>
&gt; the SILK layer. Existing codec-agnostic mechanisms were perfectly<br>
&gt; adequate for protecting CELT-mode packets, so we didn't invent somethi=
ng<br>
&gt; new.<br>
&gt;<br>
&gt; [1] <a href=3D"https://www.ietf.org/mail-archive/web/rtcweb/current/ms=
g12353.html" target=3D"_blank">
https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html</a><br>
</span>[Raju] Thanks and appreciate the reference.<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>
</div>
</div>
</blockquote>
</div>
<br>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
</div>
</span>
</body>
</html>

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Subject: Re: [rtcweb] SDP and ssrc-group,
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Am not against allowing 'apt' with FEC, but forcing a PT to be consumed for e=
ach FEC stream could get out of hand, particularly for combined simulcast/SV=
C cases.=20



> On Oct 28, 2014, at 10:51 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrot=
e:
>=20
> I think Justin meant the =E2=80=9Cinitial SSRC=E2=80=9D in the a=3Dssrc-gr=
oup lines carries significance (as the source), not the order of a=3Dssrc li=
nes.
>=20
> What if new FEC schemes defined for bundle / unified plan (such as those b=
elow) followed the example of RTX by including apt=3D100 in their fmtp param=
eters? So PT can be used to associate source and FEC flows without declaring=
 any SSRCs or groups.
>=20
> We can also improve on RTX by allowing apt=3D100,101 for FEC.
>=20
> When we consider more complex cases like RTX/FEC with simulcast and scalab=
le streams in unified plan, using PT association may be better than declarin=
g many SSRCs (which is also problematic in cases of collisions, dynamic sour=
ces, switching mixers, etc.).
>=20
> Mo
>=20
>=20
> On 10/29/14, 1:22 AM, Martin Thomson <martin.thomson@gmail.com> wrote:
> Sounds like a reasonable idea to me, but don't you have to specify a stric=
t pecking order? RTX before FEC doesn't sound right to me, but that's pure p=
rejudice/bias (FEC is always present, and optional stuff his last)
> So how would you decide that order? A hat? PT value? Order of PT on the m l=
ine? Order of the fmtp attributes? Something else? Or can we simply fix the o=
rder, since there are only a small number of variations to consider?
>=20
>> On Oct 28, 2014 9:24 PM, "Justin Uberti" <juberti@google.com> wrote:
>> So it's not clear to me that a=3Dssrc:X fmtp:Y is actually legal, without=
 any fmtp parameters, or that it clearly indicates that only PT Y is to be s=
ent on that SSRC. As such, I consider this solution to be wishful thinking.
>>=20
>> I think it would be easier if we defined FEC/FID semantics such that the i=
nitial SSRC was to be used for the primary encoding, and any subsequent SSRC=
s were used for the secondary/tertiary streams. (e.g. RTX or FEC)
>>=20
>> e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:
>>  =20
>>      m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111
>>      c=3DIN IP4 233.252.0.1
>>      a=3Drtpmap:100 VP8/90000
>>      a=3Drtpmap:101 VP9/90000
>>      a=3Drtpmap:108 rtx/90000
>>      a=3Drtpmap:109 rtx/90000
>>      a=3Dfmtp:108 apt=3D100
>>      a=3Dfmtp:109 apt=3D101
>>      a=3Drtpmap:110 interleaved-parityfec/90000
>>      a=3Dfmtp:110 L:5; D:10; ToP:2; repair-window:200000
>>      a=3Drtpmap:111 non-interleaved-parityfec/90000
>>      a=3Dfmtp:111 L:5; D:10; ToP:2; repair-window:200000
>>      a=3Dssrc:1234
>>      a=3Dssrc:2345
>>      a=3Dssrc:3456
>>      a=3Dssrc:4567
>>      a=3Dssrc-group:FID 1234 2345
>>      a=3Dssrc-group:FEC-FR 1234 3456 4567
>>=20
>> SSRC 1234 is the primary encoding (VP8 or VP9)
>> SSRC 2345 is RTX (either PT 108 or 109, as needed)
>> SSRC 3456 is FEC (either PT 110 or 111, depending on impl)
>> SSRC 4567 is FEC (either PT 110 or 111, depending on impl)
>>=20
>>=20
>>> On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) <Raju.Mak=
araju@alcatel-lucent.com> wrote:
>>> >
>>> > Makaraju, Maridi Raju (Raju) wrote:
>>> > > OPUS does have a built-in FEC so no need for red+ulpfec.
>>> >
>>> > As discussed previously on this list [1] the built-in FEC only covers
>>> > the SILK layer. Existing codec-agnostic mechanisms were perfectly
>>> > adequate for protecting CELT-mode packets, so we didn't invent somethi=
ng
>>> > new.
>>> >
>>> > [1] https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html=

>>> [Raju] Thanks and appreciate the reference.
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=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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Content-Type: text/html;
	charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>Am not against allowing 'apt' with FEC=
, but forcing a PT to be consumed for each FEC stream could get out of hand,=
 particularly for combined simulcast/SVC cases.&nbsp;<br><br><br></div><div>=
<br>On Oct 28, 2014, at 10:51 PM, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:=
mzanaty@cisco.com">mzanaty@cisco.com</a>&gt; wrote:<br><br></div><blockquote=
 type=3D"cite"><div>

<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-12=
52">


<div>I think Justin meant the =E2=80=9Cinitial SSRC=E2=80=9D in the a=3Dssrc=
-group lines carries significance (as the source), not the order of a=3Dssrc=
 lines.</div>
<div><br>
</div>
<div>What if new FEC schemes defined for bundle / unified plan (such as thos=
e below) followed the example of RTX by including apt=3D100 in their fmtp pa=
rameters? So PT can be used to associate source and FEC flows without declar=
ing any SSRCs or groups.</div>
<div><br>
</div>
<div>We can also improve on RTX by allowing apt=3D100,101 for FEC.</div>
<div><br>
</div>
<div>When we consider more complex cases like RTX/FEC with simulcast and sca=
lable streams in unified plan, using PT association may be better than decla=
ring many SSRCs (which is also problematic in cases of collisions, dynamic s=
ources, switching mixers, etc.).</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>On 10/29/14, 1:22 AM, Martin Thomson &lt;<a href=3D"mailto:martin.thoms=
on@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:</div>
</div>
<div>Sounds like a reasonable idea to me, but don't you have to specify a st=
rict pecking order? RTX before FEC doesn't sound right to me, but that's pur=
e prejudice/bias (FEC is always present, and optional stuff his last)</div>
<div>
<p dir=3D"ltr">So how would you decide that order? A hat? PT value? Order of=
 PT on the m line? Order of the fmtp attributes? Something else? Or can we s=
imply fix the order, since there are only a small number of variations to co=
nsider?</p>
<div class=3D"gmail_quote">On Oct 28, 2014 9:24 PM, "Justin Uberti" &lt;<a h=
ref=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:<br type=3D=
"attribution">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
<div dir=3D"ltr">So it's not clear to me that a=3Dssrc:X fmtp:Y is actually l=
egal, without any fmtp parameters, or that it clearly indicates that only PT=
 Y is to be sent on that SSRC. As such, I consider this solution to be wishf=
ul thinking.
<div><br>
</div>
<div>I think it would be easier if we defined FEC/FID semantics such that th=
e initial SSRC was to be used for the primary encoding, and any subsequent S=
SRCs were used for the secondary/tertiary streams. (e.g. RTX or FEC)</div>
<div><br>
</div>
<div>e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:</div>
<div>&nbsp;&nbsp;<br>
</div>
<div>
<div>&nbsp; &nbsp; &nbsp;m=3Dvideo 30000 RTP/AVP 100 101 108 109 110 111<br>=

</div>
<div>&nbsp; &nbsp; &nbsp;c=3DIN IP4 233.252.0.1</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:100 VP8/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:101 VP9/90000<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:108 rtx/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:109 rtx/90000<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:108 apt=3D100</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:109 apt=3D101<br>
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:110 interleaved-parityfec/90000</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:110 L:5; D:10; ToP:2; repair-window:200000=
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Drtpmap:111 non-interleaved-parityfec/90000</div=
>
<div>&nbsp; &nbsp; &nbsp;a=3Dfmtp:111 L:5; D:10; ToP:2; repair-window:200000=
</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:1234</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:2345</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:3456</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc:4567</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc-group:FID 1234 2345</div>
<div>&nbsp; &nbsp; &nbsp;a=3Dssrc-group:FEC-FR 1234 3456 4567</div>
</div>
<div><br>
</div>
<div>SSRC 1234 is the primary encoding (VP8 or VP9)</div>
<div>SSRC 2345 is RTX (either PT 108 or 109, as needed)</div>
<div>SSRC 3456 is FEC (either PT 110 or 111, depending on impl)</div>
<div>SSRC 4567 is FEC (either PT 110 or 111, depending on impl)</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Marid=
i Raju (Raju)
<span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" ta=
rget=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
<span>&gt;<br>
&gt; Makaraju, Maridi Raju (Raju) wrote:<br>
&gt; &gt; OPUS does have a built-in FEC so no need for red+ulpfec.<br>
&gt;<br>
&gt; As discussed previously on this list [1] the built-in FEC only covers<b=
r>
&gt; the SILK layer. Existing codec-agnostic mechanisms were perfectly<br>
&gt; adequate for protecting CELT-mode packets, so we didn't invent somethin=
g<br>
&gt; new.<br>
&gt;<br>
&gt; [1] <a href=3D"https://www.ietf.org/mail-archive/web/rtcweb/current/msg=
12353.html" target=3D"_blank">
https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html</a><br>
</span>[Raju] Thanks and appreciate the reference.<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">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
</div>
</span>


</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] SDP and ssrc-group,
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--_000_7594FB04B1934943A5C02806D1A2204B1D4D2633ESESSMB209erics_--


From nobody Wed Oct 29 00:48:13 2014
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: =?utf-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>, <rtcweb@ietf.org>
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Date: Wed, 29 Oct 2014 09:48:00 +0200
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Hi,
We discussed this mapping issue in a previous version of the application =
token draft =
http://tools.ietf.org/html/draft-even-mmusic-application-token-02 =
example in section 1.
We thought of having a mapping from appId to "pt" (section 3.3.1)
Roni

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of I?aki Baz =
Castillo
> Sent: 21 October, 2014 7:26 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] SDP and ssrc-group,
>=20
> Hi,
>=20
> May I know which SSRC (345259865 or 2693756249) will be used for the =
real
> media stream (plus RED and FEC) and which SSRC will be used for RTX?
>=20
>=20
>=20
> --------------------------
> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96
> a=3Dmid:video
> a=3Drtpmap:100 VP8/90000
> a=3Drtpmap:116 red/90000
> a=3Drtpmap:117 ulpfec/90000
> a=3Drtpmap:96 rtx/90000
> a=3Dfmtp:96 apt=3D100
> a=3Dssrc-group:FID 345259865 2693756249
> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> c0134f05-e7c2-4afd-a979-4e224de5eb91
> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> -------------------------------
>=20
>=20
>=20
>=20
> RFC 5576 does not clarify it at all:
>=20
> http://tools.ietf.org/html/rfc5576#section-4.2
>=20
> --------------------------------------------------
> 4.2.  The "ssrc-group" Media Attribute
>=20
>    a=3Dssrc-group:<semantics> <ssrc-id> ...
>=20
>    [..]
>=20
>    The <semantics> parameter is taken from the specification of the
>    "group" attribute [RFC3388].  The initial semantic values defined =
for
>    the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>    and FEC (Forward Error Correction) [RFC4756].  In each case, the
>    relationship among the grouped sources is the same as the
>    relationship among corresponding sources in media streams grouped
>    using the SDP "group" attribute.
> --------------------------------------------------
>=20
>=20
>=20
> The referenced RFC 3388 neither clarifies it:
>=20
> ---------------------------------------------------
> 7.4 FID Semantics
>=20
>    Several "m" lines grouped together using FID semantics form a media
>    flow.  A media agent handling a media flow that comprises several =
"m"
>    lines MUST send a copy of the media to every "m" line part of the
>    flow as long as the codecs and the direction attribute present in a
>    particular "m" line allow it.
>=20
>    It is assumed that the application uses only one codec at a time to
>    encode the media produced.  This codec MAY change dynamically =
during
>    the session, but at any particular moment only one codec is in use.
>=20
>    The application encodes the media using the current codec and =
checks
>    one by one all the "m" lines that are part of the flow.  If a
>    particular "m" line contains the codec being used and the direction
>    attribute is "sendonly" or "sendrecv", a copy of the encoded media =
is
>    sent to the address/port specified in that particular media stream.
>    If either the "m" line does not contain the codec being used or the
>    direction attribute is neither "sendonly" nor "sendrecv", nothing =
is
>    sent over this media stream.
> ----------------------------------------------------
>=20
>=20
>=20
>=20
> So, how is the usage of ssrc-group? Where is it really defined?
>=20
> Can I put more than 2 ssrc together in the same ssrc-group line?
>=20
> How should the receiver interpret it?
>=20
> Does a ssrc-group always mean RTX usage? Where is that specified in =
the
> above SDP?
>=20
> Does not the above SDP look a complete mixture of hacks and =
workarounds?
>=20
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Oct 29 02:11:41 2014
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] draft agenda for rtcweb at ietf91?
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Dear chairs,

where could I find the draft agenda for the upcoming meeting?

Thanks in advance,
-Victor


From nobody Wed Oct 29 03:05:49 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com>, <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 28 Oct 2014, at 23:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

> On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" =
<Michael.Tuexen@lurchi.franken.de> wrote:
>=20
> On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:
>=20
>>> What resource is being conserved by not just requiring it?
>>=20
>> 8 bytes per packet (for MID+FSN).
>>=20
>> Michael, the current NDATA draft requires it to be used for all =
messages if negotiated. Can=92t this be relaxed / optimized to use it =
for all *fragmented* messages but use DATA for non-fragmented messages =
to avoid the 8 byte overhead for small messages (like game =
telemetry/control, an often cited use of data channels).
> You can bring this up at the tsvwg mailing list.
>=20
> Mo: Will do.
>=20
> However, we normally do not optimize for byte saving... Mixing
> both makes the processing harder... Maybe we can use a bit in the =
flags to indicate if the additional fields are
> there...
>=20
> Mo: Type should already be enough without flags.
>=20
>>=20
>> The current NDATA draft also says you can=92t interleave fragmented =
messages in a stream, so head of line blocking remains for all =
fragmented messages, while only small non-fragmented messages can avoid =
it. This dilutes the utility of NDATA, perhaps enough to make apps that =
really care about head of line blocking to implement their own app layer =
fragmentation with messages well below common MTUs, hence defeating =
NDATA. Can=92t this also be relaxed since MID is part of the NDATA extra =
header?
> This would only make sense for messages marked as unordered... For =
ordered it doesn't make sense.
>=20
> Mo: The restriction applies to unordered as well as ordered. Even for =
ordered, it does make sense to avoid head of line blocking at the =
sender.=20
I don't understand how you can do it for ordered. The sender sends a =
large message on stream 0 and after that
a small on stream zero. For both ordered delivery is request. So the =
receiver can't deliver the second before the
first.
>>=20
>> If those two issues are resolved, I see no argument against making =
NDATA a MUST in data channels.
> Please let us discuss NDATA issues on the tsvwg mailing list...
>=20
> Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its =
benefits outweigh its costs. The current draft does not pass that bar =
for me. If my app cares about HOL blocking, NDATA is not very useful, so =
my app must internally fragment, hence NDATA actually becomes harmful (8 =
bytes closer to exceeding MTU, forcing SCTP fragments and hence risking =
HOL blocking).
I don't understand the point when I look at the API for data channels... =
Only for sending messages
on a data channel not requiring ordered delivery. However, the intention =
of NDATA is to mitigate
the impact of one data channel on others belonging to the same peer =
connection. This goal is met.


Best regards
Michael
>=20
> Best regards
> Michael
>>=20
>> Mo
>>=20
>> PS. Some believe data channels will be more widely used than WebRTC =
media (webtorrent, etc.), so it does make sense to consider the =
desirable properties of a general peer-to-peer transport, and drop the =
WebRTC prefix when talking about data channels.
>=20
>=20


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Subject: Re: [rtcweb] SDP and ssrc-group,
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Indeed, that was my original concern also.

Using the order of ssrc-ids on the ssrc-group may be the simplest way to 
solve it, but not sure if it would allow more complex FEC scenarios, 
like one FEC stream protecting two media flows (maybe not relevant at all).

I have been surfing at the ietf rfcs, and found and interesting one 
rfc6364 which defines  fec-source-flow and fec-repair-flow media 
attributes that can be used with a=group:FEC-FR, hat would allow exactly 
what we need:

         v=0
         o=ali 1122334455 1122334466 IN IP4 fec.example.com
         s=FEC Framework Examples
         t=0 0
         a=group:FEC-FR S1 R1
         m=video 30000 RTP/AVP 100
         c=IN IP4 233.252.0.1/127
         a=rtpmap:100 MP2T/90000
         a=fec-source-flow: id=0
         a=mid:S1
         m=application 30000 UDP/FEC
         c=IN IP4 233.252.0.2/127
         a=fec-repair-flow: encoding-id=0; ss-fssi=n:7,k:5
         a=repair-window:150ms
         a=mid:R1


We would need to port thatould neerc-group semantics, like for example

      m=video 30000 RTP/AVP 100 101 108 109 110 111
      c=IN IP4 233.252.0.1
      a=rtpmap:100 VP8/90000
      a=rtpmap:101 VP9/90000
      a=rtpmap:108 rtx/90000
      a=rtpmap:109 rtx/90000
      a=fmtp:108 apt=100
      a=fmtp:109 apt=101
      a=rtpmap:110 interleaved-parityfec/90000
      a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000
      a=rtpmap:111 non-interleaved-parityfec/90000
      a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000
      a=ssrc:1234 fec-source-flow: id=0
      a=ssrc:2345
      a=ssrc:3456 fec-repair-flow: encoding-id=0;
      a=ssrc:4567 fec-repair-flow: encoding-id=0;
      a=ssrc-group:FID 1234 2345
      a=ssrc-group:FEC-FR 1234 3456 4567

Best regards
Sergio
On 29/10/2014 5:23, Justin Uberti wrote:
> So it's not clear to me that a=ssrc:X fmtp:Y is actually legal, 
> without any fmtp parameters, or that it clearly indicates that only PT 
> Y is to be sent on that SSRC. As such, I consider this solution to be 
> wishful thinking.
>
> I think it would be easier if we defined FEC/FID semantics such that 
> the initial SSRC was to be used for the primary encoding, and any 
> subsequent SSRCs were used for the secondary/tertiary streams. (e.g. 
> RTX or FEC)
>
> e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:
>
>      m=video 30000 RTP/AVP 100 101 108 109 110 111
>      c=IN IP4 233.252.0.1
>      a=rtpmap:100 VP8/90000
>      a=rtpmap:101 VP9/90000
>      a=rtpmap:108 rtx/90000
>      a=rtpmap:109 rtx/90000
>      a=fmtp:108 apt=100
>      a=fmtp:109 apt=101
>      a=rtpmap:110 interleaved-parityfec/90000
>      a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000
>      a=rtpmap:111 non-interleaved-parityfec/90000
>      a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000
>      a=ssrc:1234
>      a=ssrc:2345
>      a=ssrc:3456
>      a=ssrc:4567
>      a=ssrc-group:FID 1234 2345
>      a=ssrc-group:FEC-FR 1234 3456 4567
>
> SSRC 1234 is the primary encoding (VP8 or VP9)
> SSRC 2345 is RTX (either PT 108 or 109, as needed)
> SSRC 3456 is FEC (either PT 110 or 111, depending on impl)
> SSRC 4567 is FEC (either PT 110 or 111, depending on impl)
>
>
> On Thu, Oct 23, 2014 at 12:29 PM, Makaraju, Maridi Raju (Raju) 
> <Raju.Makaraju@alcatel-lucent.com 
> <mailto:Raju.Makaraju@alcatel-lucent.com>> wrote:
>
>     >
>     > Makaraju, Maridi Raju (Raju) wrote:
>     > > OPUS does have a built-in FEC so no need for red+ulpfec.
>     >
>     > As discussed previously on this list [1] the built-in FEC only
>     covers
>     > the SILK layer. Existing codec-agnostic mechanisms were perfectly
>     > adequate for protecting CELT-mode packets, so we didn't invent
>     something
>     > new.
>     >
>     > [1]
>     https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html
>     [Raju] Thanks and appreciate the reference.
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">Indeed, that was my original concern
      also.<br>
      <br>
      Using the order of ssrc-ids on the ssrc-group may be the simplest
      way to solve it, but not sure if it would allow more complex FEC
      scenarios, like one FEC stream protecting two media flows (maybe
      not relevant at all).<br>
      <br>
      I have been surfing at the ietf rfcs, and found and interesting
      one rfc6364 which defines&nbsp; fec-source-flow and fec-repair-flow
      media attributes that can be used with a=group:FEC-FR, hat would
      allow exactly what we need:<br>
      <br>
      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">        v=0
        o=ali 1122334455 1122334466 IN IP4 fec.example.com
        s=FEC Framework Examples
        t=0 0
        a=group:FEC-FR S1 R1
        m=video 30000 RTP/AVP 100
        c=IN IP4 233.252.0.1/127
        a=rtpmap:100 MP2T/90000
        a=fec-source-flow: id=0
        a=mid:S1
        m=application 30000 UDP/FEC
        c=IN IP4 233.252.0.2/127
        a=fec-repair-flow: encoding-id=0; ss-fssi=n:7,k:5
        a=repair-window:150ms
        a=mid:R1</pre>
      <br>
      We would need to port thatould neerc-group semantics, like for
      example<br>
      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">
</pre>
      <div>
        <div>&nbsp; &nbsp; &nbsp;m=video 30000 RTP/AVP 100 101 108 109 110 111<br>
        </div>
        <div>&nbsp; &nbsp; &nbsp;c=IN IP4 233.252.0.1</div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:100 VP8/90000</div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:101 VP9/90000<br>
        </div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:108 rtx/90000</div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:109 rtx/90000<br>
        </div>
        <div>&nbsp; &nbsp; &nbsp;a=fmtp:108 apt=100</div>
        <div>&nbsp; &nbsp; &nbsp;a=fmtp:109 apt=101<br>
        </div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:110 interleaved-parityfec/90000</div>
        <div>&nbsp; &nbsp; &nbsp;a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000</div>
        <div>&nbsp; &nbsp; &nbsp;a=rtpmap:111 non-interleaved-parityfec/90000</div>
        <div>&nbsp; &nbsp; &nbsp;a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000</div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc:1234 fec-source-flow: id=0<br>
        </div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc:2345</div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc:3456 fec-repair-flow: encoding-id=0;</div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc:4567 fec-repair-flow: encoding-id=0;</div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc-group:FID 1234 2345</div>
        <div>&nbsp; &nbsp; &nbsp;a=ssrc-group:FEC-FR 1234 3456 4567</div>
      </div>
      <br>
      Best regards<br>
      Sergio<br>
      On 29/10/2014 5:23, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-0Z+eGPNM0JVTWq1Q90eU4pc49+O4rEOkzwTJmBiyoz0Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">So it's not clear to me that a=ssrc:X fmtp:Y is
        actually legal, without any fmtp parameters, or that it clearly
        indicates that only PT Y is to be sent on that SSRC. As such, I
        consider this solution to be wishful thinking.
        <div><br>
        </div>
        <div>I think it would be easier if we defined FEC/FID semantics
          such that the initial SSRC was to be used for the primary
          encoding, and any subsequent SSRCs were used for the
          secondary/tertiary streams. (e.g. RTX or FEC)</div>
        <div><br>
        </div>
        <div>e.g. for a stream with 2 video codecs, RTX and 1d/2d FEC:</div>
        <div>&nbsp;&nbsp;<br>
        </div>
        <div>
          <div>&nbsp; &nbsp; &nbsp;m=video 30000 RTP/AVP 100 101 108 109 110 111<br>
          </div>
          <div>&nbsp; &nbsp; &nbsp;c=IN IP4 233.252.0.1</div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:100 VP8/90000</div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:101 VP9/90000<br>
          </div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:108 rtx/90000</div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:109 rtx/90000<br>
          </div>
          <div>&nbsp; &nbsp; &nbsp;a=fmtp:108 apt=100</div>
          <div>&nbsp; &nbsp; &nbsp;a=fmtp:109 apt=101<br>
          </div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:110 interleaved-parityfec/90000</div>
          <div>&nbsp; &nbsp; &nbsp;a=fmtp:110 L:5; D:10; ToP:2; repair-window:200000</div>
          <div>&nbsp; &nbsp; &nbsp;a=rtpmap:111 non-interleaved-parityfec/90000</div>
          <div>&nbsp; &nbsp; &nbsp;a=fmtp:111 L:5; D:10; ToP:2; repair-window:200000</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc:1234</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc:2345</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc:3456</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc:4567</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc-group:FID 1234 2345</div>
          <div>&nbsp; &nbsp; &nbsp;a=ssrc-group:FEC-FR 1234 3456 4567</div>
        </div>
        <div><br>
        </div>
        <div>SSRC 1234 is the primary encoding (VP8 or VP9)</div>
        <div>SSRC 2345 is RTX (either PT 108 or 109, as needed)</div>
        <div>SSRC 3456 is FEC (either PT 110 or 111, depending on impl)</div>
        <div>SSRC 4567 is FEC (either PT 110 or 111, depending on impl)</div>
        <div><br>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Thu, Oct 23, 2014 at 12:29 PM,
          Makaraju, Maridi Raju (Raju) <span dir="ltr">&lt;<a
              moz-do-not-send="true"
              href="mailto:Raju.Makaraju@alcatel-lucent.com"
              target="_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex"><span
              class="">&gt;<br>
              &gt; Makaraju, Maridi Raju (Raju) wrote:<br>
              &gt; &gt; OPUS does have a built-in FEC so no need for
              red+ulpfec.<br>
              &gt;<br>
              &gt; As discussed previously on this list [1] the built-in
              FEC only covers<br>
              &gt; the SILK layer. Existing codec-agnostic mechanisms
              were perfectly<br>
              &gt; adequate for protecting CELT-mode packets, so we
              didn't invent something<br>
              &gt; new.<br>
              &gt;<br>
              &gt; [1] <a moz-do-not-send="true"
href="https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html"
                target="_blank">https://www.ietf.org/mail-archive/web/rtcweb/current/msg12353.html</a><br>
            </span>[Raju] Thanks and appreciate the reference.<br>
            <div class="HOEnZb">
              <div class="h5"><br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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--_000_E1FE4C082A89A246A11D7F32A95A17828E5F963BUS70UWXCHMBA02z_--


From nobody Wed Oct 29 05:30:16 2014
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Date: Wed, 29 Oct 2014 13:30:22 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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References: <CALiegfmH8rRyEDbJjQ=kzMv0nGC=S9gNsE7roE=kqJyVcfgy8g@mail.gmail.com> <095701cff34c$ad1089c0$07319d40$@gmail.com>
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Subject: Re: [rtcweb] SDP and ssrc-group,
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 From a formal perspective, I don't feel it is correct to describe the 
semantical meaning of an ssrc by referring to a payload.

Best regards
Sergio
On 29/10/2014 8:48, Roni Even wrote:
> Hi,
> We discussed this mapping issue in a previous version of the application token draft http://tools.ietf.org/html/draft-even-mmusic-application-token-02 example in section 1.
> We thought of having a mapping from appId to "pt" (section 3.3.1)
> Roni
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of I?aki Baz Castillo
>> Sent: 21 October, 2014 7:26 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] SDP and ssrc-group,
>>
>> Hi,
>>
>> May I know which SSRC (345259865 or 2693756249) will be used for the real
>> media stream (plus RED and FEC) and which SSRC will be used for RTX?
>>
>>
>>
>> --------------------------
>> m=video 62164 RTP/SAVPF 100 116 117 96
>> a=mid:video
>> a=rtpmap:100 VP8/90000
>> a=rtpmap:116 red/90000
>> a=rtpmap:117 ulpfec/90000
>> a=rtpmap:96 rtx/90000
>> a=fmtp:96 apt=100
>> a=ssrc-group:FID 345259865 2693756249
>> a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>> a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=ssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> a=ssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=ssrc:2693756249 cname:erS7E/KHLYKTejNs
>> a=ssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>> a=ssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>> a=ssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>> -------------------------------
>>
>>
>>
>>
>> RFC 5576 does not clarify it at all:
>>
>> http://tools.ietf.org/html/rfc5576#section-4.2
>>
>> --------------------------------------------------
>> 4.2.  The "ssrc-group" Media Attribute
>>
>>     a=ssrc-group:<semantics> <ssrc-id> ...
>>
>>     [..]
>>
>>     The <semantics> parameter is taken from the specification of the
>>     "group" attribute [RFC3388].  The initial semantic values defined for
>>     the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>     and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>     relationship among the grouped sources is the same as the
>>     relationship among corresponding sources in media streams grouped
>>     using the SDP "group" attribute.
>> --------------------------------------------------
>>
>>
>>
>> The referenced RFC 3388 neither clarifies it:
>>
>> ---------------------------------------------------
>> 7.4 FID Semantics
>>
>>     Several "m" lines grouped together using FID semantics form a media
>>     flow.  A media agent handling a media flow that comprises several "m"
>>     lines MUST send a copy of the media to every "m" line part of the
>>     flow as long as the codecs and the direction attribute present in a
>>     particular "m" line allow it.
>>
>>     It is assumed that the application uses only one codec at a time to
>>     encode the media produced.  This codec MAY change dynamically during
>>     the session, but at any particular moment only one codec is in use.
>>
>>     The application encodes the media using the current codec and checks
>>     one by one all the "m" lines that are part of the flow.  If a
>>     particular "m" line contains the codec being used and the direction
>>     attribute is "sendonly" or "sendrecv", a copy of the encoded media is
>>     sent to the address/port specified in that particular media stream.
>>     If either the "m" line does not contain the codec being used or the
>>     direction attribute is neither "sendonly" nor "sendrecv", nothing is
>>     sent over this media stream.
>> ----------------------------------------------------
>>
>>
>>
>>
>> So, how is the usage of ssrc-group? Where is it really defined?
>>
>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>
>> How should the receiver interpret it?
>>
>> Does a ssrc-group always mean RTX usage? Where is that specified in the
>> above SDP?
>>
>> Does not the above SDP look a complete mixture of hacks and workarounds?
>>
>>
>>
>>
>> --
>> IÃ±aki Baz Castillo
>> <ibc@aliax.net>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Sergio Garcia Murillo'" <sergio.garcia.murillo@gmail.com>, <rtcweb@ietf.org>
References: <CALiegfmH8rRyEDbJjQ=kzMv0nGC=S9gNsE7roE=kqJyVcfgy8g@mail.gmail.com> <095701cff34c$ad1089c0$07319d40$@gmail.com> <5450DDDE.5020802@gmail.com>
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Date: Wed, 29 Oct 2014 17:14:11 +0200
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Hi,
In the application token proposal we were trying to address two FEC =
issues:

1. You assume that the SDP layer will know the SSRCs of the streams when =
making an offer. This may be true for point to point call but there are =
topologies =
(https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-04)=
 where the SSRCs may not be known to a Mixer who projects incoming RTP =
streams as is. This was the reason for the application token. (this is a =
general issue with other use cases when using SSRC in SDP and bundle =
tried to address part of it for identifying the received RTP streams =
using the SDP group mid values)

2. For this case instead of using ssrc-group we proposed appId-group and =
define the mapping between an appId and the pt number to help identify =
the relation in the SDP between the FEC streams and the source streams.


We had the feeling that the using the SSRC in SDP was causing multiple =
issues and by using a token that does not have an RTP layer semantics we =
can have a cleaner solution.


Roni

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio =
Garcia
> Murillo
> Sent: 29 October, 2014 2:30 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] SDP and ssrc-group,
>=20
>  From a formal perspective, I don't feel it is correct to describe the =
semantical
> meaning of an ssrc by referring to a payload.
>=20
> Best regards
> Sergio
> On 29/10/2014 8:48, Roni Even wrote:
> > Hi,
> > We discussed this mapping issue in a previous version of the =
application token
> draft =
http://tools.ietf.org/html/draft-even-mmusic-application-token-02
> example in section 1.
> > We thought of having a mapping from appId to "pt" (section 3.3.1) =
Roni
> >
> >> -----Original Message-----
> >> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of I?aki =
Baz
> >> Castillo
> >> Sent: 21 October, 2014 7:26 PM
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] SDP and ssrc-group,
> >>
> >> Hi,
> >>
> >> May I know which SSRC (345259865 or 2693756249) will be used for =
the
> >> real media stream (plus RED and FEC) and which SSRC will be used =
for RTX?
> >>
> >>
> >>
> >> --------------------------
> >> m=3Dvideo 62164 RTP/SAVPF 100 116 117 96 a=3Dmid:video
> >> a=3Drtpmap:100 VP8/90000
> >> a=3Drtpmap:116 red/90000
> >> a=3Drtpmap:117 ulpfec/90000
> >> a=3Drtpmap:96 rtx/90000
> >> a=3Dfmtp:96 apt=3D100
> >> a=3Dssrc-group:FID 345259865 2693756249
> >> a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs
> >> a=3Dssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> >> c0134f05-e7c2-4afd-a979-4e224de5eb91
> >> a=3Dssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> >> a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> >> a=3Dssrc:2693756249 cname:erS7E/KHLYKTejNs
> >> a=3Dssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> >> c0134f05-e7c2-4afd-a979-4e224de5eb91
> >> a=3Dssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
> >> a=3Dssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
> >> -------------------------------
> >>
> >>
> >>
> >>
> >> RFC 5576 does not clarify it at all:
> >>
> >> http://tools.ietf.org/html/rfc5576#section-4.2
> >>
> >> --------------------------------------------------
> >> 4.2.  The "ssrc-group" Media Attribute
> >>
> >>     a=3Dssrc-group:<semantics> <ssrc-id> ...
> >>
> >>     [..]
> >>
> >>     The <semantics> parameter is taken from the specification of =
the
> >>     "group" attribute [RFC3388].  The initial semantic values =
defined for
> >>     the "ssrc-group" attribute are FID (Flow Identification) =
[RFC3388]
> >>     and FEC (Forward Error Correction) [RFC4756].  In each case, =
the
> >>     relationship among the grouped sources is the same as the
> >>     relationship among corresponding sources in media streams =
grouped
> >>     using the SDP "group" attribute.
> >> --------------------------------------------------
> >>
> >>
> >>
> >> The referenced RFC 3388 neither clarifies it:
> >>
> >> ---------------------------------------------------
> >> 7.4 FID Semantics
> >>
> >>     Several "m" lines grouped together using FID semantics form a =
media
> >>     flow.  A media agent handling a media flow that comprises =
several "m"
> >>     lines MUST send a copy of the media to every "m" line part of =
the
> >>     flow as long as the codecs and the direction attribute present =
in a
> >>     particular "m" line allow it.
> >>
> >>     It is assumed that the application uses only one codec at a =
time to
> >>     encode the media produced.  This codec MAY change dynamically =
during
> >>     the session, but at any particular moment only one codec is in =
use.
> >>
> >>     The application encodes the media using the current codec and =
checks
> >>     one by one all the "m" lines that are part of the flow.  If a
> >>     particular "m" line contains the codec being used and the =
direction
> >>     attribute is "sendonly" or "sendrecv", a copy of the encoded =
media is
> >>     sent to the address/port specified in that particular media =
stream.
> >>     If either the "m" line does not contain the codec being used or =
the
> >>     direction attribute is neither "sendonly" nor "sendrecv", =
nothing is
> >>     sent over this media stream.
> >> ----------------------------------------------------
> >>
> >>
> >>
> >>
> >> So, how is the usage of ssrc-group? Where is it really defined?
> >>
> >> Can I put more than 2 ssrc together in the same ssrc-group line?
> >>
> >> How should the receiver interpret it?
> >>
> >> Does a ssrc-group always mean RTX usage? Where is that specified in =
the
> >> above SDP?
> >>
> >> Does not the above SDP look a complete mixture of hacks and =
workarounds?
> >>
> >>
> >>
> >>
> >> --
> >> I=C3=B1aki Baz Castillo
> >> <ibc@aliax.net>
> >>
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Oct 29 08:21:49 2014
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Date: Wed, 29 Oct 2014 16:21:44 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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To: Roni Even <ron.even.tlv@gmail.com>, rtcweb@ietf.org
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Hi Roni,

I agree that knowing ssrcs in the SDP is a burden, but it is not only 
related to FEC/RTX. IMHO that should be handled in Plan A/B/C/whatever 
discussions which I have to confess that have not been actively following.

Best regards
Sergio
On 29/10/2014 16:14, Roni Even wrote:
> Hi,
> In the application token proposal we were trying to address two FEC issues:
>
> 1. You assume that the SDP layer will know the SSRCs of the streams when making an offer. This may be true for point to point call but there are topologies (https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-04) where the SSRCs may not be known to a Mixer who projects incoming RTP streams as is. This was the reason for the application token. (this is a general issue with other use cases when using SSRC in SDP and bundle tried to address part of it for identifying the received RTP streams using the SDP group mid values)
>
> 2. For this case instead of using ssrc-group we proposed appId-group and define the mapping between an appId and the pt number to help identify the relation in the SDP between the FEC streams and the source streams.
>
>
> We had the feeling that the using the SSRC in SDP was causing multiple issues and by using a token that does not have an RTP layer semantics we can have a cleaner solution.
>
>
> Roni
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio Garcia
>> Murillo
>> Sent: 29 October, 2014 2:30 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] SDP and ssrc-group,
>>
>>   From a formal perspective, I don't feel it is correct to describe the semantical
>> meaning of an ssrc by referring to a payload.
>>
>> Best regards
>> Sergio
>> On 29/10/2014 8:48, Roni Even wrote:
>>> Hi,
>>> We discussed this mapping issue in a previous version of the application token
>> draft http://tools.ietf.org/html/draft-even-mmusic-application-token-02
>> example in section 1.
>>> We thought of having a mapping from appId to "pt" (section 3.3.1) Roni
>>>
>>>> -----Original Message-----
>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of I?aki Baz
>>>> Castillo
>>>> Sent: 21 October, 2014 7:26 PM
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] SDP and ssrc-group,
>>>>
>>>> Hi,
>>>>
>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>> real media stream (plus RED and FEC) and which SSRC will be used for RTX?
>>>>
>>>>
>>>>
>>>> --------------------------
>>>> m=video 62164 RTP/SAVPF 100 116 117 96 a=mid:video
>>>> a=rtpmap:100 VP8/90000
>>>> a=rtpmap:116 red/90000
>>>> a=rtpmap:117 ulpfec/90000
>>>> a=rtpmap:96 rtx/90000
>>>> a=fmtp:96 apt=100
>>>> a=ssrc-group:FID 345259865 2693756249
>>>> a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>>>> a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=ssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> a=ssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=ssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>> a=ssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> a=ssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>> a=ssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>> -------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> RFC 5576 does not clarify it at all:
>>>>
>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>
>>>> --------------------------------------------------
>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>
>>>>      a=ssrc-group:<semantics> <ssrc-id> ...
>>>>
>>>>      [..]
>>>>
>>>>      The <semantics> parameter is taken from the specification of the
>>>>      "group" attribute [RFC3388].  The initial semantic values defined for
>>>>      the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>>      and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>      relationship among the grouped sources is the same as the
>>>>      relationship among corresponding sources in media streams grouped
>>>>      using the SDP "group" attribute.
>>>> --------------------------------------------------
>>>>
>>>>
>>>>
>>>> The referenced RFC 3388 neither clarifies it:
>>>>
>>>> ---------------------------------------------------
>>>> 7.4 FID Semantics
>>>>
>>>>      Several "m" lines grouped together using FID semantics form a media
>>>>      flow.  A media agent handling a media flow that comprises several "m"
>>>>      lines MUST send a copy of the media to every "m" line part of the
>>>>      flow as long as the codecs and the direction attribute present in a
>>>>      particular "m" line allow it.
>>>>
>>>>      It is assumed that the application uses only one codec at a time to
>>>>      encode the media produced.  This codec MAY change dynamically during
>>>>      the session, but at any particular moment only one codec is in use.
>>>>
>>>>      The application encodes the media using the current codec and checks
>>>>      one by one all the "m" lines that are part of the flow.  If a
>>>>      particular "m" line contains the codec being used and the direction
>>>>      attribute is "sendonly" or "sendrecv", a copy of the encoded media is
>>>>      sent to the address/port specified in that particular media stream.
>>>>      If either the "m" line does not contain the codec being used or the
>>>>      direction attribute is neither "sendonly" nor "sendrecv", nothing is
>>>>      sent over this media stream.
>>>> ----------------------------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>
>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>
>>>> How should the receiver interpret it?
>>>>
>>>> Does a ssrc-group always mean RTX usage? Where is that specified in the
>>>> above SDP?
>>>>
>>>> Does not the above SDP look a complete mixture of hacks and workarounds?
>>>>
>>>>
>>>>
>>>>
>>>> -


From nobody Wed Oct 29 09:19:51 2014
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Nice to have this information available.


The Duane Morris report is missing information about a couple of
on-going invalidity actions. We've sent information to the I-D editors.
Hopefully, they will update the document to be more complete.


-- 
Surveillance is pervasive. Go Dark.


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      <p dir="ltr"
        style="line-height:1.15;margin-top:0pt;margin-bottom:0pt;">Nice
        to have this information available.</p>
      <br>
      The Duane Morris report is missing information about a couple of
      on-going invalidity actions. We've sent information to the I-D
      editors. Hopefully, they will update the document to be more
      complete.
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    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
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From nobody Wed Oct 29 10:20:51 2014
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From: "David Benham (dbenham)" <dbenham@cisco.com>
To: "Harald Alvestrand (harald@alvestrand.no)" <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Re: [rtcweb] Notification for draft-benham-rtcweb-vp8litigation-00.txt
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We'll be having those looked in to and report/draft subsequently updated.

> Nice to have this information available.
>
>The Duane Morris report is missing information about a couple of on-going =
invalidity actions.  We've=20
>sent information to the I-D editors. Hopefully, they will update the docum=
ent to be more complete.

P.S.    Terriberry, will use a different acronym, RFRAND, in any subsequent=
 I-D update.


From nobody Wed Oct 29 12:34:59 2014
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Subject: Re: [rtcweb] SDP and ssrc-group,
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Correlation of RTP flow to m= line is being done by the mid header
extension. Correlation of an individual RTP flow to its 'role' (e.g. FEC,
RTX) within a m= line can be inferred from received data (e.g. either FEC
or RTX), or determined explicitly from a=ssrc-group attributes, perhaps
with the addition of some additional meta-information.

To be absolutely clear, I was trying to suggest that the first SSRC in an
a=ssrc-group line should indicate the primary stream, and subsequent ones
the secondary streams (e.g. FID/FEC). I was not trying to suggest any other
ordering within the SDP has any bearing on this question.

On Wed, Oct 29, 2014 at 8:21 AM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

> Hi Roni,
>
> I agree that knowing ssrcs in the SDP is a burden, but it is not only
> related to FEC/RTX. IMHO that should be handled in Plan A/B/C/whatever
> discussions which I have to confess that have not been actively following.
>
> Best regards
> Sergio
>
> On 29/10/2014 16:14, Roni Even wrote:
>
>> Hi,
>> In the application token proposal we were trying to address two FEC
>> issues:
>>
>> 1. You assume that the SDP layer will know the SSRCs of the streams when
>> making an offer. This may be true for point to point call but there are
>> topologies (https://tools.ietf.org/html/draft-ietf-avtcore-rtp-
>> topologies-update-04) where the SSRCs may not be known to a Mixer who
>> projects incoming RTP streams as is. This was the reason for the
>> application token. (this is a general issue with other use cases when using
>> SSRC in SDP and bundle tried to address part of it for identifying the
>> received RTP streams using the SDP group mid values)
>>
>> 2. For this case instead of using ssrc-group we proposed appId-group and
>> define the mapping between an appId and the pt number to help identify the
>> relation in the SDP between the FEC streams and the source streams.
>>
>>
>> We had the feeling that the using the SSRC in SDP was causing multiple
>> issues and by using a token that does not have an RTP layer semantics we
>> can have a cleaner solution.
>>
>>
>> Roni
>>
>>  -----Original Message-----
>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio Garcia
>>> Murillo
>>> Sent: 29 October, 2014 2:30 PM
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] SDP and ssrc-group,
>>>
>>>   From a formal perspective, I don't feel it is correct to describe the
>>> semantical
>>> meaning of an ssrc by referring to a payload.
>>>
>>> Best regards
>>> Sergio
>>> On 29/10/2014 8:48, Roni Even wrote:
>>>
>>>> Hi,
>>>> We discussed this mapping issue in a previous version of the
>>>> application token
>>>>
>>> draft http://tools.ietf.org/html/draft-even-mmusic-application-token-02
>>> example in section 1.
>>>
>>>> We thought of having a mapping from appId to "pt" (section 3.3.1) Roni
>>>>
>>>>  -----Original Message-----
>>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of I?aki Baz
>>>>> Castillo
>>>>> Sent: 21 October, 2014 7:26 PM
>>>>> To: rtcweb@ietf.org
>>>>> Subject: [rtcweb] SDP and ssrc-group,
>>>>>
>>>>> Hi,
>>>>>
>>>>> May I know which SSRC (345259865 or 2693756249) will be used for the
>>>>> real media stream (plus RED and FEC) and which SSRC will be used for
>>>>> RTX?
>>>>>
>>>>>
>>>>>
>>>>> --------------------------
>>>>> m=video 62164 RTP/SAVPF 100 116 117 96 a=mid:video
>>>>> a=rtpmap:100 VP8/90000
>>>>> a=rtpmap:116 red/90000
>>>>> a=rtpmap:117 ulpfec/90000
>>>>> a=rtpmap:96 rtx/90000
>>>>> a=fmtp:96 apt=100
>>>>> a=ssrc-group:FID 345259865 2693756249
>>>>> a=ssrc:345259865 cname:erS7E/KHLYKTejNs
>>>>> a=ssrc:345259865 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=ssrc:345259865 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=ssrc:345259865 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=ssrc:2693756249 cname:erS7E/KHLYKTejNs
>>>>> a=ssrc:2693756249 msid:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> a=ssrc:2693756249 mslabel:DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2EtfqBY5
>>>>> a=ssrc:2693756249 label:c0134f05-e7c2-4afd-a979-4e224de5eb91
>>>>> -------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> RFC 5576 does not clarify it at all:
>>>>>
>>>>> http://tools.ietf.org/html/rfc5576#section-4.2
>>>>>
>>>>> --------------------------------------------------
>>>>> 4.2.  The "ssrc-group" Media Attribute
>>>>>
>>>>>      a=ssrc-group:<semantics> <ssrc-id> ...
>>>>>
>>>>>      [..]
>>>>>
>>>>>      The <semantics> parameter is taken from the specification of the
>>>>>      "group" attribute [RFC3388].  The initial semantic values defined
>>>>> for
>>>>>      the "ssrc-group" attribute are FID (Flow Identification) [RFC3388]
>>>>>      and FEC (Forward Error Correction) [RFC4756].  In each case, the
>>>>>      relationship among the grouped sources is the same as the
>>>>>      relationship among corresponding sources in media streams grouped
>>>>>      using the SDP "group" attribute.
>>>>> --------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>> The referenced RFC 3388 neither clarifies it:
>>>>>
>>>>> ---------------------------------------------------
>>>>> 7.4 FID Semantics
>>>>>
>>>>>      Several "m" lines grouped together using FID semantics form a
>>>>> media
>>>>>      flow.  A media agent handling a media flow that comprises several
>>>>> "m"
>>>>>      lines MUST send a copy of the media to every "m" line part of the
>>>>>      flow as long as the codecs and the direction attribute present in
>>>>> a
>>>>>      particular "m" line allow it.
>>>>>
>>>>>      It is assumed that the application uses only one codec at a time
>>>>> to
>>>>>      encode the media produced.  This codec MAY change dynamically
>>>>> during
>>>>>      the session, but at any particular moment only one codec is in
>>>>> use.
>>>>>
>>>>>      The application encodes the media using the current codec and
>>>>> checks
>>>>>      one by one all the "m" lines that are part of the flow.  If a
>>>>>      particular "m" line contains the codec being used and the
>>>>> direction
>>>>>      attribute is "sendonly" or "sendrecv", a copy of the encoded
>>>>> media is
>>>>>      sent to the address/port specified in that particular media
>>>>> stream.
>>>>>      If either the "m" line does not contain the codec being used or
>>>>> the
>>>>>      direction attribute is neither "sendonly" nor "sendrecv", nothing
>>>>> is
>>>>>      sent over this media stream.
>>>>> ----------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> So, how is the usage of ssrc-group? Where is it really defined?
>>>>>
>>>>> Can I put more than 2 ssrc together in the same ssrc-group line?
>>>>>
>>>>> How should the receiver interpret it?
>>>>>
>>>>> Does a ssrc-group always mean RTX usage? Where is that specified in the
>>>>> above SDP?
>>>>>
>>>>> Does not the above SDP look a complete mixture of hacks and
>>>>> workarounds?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> -
>>>>>
>>>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Correlation of RTP flow to m=3D line is being done by the =
mid header extension. Correlation of an individual RTP flow to its &#39;rol=
e&#39; (e.g. FEC, RTX) within a m=3D line can be inferred from received dat=
a (e.g. either FEC or RTX), or determined explicitly from a=3Dssrc-group at=
tributes, perhaps with the addition of some additional meta-information.<di=
v><br></div><div>To be absolutely clear, I was trying to suggest that the f=
irst SSRC in an a=3Dssrc-group line should indicate the primary stream, and=
 subsequent ones the secondary streams (e.g. FID/FEC). I was not trying to =
suggest any other ordering within the SDP has any bearing on this question.=
</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On We=
d, Oct 29, 2014 at 8:21 AM, Sergio Garcia Murillo <span dir=3D"ltr">&lt;<a =
href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.ga=
rcia.murillo@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">Hi Roni,<br>
<br>
I agree that knowing ssrcs in the SDP is a burden, but it is not only relat=
ed to FEC/RTX. IMHO that should be handled in Plan A/B/C/whatever discussio=
ns which I have to confess that have not been actively following.<br>
<br>
Best regards<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Sergio</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
On 29/10/2014 16:14, Roni Even wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi,<br>
In the application token proposal we were trying to address two FEC issues:=
<br>
<br>
1. You assume that the SDP layer will know the SSRCs of the streams when ma=
king an offer. This may be true for point to point call but there are topol=
ogies (<a href=3D"https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topolo=
gies-update-04" target=3D"_blank">https://tools.ietf.org/html/<u></u>draft-=
ietf-avtcore-rtp-<u></u>topologies-update-04</a>) where the SSRCs may not b=
e known to a Mixer who projects incoming RTP streams as is. This was the re=
ason for the application token. (this is a general issue with other use cas=
es when using SSRC in SDP and bundle tried to address part of it for identi=
fying the received RTP streams using the SDP group mid values)<br>
<br>
2. For this case instead of using ssrc-group we proposed appId-group and de=
fine the mapping between an appId and the pt number to help identify the re=
lation in the SDP between the FEC streams and the source streams.<br>
<br>
<br>
We had the feeling that the using the SSRC in SDP was causing multiple issu=
es and by using a token that does not have an RTP layer semantics we can ha=
ve a cleaner solution.<br>
<br>
<br>
Roni<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_=
blank">rtcweb-bounces@ietf.<u></u>org</a>] On Behalf Of Sergio Garcia<br>
Murillo<br>
Sent: 29 October, 2014 2:30 PM<br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: Re: [rtcweb] SDP and ssrc-group,<br>
<br>
=C2=A0 From a formal perspective, I don&#39;t feel it is correct to describ=
e the semantical<br>
meaning of an ssrc by referring to a payload.<br>
<br>
Best regards<br>
Sergio<br>
On 29/10/2014 8:48, Roni Even wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi,<br>
We discussed this mapping issue in a previous version of the application to=
ken<br>
</blockquote>
draft <a href=3D"http://tools.ietf.org/html/draft-even-mmusic-application-t=
oken-02" target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-even-mmu=
sic-application-<u></u>token-02</a><br>
example in section 1.<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
We thought of having a mapping from appId to &quot;pt&quot; (section 3.3.1)=
 Roni<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_=
blank">rtcweb-bounces@ietf.<u></u>org</a>] On Behalf Of I?aki Baz<br>
Castillo<br>
Sent: 21 October, 2014 7:26 PM<br>
To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
Subject: [rtcweb] SDP and ssrc-group,<br>
<br>
Hi,<br>
<br>
May I know which SSRC (345259865 or <a href=3D"tel:2693756249" value=3D"+12=
693756249" target=3D"_blank">2693756249</a>) will be used for the<br>
real media stream (plus RED and FEC) and which SSRC will be used for RTX?<b=
r>
<br>
<br>
<br>
--------------------------<br>
m=3Dvideo 62164 RTP/SAVPF 100 116 117 96 a=3Dmid:video<br>
a=3Drtpmap:100 VP8/90000<br>
a=3Drtpmap:116 red/90000<br>
a=3Drtpmap:117 ulpfec/90000<br>
a=3Drtpmap:96 rtx/90000<br>
a=3Dfmtp:96 apt=3D100<br>
a=3Dssrc-group:FID 345259865 <a href=3D"tel:2693756249" value=3D"+126937562=
49" target=3D"_blank">2693756249</a><br>
a=3Dssrc:345259865 cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:345259865 msid:<u></u>DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2E<u></u>tfqBY5<=
br>
c0134f05-e7c2-4afd-a979-<u></u>4e224de5eb91<br>
a=3Dssrc:345259865 mslabel:<u></u>DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2E<u></u>tfqB=
Y5<br>
a=3Dssrc:345259865 label:c0134f05-e7c2-4afd-a979-<u></u>4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> cname:erS7E/KHLYKTejNs<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> msid:<u></u>DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2E<u></u>tfqBY5<br=
>
c0134f05-e7c2-4afd-a979-<u></u>4e224de5eb91<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> mslabel:<u></u>DWpWct9bWKzTMNYZn5bKVgwZ8Mfy2E<u></u>tfqBY5=
<br>
a=3Dssrc:<a href=3D"tel:2693756249" value=3D"+12693756249" target=3D"_blank=
">2693756249</a> label:c0134f05-e7c2-4afd-a979-<u></u>4e224de5eb91<br>
------------------------------<u></u>-<br>
<br>
<br>
<br>
<br>
RFC 5576 does not clarify it at all:<br>
<br>
<a href=3D"http://tools.ietf.org/html/rfc5576#section-4.2" target=3D"_blank=
">http://tools.ietf.org/html/<u></u>rfc5576#section-4.2</a><br>
<br>
------------------------------<u></u>--------------------<br>
4.2.=C2=A0 The &quot;ssrc-group&quot; Media Attribute<br>
<br>
=C2=A0 =C2=A0 =C2=A0a=3Dssrc-group:&lt;semantics&gt; &lt;ssrc-id&gt; ...<br=
>
<br>
=C2=A0 =C2=A0 =C2=A0[..]<br>
<br>
=C2=A0 =C2=A0 =C2=A0The &lt;semantics&gt; parameter is taken from the speci=
fication of the<br>
=C2=A0 =C2=A0 =C2=A0&quot;group&quot; attribute [RFC3388].=C2=A0 The initia=
l semantic values defined for<br>
=C2=A0 =C2=A0 =C2=A0the &quot;ssrc-group&quot; attribute are FID (Flow Iden=
tification) [RFC3388]<br>
=C2=A0 =C2=A0 =C2=A0and FEC (Forward Error Correction) [RFC4756].=C2=A0 In =
each case, the<br>
=C2=A0 =C2=A0 =C2=A0relationship among the grouped sources is the same as t=
he<br>
=C2=A0 =C2=A0 =C2=A0relationship among corresponding sources in media strea=
ms grouped<br>
=C2=A0 =C2=A0 =C2=A0using the SDP &quot;group&quot; attribute.<br>
------------------------------<u></u>--------------------<br>
<br>
<br>
<br>
The referenced RFC 3388 neither clarifies it:<br>
<br>
------------------------------<u></u>---------------------<br>
7.4 FID Semantics<br>
<br>
=C2=A0 =C2=A0 =C2=A0Several &quot;m&quot; lines grouped together using FID =
semantics form a media<br>
=C2=A0 =C2=A0 =C2=A0flow.=C2=A0 A media agent handling a media flow that co=
mprises several &quot;m&quot;<br>
=C2=A0 =C2=A0 =C2=A0lines MUST send a copy of the media to every &quot;m&qu=
ot; line part of the<br>
=C2=A0 =C2=A0 =C2=A0flow as long as the codecs and the direction attribute =
present in a<br>
=C2=A0 =C2=A0 =C2=A0particular &quot;m&quot; line allow it.<br>
<br>
=C2=A0 =C2=A0 =C2=A0It is assumed that the application uses only one codec =
at a time to<br>
=C2=A0 =C2=A0 =C2=A0encode the media produced.=C2=A0 This codec MAY change =
dynamically during<br>
=C2=A0 =C2=A0 =C2=A0the session, but at any particular moment only one code=
c is in use.<br>
<br>
=C2=A0 =C2=A0 =C2=A0The application encodes the media using the current cod=
ec and checks<br>
=C2=A0 =C2=A0 =C2=A0one by one all the &quot;m&quot; lines that are part of=
 the flow.=C2=A0 If a<br>
=C2=A0 =C2=A0 =C2=A0particular &quot;m&quot; line contains the codec being =
used and the direction<br>
=C2=A0 =C2=A0 =C2=A0attribute is &quot;sendonly&quot; or &quot;sendrecv&quo=
t;, a copy of the encoded media is<br>
=C2=A0 =C2=A0 =C2=A0sent to the address/port specified in that particular m=
edia stream.<br>
=C2=A0 =C2=A0 =C2=A0If either the &quot;m&quot; line does not contain the c=
odec being used or the<br>
=C2=A0 =C2=A0 =C2=A0direction attribute is neither &quot;sendonly&quot; nor=
 &quot;sendrecv&quot;, nothing is<br>
=C2=A0 =C2=A0 =C2=A0sent over this media stream.<br>
------------------------------<u></u>----------------------<br>
<br>
<br>
<br>
<br>
So, how is the usage of ssrc-group? Where is it really defined?<br>
<br>
Can I put more than 2 ssrc together in the same ssrc-group line?<br>
<br>
How should the receiver interpret it?<br>
<br>
Does a ssrc-group always mean RTX usage? Where is that specified in the<br>
above SDP?<br>
<br>
Does not the above SDP look a complete mixture of hacks and workarounds?<br=
>
<br>
<br>
<br>
<br>
-<br>
</blockquote></blockquote></blockquote></blockquote>
<br></div></div><div class=3D"HOEnZb"><div class=3D"h5">
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Date: Wed, 29 Oct 2014 20:17:14 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com> <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com> <8CA6F4DE-8978-458B-9C77-21CFA33C48CE@lurchi.franken.de>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi Michael,
I moved to tsvwg as requested. See my reply to Karen, and one reply inline.
http://www.ietf.org/mail-archive/web/tsvwg/current/msg12899.html

For rtcweb, the main point is this. Will rtcweb apps care about blocking on=
 potentially fragmented messages? I think they definitely will, if they car=
e about blocking at all. Even simple JSON objects, which seem like natural =
message payloads, can easily exceed common MTUs. It doesn=92t take huge fil=
e transfers to require fragmentation.

Cheers,
Mo

On 10/29/14, 6:05 AM, Michael Tuexen <Michael.Tuexen@lurchi.franken.de<mail=
to:Michael.Tuexen@lurchi.franken.de>> wrote:
On 28 Oct 2014, at 23:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mailto:mza=
naty@cisco.com>> wrote:
On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" <Michael.Tuexen@lurchi.franke=
n.de<mailto:Michael.Tuexen@lurchi.franken.de>> wrote:
On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mailto:mza=
naty@cisco.com>> wrote:
What resource is being conserved by not just requiring it?
8 bytes per packet (for MID+FSN).
Michael, the current NDATA draft requires it to be used for all messages if=
 negotiated. Can=92t this be relaxed / optimized to use it for all *fragmen=
ted* messages but use DATA for non-fragmented messages to avoid the 8 byte =
overhead for small messages (like game telemetry/control, an often cited us=
e of data channels).
You can bring this up at the tsvwg mailing list.
Mo: Will do.
However, we normally do not optimize for byte saving... Mixing
both makes the processing harder... Maybe we can use a bit in the flags to =
indicate if the additional fields are
there...
Mo: Type should already be enough without flags.
The current NDATA draft also says you can=92t interleave fragmented message=
s in a stream, so head of line blocking remains for all fragmented messages=
, while only small non-fragmented messages can avoid it. This dilutes the u=
tility of NDATA, perhaps enough to make apps that really care about head of=
 line blocking to implement their own app layer fragmentation with messages=
 well below common MTUs, hence defeating NDATA. Can=92t this also be relaxe=
d since MID is part of the NDATA extra header?
This would only make sense for messages marked as unordered... For ordered =
it doesn't make sense.
Mo: The restriction applies to unordered as well as ordered. Even for order=
ed, it does make sense to avoid head of line blocking at the sender.
I don't understand how you can do it for ordered. The sender sends a large =
message on stream 0 and after that
a small on stream zero. For both ordered delivery is request. So the receiv=
er can't deliver the second before the
first.

Mo: Blocking sender transmission is very different from blocking receiver d=
elivery to the app. It is wrong to assume all ordered protocols should bloc=
k sender transmission out of order. Windows and SACK improve performance of=
 ordered protocols despite the receiver blocking delivery to the app.

If those two issues are resolved, I see no argument against making NDATA a =
MUST in data channels.
Please let us discuss NDATA issues on the tsvwg mailing list...
Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its benefits=
 outweigh its costs. The current draft does not pass that bar for me. If my=
 app cares about HOL blocking, NDATA is not very useful, so my app must int=
ernally fragment, hence NDATA actually becomes harmful (8 bytes closer to e=
xceeding MTU, forcing SCTP fragments and hence risking HOL blocking).
I don't understand the point when I look at the API for data channels... On=
ly for sending messages
on a data channel not requiring ordered delivery. However, the intention of=
 NDATA is to mitigate
the impact of one data channel on others belonging to the same peer connect=
ion. This goal is met.
Best regards
Michael
Best regards
Michael
Mo
PS. Some believe data channels will be more widely used than WebRTC media (=
webtorrent, etc.), so it does make sense to consider the desirable properti=
es of a general peer-to-peer transport, and drop the WebRTC prefix when tal=
king about data channels.



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</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; font-family: Arial, sans-serif; font-size: 12px=
; color: rgb(0, 0, 0);">
<div>Hi Michael,</div>
<div>I moved to tsvwg as requested. See my reply to Karen, and one reply in=
line.</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/tsvwg/current/msg12899=
.html">http://www.ietf.org/mail-archive/web/tsvwg/current/msg12899.html</a>=
</div>
<div><br>
</div>
<div>For rtcweb, the main point is this. Will rtcweb apps care about blocki=
ng on potentially fragmented messages? I think they definitely will, if the=
y care about blocking at all. Even simple JSON objects, which seem like nat=
ural message payloads, can easily
 exceed common MTUs. It doesn=92t take huge file transfers to require fragm=
entation.</div>
<div><br>
</div>
<div>Cheers,</div>
<div>Mo</div>
<div><br>
</div>
<div>On 10/29/14, 6:05 AM, Michael Tuexen &lt;<a href=3D"mailto:Michael.Tue=
xen@lurchi.franken.de">Michael.Tuexen@lurchi.franken.de</a>&gt; wrote:</div=
>
<div>On 28 Oct 2014, at 23:28, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:mz=
anaty@cisco.com">mzanaty@cisco.com</a>&gt; wrote:</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>On Oct 28, 2014, at 5:11 PM, &quot;Michael Tuexen&quot; &lt;<a href=3D=
"mailto:Michael.Tuexen@lurchi.franken.de">Michael.Tuexen@lurchi.franken.de<=
/a>&gt; wrote:</div>
<div></div>
<div>On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:mz=
anaty@cisco.com">mzanaty@cisco.com</a>&gt; wrote:</div>
<div></div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>What resource is being conserved by not just requiring it?</div>
</blockquote>
<div></div>
<div>8 bytes per packet (for MID&#43;FSN).</div>
<div></div>
<div>Michael, the current NDATA draft requires it to be used for all messag=
es if negotiated. Can=92t this be relaxed / optimized to use it for all *fr=
agmented* messages but use DATA for non-fragmented messages to avoid the 8 =
byte overhead for small messages (like
 game telemetry/control, an often cited use of data channels).</div>
</blockquote>
<div>You can bring this up at the tsvwg mailing list.</div>
<div></div>
<div>Mo: Will do.</div>
<div></div>
<div>However, we normally do not optimize for byte saving... Mixing</div>
<div>both makes the processing harder... Maybe we can use a bit in the flag=
s to indicate if the additional fields are</div>
<div>there...</div>
<div></div>
<div>Mo: Type should already be enough without flags.</div>
<div></div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>The current NDATA draft also says you can=92t interleave fragmented me=
ssages in a stream, so head of line blocking remains for all fragmented mes=
sages, while only small non-fragmented messages can avoid it. This dilutes =
the utility of NDATA, perhaps enough
 to make apps that really care about head of line blocking to implement the=
ir own app layer fragmentation with messages well below common MTUs, hence =
defeating NDATA. Can=92t this also be relaxed since MID is part of the NDAT=
A extra header?</div>
</blockquote>
<div>This would only make sense for messages marked as unordered... For ord=
ered it doesn't make sense.</div>
<div></div>
<div>Mo: The restriction applies to unordered as well as ordered. Even for =
ordered, it does make sense to avoid head of line blocking at the sender.
</div>
</blockquote>
<div>I don't understand how you can do it for ordered. The sender sends a l=
arge message on stream 0 and after that</div>
<div>a small on stream zero. For both ordered delivery is request. So the r=
eceiver can't deliver the second before the</div>
<div>first.</div>
<div><br>
</div>
<div>Mo: Blocking sender transmission is very different from blocking recei=
ver delivery to the app. It is wrong to assume all ordered protocols should=
 block sender transmission out of order. Windows and SACK improve performan=
ce of ordered protocols despite
 the receiver blocking delivery to the app.</div>
<div>&nbsp;</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>If those two issues are resolved, I see no argument against making NDA=
TA a MUST in data channels.</div>
</blockquote>
<div>Please let us discuss NDATA issues on the tsvwg mailing list...</div>
<div></div>
<div>Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its ben=
efits outweigh its costs. The current draft does not pass that bar for me. =
If my app cares about HOL blocking, NDATA is not very useful, so my app mus=
t internally fragment, hence NDATA
 actually becomes harmful (8 bytes closer to exceeding MTU, forcing SCTP fr=
agments and hence risking HOL blocking).</div>
</blockquote>
<div>I don't understand the point when I look at the API for data channels.=
.. Only for sending messages</div>
<div>on a data channel not requiring ordered delivery. However, the intenti=
on of NDATA is to mitigate</div>
<div>the impact of one data channel on others belonging to the same peer co=
nnection. This goal is met.</div>
<div>Best regards</div>
<div>Michael</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>Best regards</div>
<div>Michael</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div></div>
<div>Mo</div>
<div></div>
<div>PS. Some believe data channels will be more widely used than WebRTC me=
dia (webtorrent, etc.), so it does make sense to consider the desirable pro=
perties of a general peer-to-peer transport, and drop the WebRTC prefix whe=
n talking about data channels.</div>
</blockquote>
<div></div>
<div></div>
</blockquote>
<div><br>
</div>
<div><br>
</div>
</body>
</html>

--_000_D076BBBF3D478mzanatyciscocom_--


From nobody Thu Oct 30 11:54:53 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Date: Thu, 30 Oct 2014 19:53:56 +0100
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com> <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com> <8CA6F4DE-8978-458B-9C77-21CFA33C48CE@lurchi.franken. de> <D076BBBF.3D478%mzanaty@cisco.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 29 Oct 2014, at 21:17, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

> Hi Michael,
> I moved to tsvwg as requested. See my reply to Karen, and one reply =
inline.
> http://www.ietf.org/mail-archive/web/tsvwg/current/msg12899.html
Great, thanks. I'll respond, but I'm really slow right now, since my day =
job
needs more time than I would like...
>=20
> For rtcweb, the main point is this. Will rtcweb apps care about =
blocking on potentially fragmented messages? I think they definitely =
will, if they care about blocking at all. Even simple JSON objects, =
which seem like natural message payloads, can easily exceed common MTUs. =
It doesn=92t take huge file transfers to require fragmentation.
Can you be a bit more specific? My understanding is:
1. Messages sent on one data channel should not block messages sent on a =
different channel.
   I hope we agree on this and this goal can be reached with N-DATA.
2. If you send a larger message on an ordered data channel, it will =
block other
   messages sent on the same data channel. Since it is an ordered =
channel, this can't
   be avoided. I hope we also agree on this.
3. If you send a large message on an unordered data channel, and a small =
one after
   that, there is no statement made in which these messages would be =
delivered.
   Are you focussing on this and do you want to require some sort of =
reordering?

Best regards
Michael
>=20
> Cheers,
> Mo
>=20
> On 10/29/14, 6:05 AM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> On 28 Oct 2014, at 23:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:
>> On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" =
<Michael.Tuexen@lurchi.franken.de> wrote:
>> On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:
>>>> What resource is being conserved by not just requiring it?
>>> 8 bytes per packet (for MID+FSN).
>>> Michael, the current NDATA draft requires it to be used for all =
messages if negotiated. Can=92t this be relaxed / optimized to use it =
for all *fragmented* messages but use DATA for non-fragmented messages =
to avoid the 8 byte overhead for small messages (like game =
telemetry/control, an often cited use of data channels).
>> You can bring this up at the tsvwg mailing list.
>> Mo: Will do.
>> However, we normally do not optimize for byte saving... Mixing
>> both makes the processing harder... Maybe we can use a bit in the =
flags to indicate if the additional fields are
>> there...
>> Mo: Type should already be enough without flags.
>>> The current NDATA draft also says you can=92t interleave fragmented =
messages in a stream, so head of line blocking remains for all =
fragmented messages, while only small non-fragmented messages can avoid =
it. This dilutes the utility of NDATA, perhaps enough to make apps that =
really care about head of line blocking to implement their own app layer =
fragmentation with messages well below common MTUs, hence defeating =
NDATA. Can=92t this also be relaxed since MID is part of the NDATA extra =
header?
>> This would only make sense for messages marked as unordered... For =
ordered it doesn't make sense.
>> Mo: The restriction applies to unordered as well as ordered. Even for =
ordered, it does make sense to avoid head of line blocking at the =
sender.
> I don't understand how you can do it for ordered. The sender sends a =
large message on stream 0 and after that
> a small on stream zero. For both ordered delivery is request. So the =
receiver can't deliver the second before the
> first.
>=20
> Mo: Blocking sender transmission is very different from blocking =
receiver delivery to the app. It is wrong to assume all ordered =
protocols should block sender transmission out of order. Windows and =
SACK improve performance of ordered protocols despite the receiver =
blocking delivery to the app.
> =20
>>> If those two issues are resolved, I see no argument against making =
NDATA a MUST in data channels.
>> Please let us discuss NDATA issues on the tsvwg mailing list...
>> Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its =
benefits outweigh its costs. The current draft does not pass that bar =
for me. If my app cares about HOL blocking, NDATA is not very useful, so =
my app must internally fragment, hence NDATA actually becomes harmful (8 =
bytes closer to exceeding MTU, forcing SCTP fragments and hence risking =
HOL blocking).
> I don't understand the point when I look at the API for data =
channels... Only for sending messages
> on a data channel not requiring ordered delivery. However, the =
intention of NDATA is to mitigate
> the impact of one data channel on others belonging to the same peer =
connection. This goal is met.
> Best regards
> Michael
>> Best regards
>> Michael
>>> Mo
>>> PS. Some believe data channels will be more widely used than WebRTC =
media (webtorrent, etc.), so it does make sense to consider the =
desirable properties of a general peer-to-peer transport, and drop the =
WebRTC prefix when talking about data channels.
>=20
>=20


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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com> <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com> <8CA6F4DE-8978-458B-9C77-21CFA33C48CE@lurchi.franken. de> <D076BBBF.3D478%mzanaty@cisco.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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On 29 Oct 2014, at 21:17, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

> Hi Michael,
> I moved to tsvwg as requested. See my reply to Karen, and one reply =
inline.
> http://www.ietf.org/mail-archive/web/tsvwg/current/msg12899.html
>=20
> For rtcweb, the main point is this. Will rtcweb apps care about =
blocking on potentially fragmented messages? I think they definitely =
will, if they care about blocking at all. Even simple JSON objects, =
which seem like natural message payloads, can easily exceed common MTUs. =
It doesn=92t take huge file transfers to require fragmentation.
After thinking about it a bit more:

Assume an unordered data channel dc and the application calls
dc.send(big_msg);
dc.send(small_msg);
How do you decide whether the intended behaviour is either
(a) deliver big_msg before small_msg
(b) deliver small_msg before big_msg
I don't see any way in the W3C API to indicate this. Both alternative
might the appropriate in different scenarios.

Best regards
Michael
>=20
> Cheers,
> Mo
>=20
> On 10/29/14, 6:05 AM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> On 28 Oct 2014, at 23:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:
>> On Oct 28, 2014, at 5:11 PM, "Michael Tuexen" =
<Michael.Tuexen@lurchi.franken.de> wrote:
>> On 28 Oct 2014, at 21:52, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:
>>>> What resource is being conserved by not just requiring it?
>>> 8 bytes per packet (for MID+FSN).
>>> Michael, the current NDATA draft requires it to be used for all =
messages if negotiated. Can=92t this be relaxed / optimized to use it =
for all *fragmented* messages but use DATA for non-fragmented messages =
to avoid the 8 byte overhead for small messages (like game =
telemetry/control, an often cited use of data channels).
>> You can bring this up at the tsvwg mailing list.
>> Mo: Will do.
>> However, we normally do not optimize for byte saving... Mixing
>> both makes the processing harder... Maybe we can use a bit in the =
flags to indicate if the additional fields are
>> there...
>> Mo: Type should already be enough without flags.
>>> The current NDATA draft also says you can=92t interleave fragmented =
messages in a stream, so head of line blocking remains for all =
fragmented messages, while only small non-fragmented messages can avoid =
it. This dilutes the utility of NDATA, perhaps enough to make apps that =
really care about head of line blocking to implement their own app layer =
fragmentation with messages well below common MTUs, hence defeating =
NDATA. Can=92t this also be relaxed since MID is part of the NDATA extra =
header?
>> This would only make sense for messages marked as unordered... For =
ordered it doesn't make sense.
>> Mo: The restriction applies to unordered as well as ordered. Even for =
ordered, it does make sense to avoid head of line blocking at the =
sender.
> I don't understand how you can do it for ordered. The sender sends a =
large message on stream 0 and after that
> a small on stream zero. For both ordered delivery is request. So the =
receiver can't deliver the second before the
> first.
>=20
> Mo: Blocking sender transmission is very different from blocking =
receiver delivery to the app. It is wrong to assume all ordered =
protocols should block sender transmission out of order. Windows and =
SACK improve performance of ordered protocols despite the receiver =
blocking delivery to the app.
> =20
>>> If those two issues are resolved, I see no argument against making =
NDATA a MUST in data channels.
>> Please let us discuss NDATA issues on the tsvwg mailing list...
>> Mo: Agreed, will do. But I think rtcweb MUST use NDATA only if its =
benefits outweigh its costs. The current draft does not pass that bar =
for me. If my app cares about HOL blocking, NDATA is not very useful, so =
my app must internally fragment, hence NDATA actually becomes harmful (8 =
bytes closer to exceeding MTU, forcing SCTP fragments and hence risking =
HOL blocking).
> I don't understand the point when I look at the API for data =
channels... Only for sending messages
> on a data channel not requiring ordered delivery. However, the =
intention of NDATA is to mitigate
> the impact of one data channel on others belonging to the same peer =
connection. This goal is met.
> Best regards
> Michael
>> Best regards
>> Michael
>>> Mo
>>> PS. Some believe data channels will be more widely used than WebRTC =
media (webtorrent, etc.), so it does make sense to consider the =
desirable properties of a general peer-to-peer transport, and drop the =
WebRTC prefix when talking about data channels.
>=20
>=20


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Meaning of SHOULD support/use interleaving
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Date: Fri, 31 Oct 2014 18:17:55 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1D4CCEEF@ESESSMB209.ericsson.se> <EA00639E-2008-4BD2-88F2-27AAEE9DA213@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CD241@ESESSMB209.ericsson.se> <544E8D92.8010401@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D4CE9DA@ESESSMB209.ericsson.se> <5B74C903-7C9A-4D74-B5CC-D0643A377935@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4CEBAB@ESESSMB209.ericsson.se> <9A1C16CE-1423-448B-943F-33B3068156F3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D0CE8@ESESSMB209.ericsson.se> <CD364160-0A33-4E74-B114-04BDA33E40D3@lurchi.franken.de> <7594FB04B1934943A5C02806D1A2204B1D4D1044@ESESSMB209.ericsson.se> <CABcZeBOuiTk8Ont314aenemUVPxkiLk7jScpZ3DR-90TuzN4NQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4D14F7@ESESSMB209.ericsson.se> <D0757214.3D0B4%mzanaty@cisco.com> <4704857B-ED34-4B3D-9CC6-A60801586F6B@lurchi.franken.de> <3904F698-8B45-44E9-9E43-81D53467E3A6@cisco.com> <8CA6F4DE-8978-458B-9C77-21CFA33C48CE@lurchi.franken.de> <D076BBBF.3D478%mzanaty@cisco.com> <9DFFB0BA-078A-44BC-BE78-E31077A2174C@lurchi.franken.de>
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Subject: Re: [rtcweb] Meaning of SHOULD support/use interleaving
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Hi Michael,
Merging your points and replying here since we are firmly in rtcweb cases.
We can summarize for tsvwg once concluded (if we reach any conclusion).

Consider real rtcweb apps that have many potentially large blobs to
transfer, like proxy browsing, webtorrent, chats with file exchange, games
with object exchange, etc. Developers may think sending these blobs as
messages on an unordered channel would work without HOL blocking. But they
would be wrong since the current NDATA spec actually blocks, effectively
behaving like an ordered channel.

Once they debug this, they will rewrite the app to either:
1) create a new channel for every blob, which is excessive overhead for
the app, browser and SCTP stack (but NDATA does at least enable this
solution), or
2) fragment blobs in the app, which adds (some but less) overhead for the
app (and defeats NDATA).

These are the only ways to get unordered, non-blocking delivery. At least
NDATA makes the app hack 1) possible since channels don=B9t block each
other. But it would be far more useful if unordered channels did not block
like ordered ones.

More inline.

Cheers,
Mo

On 10/30/14, 2:53 PM, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
wrote:
Can you be a bit more specific? My understanding is:
1. Messages sent on one data channel should not block messages sent on a
different channel.
   I hope we agree on this and this goal can be reached with N-DATA.
2. If you send a larger message on an ordered data channel, it will block
other
   messages sent on the same data channel. Since it is an ordered channel,
this can't
   be avoided. I hope we also agree on this.
3. If you send a large message on an unordered data channel, and a small
one after
   that, there is no statement made in which these messages would be
delivered.
   Are you focussing on this and do you want to require some sort of
reordering?

Mo: Agreed on 1 and 2, with the admittedly minor caveat mentioned earlier
on 2 about blocking sender transmission vs. receiver delivery to app. For
3, which is indeed my focus, agreed the W3C spec makes no statement about
order (since the channel is unordered!), but the NDATA spec actually does
mandate HOL order for messages>MTU, which is counter to the whole point of
an unordered channel. A developer must override defaults to get an
unordered channel, presumably to get faster, non-blocking delivery for the
individual messages. Otherwise, what is the point of unordered channels if
they block just like ordered ones?


On 10/30/14, 5:55 PM, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
wrote:
Assume an unordered data channel dc and the application calls
dc.send(big_msg);
dc.send(small_msg);
How do you decide whether the intended behaviour is either
(a) deliver big_msg before small_msg
(b) deliver small_msg before big_msg
I don't see any way in the W3C API to indicate this. Both alternative
might the appropriate in different scenarios.

Mo: The point of unordered channels is to send and receive everything
whenever it becomes available. The transport does not need to worry about
providing any specific ordering, but it is not expected to block based on
internal ordering rules.

